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gerrit-public.fairphone.software / platform / external / webrtc / 876d1d3fed2b621fb63b0af0c0ffb4cb1b04f430 / media / engine / webrtc_voice_engine.cc
  1. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  2. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from media/engine/webrtcvoiceengine.cc]
  3. ba50223 Make voiceengine/audio transport stop using voice_detection() interface by Sam Zackrisson · 6 years ago
  4. 53eae87 Add PeerConnection option to enable RTX handling in the audio jitter buffer. by Jakob Ivarsson · 6 years ago
  5. 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
  6. e1301a8 Revert "Implement read-only codecPayloadType in RtpParameters" by Henrik Grunell · 6 years ago
  7. 806e06d Implement read-only codecPayloadType in RtpParameters by Florent Castelli · 6 years ago
  8. 10403ae Add PeerConnection option to configure minimum audio jitter buffer delay. by Jakob Ivarsson · 6 years ago
  9. 352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
  10. 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
  11. c69a56e Remove more unneeded things from ChannelSend by Fredrik Solenberg · 6 years ago
  12. 84848f2 Adds interfaces for audio and video engines. by Sebastian Jansson · 6 years ago
  13. 8544799 Introduce DLOG to video and voiceengine. by Jonas Olsson · 6 years ago
  14. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  15. e693381 Delete struct rtc::PacketTime. by Niels Möller · 6 years ago
  16. 15ca5a9 Add implicit conversion between rtc:PacketTime and int64_t. by Niels Möller · 6 years ago
  17. 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
  18. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  19. 2edab4c Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. by Niels Möller · 6 years ago
  20. 648d28a Media engine and channel support for per-channel dscp values, specified by RtpParameter by Tim Haloun · 6 years ago
  21. 3c7d599 Replace _stricmp with absl::EqualsIgnoreCase by Niels Möller · 6 years ago
  22. bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
  23. be65d48 Remove AECM comfort noise setting from API by Sam Zackrisson · 6 years ago
  24. 26968ba Delete unused utf8 conversion utilities by Niels Möller · 6 years ago
  25. 20a49f3 Don't try to use CN if voice codec isn't mono by Karl Wiberg · 6 years ago
  26. 64be7fa Move FecController to RtpVideoSender. by Stefan Holmer · 6 years ago
  27. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  28. 892acf0 Add support for send_encodings parameters in addTransceiver by Florent Castelli · 6 years ago
  29. 7988e5c Remove echo_cancellation() and echo_control_mobile() interface access outside APM by Sam Zackrisson · 6 years ago
  30. 84df1c7 Make fewer copies when using StringBuilder. by Jonas Olsson · 6 years ago
  31. 366a50c Remove simple stringstream usages. by Jonas Olsson · 6 years ago
  32. cc22f51 Removing the intelligibility enhancer. by Alessio Bazzica · 6 years ago
  33. 7008287 Delete struct webrtc::PacketTime. by Niels Möller · 6 years ago
  34. a76af0c Move base64.h to the proper location. by Artem Titov · 6 years ago
  35. 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
  36. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  37. 00c7183 Replace rtc::Optional with absl::optional in media, ortc, p2p by Danil Chapovalov · 6 years ago
  38. abe301f Add HeaderExtensions to RtpParameters by Florent Castelli · 6 years ago
  39. dacec71 Add Rtcp parameters for PeerConnection senders by Florent Castelli · 6 years ago
  40. 5897a6e Adds support for signaling a=msid lines without a=ssrc lines. by Seth Hampson · 6 years ago
  41. abbe841 This CL removes all usages of our custom ostream << overloads. by Jonas Olsson · 6 years ago
  42. bb50ce5 Wire up MID send value to the PeerConnection API by Steve Anton · 6 years ago
  43. 77490b9 Pass a real audio codec pair ID to encoders that we create by Karl Wiberg · 6 years ago
  44. 0812634 Pass a real audio codec pair ID to decoders that we create by Karl Wiberg · 7 years ago
  45. 4ccc1c4 Don't destroy a receive stream's sink before reassigning it. by Oskar Sundbom · 7 years ago
  46. ab1aee0 Reland "Deprecate the adaptive level controller" by Sam Zackrisson · 7 years ago
  47. 52f8188 Revert "Deprecate the adaptive level controller" by Sam Zackrisson · 7 years ago
  48. 6f37ed7 Deprecate the adaptive level controller by Sam Zackrisson · 7 years ago
  49. 845e878 Name change from stream label to stream id for spec compliance. by Seth Hampson · 7 years ago
  50. 5a26a3a Remove public sync_label from StreamParams by Steve Anton · 7 years ago
  51. c668108 Maintain audio receive stream gain across recreations by Oskar Sundbom · 7 years ago
  52. 8f83b42 Moved bitrate config interface from Call class. by Sebastian Jansson · 7 years ago
  53. fc8d26b Reland "Moved BitrateConfig out of Call::Config." by Sebastian Jansson · 7 years ago
  54. a1f6661 Check that channel is in "send" before OKing DTMF by Harald Alvestrand · 7 years ago
  55. e4bf600 Revert "Moved BitrateConfig out of Call::Config." by Lu Liu · 7 years ago
  56. 5897fe2 Moved BitrateConfig out of Call::Config. by Sebastian Jansson · 7 years ago
  57. f120cba Delete AudioMonitor and related code. by Niels Möller · 7 years ago
  58. ba37b4b Change return type of RtpSenderInterface::SetParameters from bool to RTCError by Zach Stein · 7 years ago
  59. 4613bdf Recreate AudioReceiveStreams when header extensions change. by Fredrik Solenberg · 7 years ago
  60. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  61. 3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. by Fredrik Solenberg · 7 years ago
  62. 24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
  63. 7622048 Add an AudioOptions field to force software echo cancellation on iOS. by Jonathan Yu · 7 years ago
  64. 8b77aea Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Lu Liu · 7 years ago
  65. d2b912a Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator. by Seth Hampson · 7 years ago
  66. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  67. fa266ef Fix the crash when GetSources is called with non-existing ssrc. by Zhi Huang · 7 years ago
  68. e26456a Removes usage of AGC APIs in the ADM. by henrika · 7 years ago
  69. 606a597 Remove adjust_agc_delta from WebRtcVoiceEngine by Steve Anton · 7 years ago
  70. 292a73e Deliver packet to Call as rtc::CopyOnWriteBuffer by Danil Chapovalov · 7 years ago
  71. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  72. 55900fd Move APM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  73. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  74. 7880758 Optional: Use nullopt and implicit construction in /media by Oskar Sundbom · 7 years ago
  75. c97cf03 Removes unused sample-rate APIs from the ADM. by henrika · 7 years ago
  76. 5f5918f Merge MediaChannel's OnTransportOverheadChanged and OnNetworkRouteChanged callbacks. by Zhi Huang · 7 years ago
  77. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  78. e78bcb9 Enable cpplint in media/ by Steve Anton · 7 years ago
  79. 6592f2c Removes more unused ADM APIs: by henrika · 7 years ago
  80. c8c71b9 Stop using ANDROID macro in favour of WEBRTC_ANDROID. by Mirko Bonadei · 7 years ago
  81. b3944f0 Media track ID visibility at BWE level by Alex Narest · 7 years ago
  82. 4332d09 Reland "Reland "Remove WEBRTC_TRACE."" by Fredrik Solenberg · 7 years ago
  83. 39cefdb Revert "Reland "Remove WEBRTC_TRACE."" by Fredrik Solenberg · 7 years ago
  84. 68007e9 Reland "Remove WEBRTC_TRACE." by Fredrik Solenberg · 7 years ago
  85. 729b910 Revert "Remove WEBRTC_TRACE." by Fredrik Solenberg · 7 years ago
  86. 2209b90 Remove WEBRTC_TRACE. by Fredrik Solenberg · 7 years ago
  87. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
  88. 4e2deab Only return stats for the most recent unsignaled audio stream. by deadbeef · 7 years ago
  89. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
  90. 35dee81 Clean out unused methods from VoiceEngine and VoEBase. by solenberg · 7 years ago
  91. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  92. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/media/engine/webrtcvoiceengine.cc]
  93. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
  94. 0e320ec Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 7 years ago
  95. 2dbc69f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 7 years ago
  96. b1c9d1d Avoid that previous settings in APM are overwritten by WebRtcVoiceEngine by peah · 7 years ago
  97. 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
  98. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  99. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  100. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago