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webrtc / src / / 11376fb992f2b6f080f550e419e6558b1d953701 / media / engine / webrtc_voice_engine.h
  1. 5866e1a Rename Set(Send|Recv)Parameters Set(Sender|Receiver)Parameters by Philipp Hancke · 1 year, 7 months ago
  2. 43a5dd8 Implement codec selection api by Florent Castelli · 1 year, 7 months ago
  3. a9d5141 Rename cricket::RtpParameters and derived classes by Philipp Hancke · 1 year, 8 months ago
  4. d797cb6 Remove all split channels related code by Florent Castelli · 1 year, 9 months ago
  5. 84fdf99 Convert Media*Channel to contain a webrtc::Transport by Harald Alvestrand · 1 year, 9 months ago
  6. ee97e6a Move GetSendCodec() to MediaSendChannelInterface by Florent Castelli · 1 year, 9 months ago
  7. 09e0086 Remove ImplForTesting function from MediaChannel by Harald Alvestrand · 1 year, 10 months ago
  8. 77c6230 Add create functions for voice media send and receive channels. by Harald Alvestrand · 1 year, 10 months ago
  9. 2f0c078 Split WebRtcVoiceChannel into Send and Receive classes by Harald Alvestrand · 1 year, 10 months ago
  10. 9a34d80 Apply the "shim" pattern for WebRtcVoiceEngine by Harald Alvestrand · 1 year, 10 months ago
  11. 4ad141e Add callback for send codec in audio too by Harald Alvestrand · 1 year, 10 months ago
  12. a9bba04 Updating AsyncAudioProcessing API, part 1. by Peter Hanspers · 1 year, 10 months ago
  13. 13897e6 Change SSRC-passing for MediaChannel from external to callback by Harald Alvestrand · 1 year, 10 months ago
  14. c848268 Use SequenceChecker(SequenceChecker::kDetached) in a few places. by Tommi · 2 years ago
  15. 2f55370 Reland "Use two MediaChannels for 2 directions." by Harald Alvestrand · 2 years, 1 month ago
  16. 18c869b Revert "Use two MediaChannels for 2 directions." by Harald Alvestrand · 2 years, 1 month ago
  17. 8981a6f Use two MediaChannels for 2 directions. by Harald Alvestrand · 2 years, 1 month ago
  18. 16579cc Change MediaChannel to have a Role parameter by Harald Alvestrand · 2 years, 2 months ago
  19. 101c6aa Remove leftover function signatures. by Fredrik Solenberg · 2 years, 2 months ago
  20. 9ad10bc Only generate codec stats for the voice send and receive codec by Philipp Hancke · 2 years, 2 months ago
  21. efbe753 Add RTCAudioPlayoutStats to GetStats(). by Fredrik Hernqvist · 2 years, 3 months ago
  22. 89ca299 Use parsed packet from RtpTransport::DemuxPacket in engine and call by Per K · 2 years, 3 months ago
  23. 175f06f Reland "Remove 'trackId' dependency in stats selector algorithm." by Henrik Boström · 2 years, 3 months ago
  24. 1251c64 Split stats generation for MediaChannel into sender and receiver APIs by Harald Alvestrand · 2 years, 3 months ago
  25. 9253240 Reland "Remove use of ReceiveStreamRtpConfig:transport_cc" by Per K · 2 years, 3 months ago
  26. be5c713 Revert "Remove use of ReceiveStreamRtpConfig:transport_cc" by Olga Sharonova · 2 years, 3 months ago
  27. 97ba853 Remove use of ReceiveStreamRtpConfig:transport_cc by Per K · 2 years, 3 months ago
  28. d3ac3b6 Remove AsSendChannel/AsReceiveChannel methods by Harald Alvestrand · 2 years, 3 months ago
  29. acabb36 pc: Add asynchronous RtpSender::SetParameters() call by Florent Castelli · 2 years, 5 months ago
  30. a5ba250 Fix apply frame transformer to MID demuxed audio streams by Lennart Grahl · 2 years, 6 months ago
  31. da6291e WebRTC voice engine: Remove duplicate and confusing logs by Sam Zackrisson · 2 years, 8 months ago
  32. c374d11 Move to_queued_task.h and pending_task_safety_flag.h into public API by Artem Titov · 2 years, 10 months ago
  33. 1def899 Remove legacy (unused) config param: jitter_buffer_enable_rtx_handling by Tommi · 2 years, 11 months ago
  34. e62c2f2 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf by Jonas Oreland · 3 years ago
  35. ff05c5c audio/red: cleanup killswitch by Philipp Hancke · 3 years, 1 month ago
  36. 93348d8 Remove unused audio options and corresponding media constraints by Alessio Bazzica · 3 years, 2 months ago
  37. 94f0194 Remove transport_name_ from Channel. by Tomas Gunnarsson · 3 years, 3 months ago
  38. 2562cf0 Reland "Wire up non-sender RTT for audio, and implement related standardized stats." by Ivo Creusen · 3 years, 7 months ago
  39. 2c41cba Revert "Wire up non-sender RTT for audio, and implement related standardized stats." by Björn Terelius · 3 years, 7 months ago
  40. fb0dca6 Wire up non-sender RTT for audio, and implement related standardized stats. by Ivo Creusen · 3 years, 7 months ago
  41. 773a222 red: enable opus-red by default by Philipp Hancke · 3 years, 8 months ago
  42. 819286e Clean out some leftover junk. by Harald Alvestrand · 3 years, 11 months ago
  43. cf2aeff Remove lock from MediaChannel by Tommi · 3 years, 11 months ago
  44. a63bee5 Remove Mutex from BaseChannel. by Tommi · 4 years ago
  45. eb9c3f2 Handle OnPacketSent on the network thread via MediaChannel. by Tomas Gunnarsson · 4 years ago
  46. 15e078c Fix unsignalled ssrc race in WebRtcVideoChannel. by Henrik Boström · 4 years ago
  47. d15a575 Use SequenceChecker from public API by Artem Titov · 4 years, 2 months ago
  48. c8421c4 Replace rtc::ThreadChecker with webrtc::SequenceChecker by Artem Titov · 4 years, 2 months ago
  49. 33c0ab4 Call MediaChannel::OnPacketReceived on the network thread. by Tomas Gunnarsson · 4 years, 2 months ago
  50. 8467cf2 Reduce redundant flags for audio stream playout state. by Tomas Gunnarsson · 4 years, 2 months ago
  51. 09ceed2 Async audio processing API by Olga Sharonova · 4 years, 6 months ago
  52. de95329 Delete macros RTC_DISALLOW_ASSIGN and RTC_DISALLOW_IMPLICIT_CONSTRUCTORS by Niels Möller · 4 years, 6 months ago
  53. ceb4495 Reland: Wires up WebrtcKeyValueBasedConfig in media engines. by Erik Språng · 4 years, 6 months ago
  54. 5956a17 Revert "Wires up WebrtcKeyValueBasedConfig in media engines." by Artem Titov · 4 years, 7 months ago
  55. 591b2ab Wires up WebrtcKeyValueBasedConfig in media engines. by Erik Språng · 4 years, 7 months ago
  56. 6b4d962 Fix standard GetStats to not modify NetEq state. by Niels Möller · 4 years, 7 months ago
  57. c94650d Remove AudioProcessing::SetExtraOptions. by Mirko Bonadei · 4 years, 7 months ago
  58. 3e9af7f Insert audio frame transformer between depacketizer and decoder. by Marina Ciocea · 5 years ago
  59. d2aa8f9 Insert audio frame transformer between encoder and packetizer. by Marina Ciocea · 5 years ago
  60. 0357b3e RtpTransceiverInterface: add header_extensions_to_offer() by Markus Handell · 5 years ago
  61. 749f660 Enable SSRC 0 in MediaChannel methods by Saurav Das · 5 years ago
  62. 934afc6 Deprecate RtpReceiver's SetParameters method by Saurav Das · 5 years ago
  63. 03fbace Remove apm_helpers, consolidate audio config in WebRtcVoiceEngine by Sam Zackrisson · 5 years ago
  64. ff27da5 Add/remove receive streams with SSRC 0 from media channels by Saurav Das · 5 years ago
  65. 1b83a9e Only handle each RTCP once. by Sebastian Jansson · 6 years ago
  66. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 6 years ago
  67. f40a340 Remove deprecated code related to AEC2 by Per Åhgren · 6 years ago
  68. d2845f8 Removes unused AudioAllocationSettings from voice engine. by Sebastian Jansson · 6 years ago
  69. e8e4dc4 Change StartAecDump methods to work with FILE* and FileWrapper by Niels Möller · 6 years ago
  70. 220f4be Remove some media/ --> pc/ test dependencies by Steve Anton · 6 years ago
  71. 8a9778e Delete unused StartAecDump method with filename argument by Niels Möller · 6 years ago
  72. 4c7112a Reland "in WebrtcVoiceEngine allow to set TaskQueueFactory" by Danil Chapovalov · 6 years ago
  73. f0d1c03 Add replacement interface for webrtc::GainConrol by Sam Zackrisson · 6 years ago
  74. e27ccf9 Revert "in WebrtcVoiceEngine allow to set TaskQueueFactory" by Amit Hilbuch · 6 years ago
  75. a39254d in WebrtcVoiceEngine allow to set TaskQueueFactory by Danil Chapovalov · 6 years ago
  76. 647d5e6 Increase the default maximum jitter buffer size to 200 packets. by Jakob Ivarsson · 6 years ago
  77. e7a5f7b Modifying MediaChannel to accept CopyOnWriteBuffer by value. by Amit Hilbuch · 6 years ago
  78. e25f595 Guard preferred_dscp with the network interface lock by Steve Anton · 6 years ago
  79. 7ea4605 Add latency to remote source api. by Ruslan Burakov · 6 years ago
  80. d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 6 years ago
  81. 470a5ea Introduces common AudioAllocationSettings class. by Sebastian Jansson · 6 years ago
  82. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  83. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from media/engine/webrtcvoiceengine.h]
  84. 53eae87 Add PeerConnection option to enable RTX handling in the audio jitter buffer. by Jakob Ivarsson · 6 years ago
  85. 10403ae Add PeerConnection option to configure minimum audio jitter buffer delay. by Jakob Ivarsson · 6 years ago
  86. 84848f2 Adds interfaces for audio and video engines. by Sebastian Jansson · 6 years ago
  87. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  88. e693381 Delete struct rtc::PacketTime. by Niels Möller · 6 years ago
  89. 436ebca Fix extra setdscp call introduced by bad merge. by Tim Haloun · 6 years ago
  90. 648d28a Media engine and channel support for per-channel dscp values, specified by RtpParameter by Tim Haloun · 6 years ago
  91. bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
  92. 0378997 Adds flags indicating presence in allocation and feedback per packet. by Sebastian Jansson · 6 years ago
  93. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  94. 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
  95. 892acf0 Add support for send_encodings parameters in addTransceiver by Florent Castelli · 7 years ago
  96. cc22f51 Removing the intelligibility enhancer. by Alessio Bazzica · 7 years ago
  97. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  98. 00c71836 Replace rtc::Optional with absl::optional in media, ortc, p2p by Danil Chapovalov · 7 years ago
  99. 6e641e6 Signal detailed packet info for each packet sent. by Qingsi Wang · 7 years ago
  100. 5897a6e Adds support for signaling a=msid lines without a=ssrc lines. by Seth Hampson · 7 years ago