Google Git
Sign in
webrtc / src / a9e1b49704b233cf660cc6670ceaef8ef89f8bb0 / audio / audio_send_stream.cc
  1. 6287280 Migrate audio/ to use webrtc::Mutex by Markus Handell · 4 years, 8 months ago
  2. 2b4d2f3 Removes locking in TransportFeedbackProxy. by Erik Språng · 4 years, 8 months ago
  3. edcd966 negotiate RED codec for audio by Philipp Hancke · 4 years, 9 months ago
  4. 1a49756 fix typos in comments by Philipp Hancke · 4 years, 10 months ago
  5. cf6544a Avoids unnecessary calls to audio encoder. by Erik Språng · 4 years, 10 months ago
  6. 04e1bab Replaces OverheadObserver with simple getter. by Erik Språng · 4 years, 10 months ago
  7. 9abc6bd Reduce audiosendstream dependency on rttstats (dead code). by Tommi · 4 years, 11 months ago
  8. cc73ed3 APM: Add build flag to allow building WebRTC without APM by Per Åhgren · 4 years, 11 months ago
  9. d2aa8f9 Insert audio frame transformer between encoder and packetizer. by Marina Ciocea · 5 years ago
  10. 0c96449 Clamp stable target bitrate to min/max allocated bitrate. by Jakob Ivarsson · 5 years ago
  11. c771084 Remove RTC_NOTREACHED from audio_send_stream when ANA didn't work by Alejandro Luebs · 5 years ago
  12. 01ab084 Add minimum overhead to configured priorty bitrate instead of maximum. by Jakob Ivarsson · 5 years ago
  13. d14525e Make sure that the audio stream is allocated with the correct overhead. by Jakob Ivarsson · 5 years ago
  14. 74dadc1 Ready to support of absolute capture timestamp header extension. by Minyue Li · 5 years ago
  15. cad3e0e Replace DataSize and DataRate factories with newer versions by Danil Chapovalov · 5 years ago
  16. 0c626af Use newer version of TimeDelta and TimeStamp factories in webrtc by Danil Chapovalov · 5 years ago
  17. bef818d Default disables legacy overhead calculation. by Sebastian Jansson · 5 years ago
  18. c3eb9fd Reland "Reland "Only include overhead if using send side bandwidth estimation."" by Sebastian Jansson · 5 years ago
  19. 4356490 Revert "Reland "Only include overhead if using send side bandwidth estimation."" by Mirko Bonadei · 5 years ago
  20. 086055d Reland "Only include overhead if using send side bandwidth estimation." by Sebastian Jansson · 5 years ago
  21. c709412 Revert "Only include overhead if using send side bandwidth estimation." by Sebastian Jansson · 5 years ago
  22. 8c79c6e Only include overhead if using send side bandwidth estimation. by Sebastian Jansson · 5 years ago
  23. 6298b56 Cleanup: Using RtpRtcp directly from AudioSendStream by Sebastian Jansson · 5 years ago
  24. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 5 years ago
  25. cd2a92f Removes RPLR based FEC controller. by Sebastian Jansson · 5 years ago
  26. 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 5 years ago
  27. f39c815 Cleanup: Replacing set extension status bool with CHECK. by Sebastian Jansson · 5 years ago
  28. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
  29. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
  30. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
  31. cd0eedb Don't allocate audio if we have no transport sequence number. by Sebastian Jansson · 5 years ago
  32. 0a6510d Removes rtp_transport checks in AudioSendStream by Sebastian Jansson · 5 years ago
  33. 35cf9e7 Replaces static modifier functions in AudioSendStream. by Sebastian Jansson · 5 years ago
  34. 0429f78 Base overhead calculation for audio priority rate on available data. by Sebastian Jansson · 5 years ago
  35. f23131f Removing AudioAllocationSettings moving functionality to AudioSendStream. by Sebastian Jansson · 5 years ago
  36. 62aee93 Adds trial to calculate audio overhead based on available data. by Sebastian Jansson · 5 years ago
  37. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
  38. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 5 years ago
  39. f13df86 Delete audio methods SignalNetworkState by Niels Möller · 6 years ago
  40. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 6 years ago
  41. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 6 years ago
  42. 71c6b56 Allow sending abs-send-time for audio streams. by Sebastian Jansson · 6 years ago
  43. 83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 6 years ago
  44. 1704801 Prevent concurrent access to AudioSendStream's configuration. by Yves Gerey · 6 years ago
  45. aa59eca Move RtpPacketSender and merge it with RtpPacketPacer. by Erik Språng · 6 years ago
  46. 0182a03 Reland "Remove the injectable bitrate allocation strategy API." by Jonas Olsson · 6 years ago
  47. 4c2c412 Set local ssrc at construction (audio) by Erik Språng · 6 years ago
  48. e95b57c Revert "Remove the injectable bitrate allocation strategy API." by Mirko Bonadei · 6 years ago
