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gerrit-public.fairphone.software / platform / external / webrtc / 27c293665e3a2b0e2de12236529773a7b9057f6e / media / engine / webrtc_voice_engine.cc
  1. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 5 years ago
  2. 03fbace Remove apm_helpers, consolidate audio config in WebRtcVoiceEngine by Sam Zackrisson · 5 years ago
  3. 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 5 years ago
  4. 0bad15f Remove the noise_suppression() pointer to submodule interface by saza · 5 years ago
  5. 8038541 Update the header extensions capabilities with mid, rid and rrid by Florent Castelli · 5 years ago
  6. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
  7. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
  8. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
  9. ff27da5 Add/remove receive streams with SSRC 0 from media channels by Saurav Das · 5 years ago
  10. 1b83a9e Only handle each RTCP once. by Sebastian Jansson · 5 years ago
  11. 53227cc Remove webrtc::MinPositive from api/. by Mirko Bonadei · 5 years ago
  12. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 5 years ago
  13. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
  14. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 5 years ago
  15. bbeb109 Reporting audio device underrun counter by Alex Narest · 5 years ago
  16. 71c6b56 Allow sending abs-send-time for audio streams. by Sebastian Jansson · 5 years ago
  17. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 5 years ago
  18. f40a340 Remove deprecated code related to AEC2 by Per Åhgren · 5 years ago
  19. d2845f8 Removes unused AudioAllocationSettings from voice engine. by Sebastian Jansson · 5 years ago
  20. bedb7a8 Revert "Reporting of decoding_codec_plc events" by Mirko Bonadei · 5 years ago
  21. 0a88ea0 Reporting of decoding_codec_plc events by Alex Narest · 5 years ago
  22. 3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 5 years ago
  23. e8e4dc4 Change StartAecDump methods to work with FILE* and FileWrapper by Niels Möller · 5 years ago
  24. 220f4be Remove some media/ --> pc/ test dependencies by Steve Anton · 5 years ago
  25. 6e436d1 [audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 5 years ago
  26. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 5 years ago
  27. 9356252 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 5 years ago
  28. 8d8ffdb Expose new audio stats on the API by Ivo Creusen · 5 years ago
  29. 44125fa Reland "Piping audio interruption metrics to API layer" by Henrik Lundin · 5 years ago
  30. fc02a79 Revert "Piping audio interruption metrics to API layer" by Henrik Andreassson · 5 years ago
  31. 8a9778e Delete unused StartAecDump method with filename argument by Niels Möller · 5 years ago
  32. 299c4e6 Piping audio interruption metrics to API layer by Henrik Lundin · 5 years ago
  33. 63658d0 Revert "Ensure that we always set values for min and max audio bitrate." by Daniel Lee · 6 years ago
  34. e47aee3 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 6 years ago
  35. cf96e0f Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent. by Henrik Boström · 6 years ago
  36. 01738c6 Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. by Henrik Boström · 6 years ago
  37. c01367d Deprecating ThreadChecker specific interface. by Sebastian Jansson · 6 years ago
  38. 1c41be6 Propagate TaskQueueFactory to AudioDeviceBuffer by Danil Chapovalov · 6 years ago
  39. 4c7112a Reland "in WebrtcVoiceEngine allow to set TaskQueueFactory" by Danil Chapovalov · 6 years ago
  40. f0d1c03 Add replacement interface for webrtc::GainConrol by Sam Zackrisson · 6 years ago
  41. e27ccf9 Revert "in WebrtcVoiceEngine allow to set TaskQueueFactory" by Amit Hilbuch · 6 years ago
  42. a39254d in WebrtcVoiceEngine allow to set TaskQueueFactory by Danil Chapovalov · 6 years ago
  43. 647d5e6 Increase the default maximum jitter buffer size to 200 packets. by Jakob Ivarsson · 6 years ago
  44. e7a5f7b Modifying MediaChannel to accept CopyOnWriteBuffer by value. by Amit Hilbuch · 6 years ago
  45. e25f595 Guard preferred_dscp with the network interface lock by Steve Anton · 6 years ago
  46. 232b3fd Expose relative packet arrival delay metric in stats API. by Jakob Ivarsson · 6 years ago
  47. 7ea4605 Add latency to remote source api. by Ruslan Burakov · 6 years ago
  48. 914351d Reland "Always offer transport sequence number header extension for audio"" by Per Kjellander · 6 years ago
  49. 397c06f Revert "Always offer transport sequence number header extension for audio" by Ying Wang · 6 years ago
  50. fd965c0 Always offer transport sequence number header extension for audio by Per Kjellander · 6 years ago
  51. 157540a Stop hard-coding default IDs for RTP extensions by Elad Alon · 6 years ago
  52. e1dcce2 Remove HAVE_WEBRTC_VOICE. by Fredrik Solenberg · 6 years ago
  53. c1a0bcb Implement the encoding RtpParameter scaleResolutionDownBy by Florent Castelli · 6 years ago
  54. 2c9ebef Use Abseil container algorithms in media/ by Steve Anton · 6 years ago
  55. 470a5ea Introduces common AudioAllocationSettings class. by Sebastian Jansson · 6 years ago
  56. 0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 6 years ago
  57. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  58. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from media/engine/webrtcvoiceengine.cc]
  59. ba50223 Make voiceengine/audio transport stop using voice_detection() interface by Sam Zackrisson · 6 years ago
  60. 53eae87 Add PeerConnection option to enable RTX handling in the audio jitter buffer. by Jakob Ivarsson · 6 years ago
  61. 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
  62. e1301a8 Revert "Implement read-only codecPayloadType in RtpParameters" by Henrik Grunell · 6 years ago
  63. 806e06d Implement read-only codecPayloadType in RtpParameters by Florent Castelli · 6 years ago
  64. 10403ae Add PeerConnection option to configure minimum audio jitter buffer delay. by Jakob Ivarsson · 6 years ago
  65. 352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
  66. 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
  67. c69a56e Remove more unneeded things from ChannelSend by Fredrik Solenberg · 6 years ago
  68. 84848f2 Adds interfaces for audio and video engines. by Sebastian Jansson · 6 years ago
  69. 8544799 Introduce DLOG to video and voiceengine. by Jonas Olsson · 6 years ago
  70. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  71. e693381 Delete struct rtc::PacketTime. by Niels Möller · 6 years ago
  72. 15ca5a9 Add implicit conversion between rtc:PacketTime and int64_t. by Niels Möller · 6 years ago
  73. 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
  74. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  75. 2edab4c Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. by Niels Möller · 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. 3c7d599 Replace _stricmp with absl::EqualsIgnoreCase by Niels Möller · 6 years ago
  78. bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
  79. be65d48 Remove AECM comfort noise setting from API by Sam Zackrisson · 6 years ago
  80. 26968ba Delete unused utf8 conversion utilities by Niels Möller · 6 years ago
  81. 20a49f3 Don't try to use CN if voice codec isn't mono by Karl Wiberg · 6 years ago
  82. 64be7fa Move FecController to RtpVideoSender. by Stefan Holmer · 6 years ago
  83. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  84. 892acf0 Add support for send_encodings parameters in addTransceiver by Florent Castelli · 6 years ago
  85. 7988e5c Remove echo_cancellation() and echo_control_mobile() interface access outside APM by Sam Zackrisson · 6 years ago
  86. 84df1c7 Make fewer copies when using StringBuilder. by Jonas Olsson · 6 years ago
  87. 366a50c Remove simple stringstream usages. by Jonas Olsson · 6 years ago
  88. cc22f51 Removing the intelligibility enhancer. by Alessio Bazzica · 6 years ago
  89. 7008287 Delete struct webrtc::PacketTime. by Niels Möller · 6 years ago
  90. a76af0c Move base64.h to the proper location. by Artem Titov · 6 years ago
  91. bcf9180 Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream. by Alex Narest · 6 years ago
  92. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  93. 00c7183 Replace rtc::Optional with absl::optional in media, ortc, p2p by Danil Chapovalov · 6 years ago
  94. abe301f Add HeaderExtensions to RtpParameters by Florent Castelli · 6 years ago
  95. dacec71 Add Rtcp parameters for PeerConnection senders by Florent Castelli · 6 years ago
  96. 5897a6e Adds support for signaling a=msid lines without a=ssrc lines. by Seth Hampson · 7 years ago
  97. abbe841 This CL removes all usages of our custom ostream << overloads. by Jonas Olsson · 7 years ago
  98. bb50ce5 Wire up MID send value to the PeerConnection API by Steve Anton · 7 years ago
  99. 77490b9 Pass a real audio codec pair ID to encoders that we create by Karl Wiberg · 7 years ago
  100. 0812634 Pass a real audio codec pair ID to decoders that we create by Karl Wiberg · 7 years ago