  49. 80cb3f6 Remove the injectable bitrate allocation strategy API. by Jonas Olsson · 6 years ago
  50. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 6 years ago
  51. 6e436d1 [audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 6 years ago
  52. a352248 Add a config flag to disable the audio ALR probing request. by Christoffer Rodbro · 6 years ago
  53. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 6 years ago
  54. 8f119ca Enable experiments with audio bitrate priority. by Jonas Olsson · 6 years ago
  55. 9356252 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 6 years ago
  56. 413ccc4 Stop DCHECK which occurs in ANA BitrateController when overhead is zero. by Bjorn A Mellem · 6 years ago
  57. 63658d0 Revert "Ensure that we always set values for min and max audio bitrate." by Daniel Lee · 6 years ago
  58. e47aee3 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 6 years ago
  59. cf96e0f Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent. by Henrik Boström · 6 years ago
  60. c01367d Deprecating ThreadChecker specific interface. by Sebastian Jansson · 6 years ago
  61. 741daaf Move rtc::FunctionView to the public API by Artem Titov · 6 years ago
  62. 44dd9f2 Adds ChannelSend specific encoder task queue. by Sebastian Jansson · 6 years ago
  63. 0b69826 Don't inject worker queue into send streams. by Sebastian Jansson · 6 years ago
  64. 8672cac Trigger audio bitrate allocation update on overhead change. by Sebastian Jansson · 6 years ago
  65. ee5ccbc Move ownership of RTPSenderAudio to ChannelSend. by Niels Möller · 6 years ago
  66. 110c64b Delete unused key WebRTC-Audio-SendSideBwe-For-Video. by Christoffer Rodbro · 6 years ago
  67. 8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 6 years ago
  68. 977b335 Injecting Clock into audio streams. by Sebastian Jansson · 6 years ago
  69. 914351d Reland "Always offer transport sequence number header extension for audio"" by Per Kjellander · 6 years ago
  70. 397c06f Revert "Always offer transport sequence number header extension for audio" by Ying Wang · 6 years ago
  71. fd965c0 Always offer transport sequence number header extension for audio by Per Kjellander · 6 years ago
  72. 14a7cf9 Adds CallEncoder to ChannelSend. by Sebastian Jansson · 6 years ago
  73. 464a557 Adds audio priority bitrate field trial parameter. by Sebastian Jansson · 6 years ago
  74. 626015d Make AudioSendStream to be OverheadObserver by Anton Sukhanov · 6 years ago
  75. 470a5ea Introduces common AudioAllocationSettings class. by Sebastian Jansson · 6 years ago
  76. 79f0d4d Enables feature to account for unacknowledged data. by Sebastian Jansson · 6 years ago
  77. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  78. 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
  79. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
  80. e977199 Delete ChannelSend::RegisterTransport, replacing by construction argument by Niels Möller · 6 years ago
  81. ff05816 Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric by Sam Zackrisson · 6 years ago
  82. 254d869 Routing BitrateAllocationUpdate to audio codec. by Sebastian Jansson · 6 years ago
  83. 13e5903 Using unit classes in BitrateAllocationUpdate struct. by Sebastian Jansson · 6 years ago
  84. c69a56e Remove more unneeded things from ChannelSend by Fredrik Solenberg · 6 years ago
  85. 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
  86. dced9f6 Delete class ChannelSendProxy by Niels Möller · 6 years ago
  87. 645a3af Remove unused/unnecessary things from ChannelSend. by Fredrik Solenberg · 6 years ago
  88. 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
  89. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  90. c572ff3 Add default constructor for rtc::Event by Niels Möller · 6 years ago
  91. 2365936 Hide the AudioEncoderCng class behind a create function by Karl Wiberg · 6 years ago
  92. 56ef305 Move event logging of config into AudioSendStream. by Oskar Sundbom · 6 years ago
  93. 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
  94. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  95. 359d60a Adds target rate to audio send stream stats. by Sebastian Jansson · 6 years ago
  96. c0e4d45 Adds BitrateAllocation struct to OnBitrateUpdated. by Sebastian Jansson · 6 years ago
  97. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  98. 67b011d Use BitrateAllocatorInterface in AudioSendStream and VideoSendStream by Niels Möller · 6 years ago
  99. 648d28a Media engine and channel support for per-channel dscp values, specified by RtpParameter by Tim Haloun · 6 years ago
  100. bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago