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webrtc / src / 07f90d914c35ef8da0ac85b6e9107e8485ab9b66 / audio / channel_send.cc
  1. ec38238 Ensure the AudioCodingModule is reset when sending is stopped. by Lionel Koenig · 6 months ago
  2. 8037fc6 Migrate absl::optional to std::optional by Florent Castelli · 6 months ago
  3. 24366b0 Propagate Environment to audio RtpRtcp modules by Danil Chapovalov · 6 months ago
  4. f009e38 Fix AudioSendStream reconfigure - stop processing during unconfigured state by Guy Hershenbaum · 7 months ago
  5. 2406aaf Add accounting of actual audio bit usage by Dan Tan · 7 months ago
  6. 77ffbd3 Include-what-you-use api/rtc_event_log/ by Björn Terelius · 9 months ago
  7. c157f29 Pass Environment into audio ChannelSend by Danil Chapovalov · 10 months ago
  8. 8d07046 Pass the absolute capture timestamp to rtcp by Lionel Koenig · 10 months ago
  9. 64437e8 Calculate the audio level of audio packets before encoded transforms by Tony Herre · 11 months ago
  10. 3703b35 Using Ntp times for the absolute send time. by Jesús de Vicente Peña · 11 months ago
  11. 02af840 PacketRouter directly notify RtpTransportControllerSender when sending by Per K · 12 months ago
  12. 4a97488 Rename AudioLevel to AudioLevelExtension in rtp_header_extensions.h by Joachim Reiersen · 1 year, 1 month ago
  13. b1799b0 Cleanup usage of the rtc::TaskQueue in audio/ by Danil Chapovalov · 1 year, 2 months ago
  14. 0f1b9a9 Replace rtc::TaskQueue* with TaskQueueBase* in audio channel send frame transformer by Danil Chapovalov · 1 year, 2 months ago
  15. 5f3ac43 Ensure cloning and then sending audio encoded frames propagates CSRCs by Tony Herre · 1 year, 3 months ago
  16. d209893 Expose audio mimeType for insertable streams by Philipp Hancke · 1 year, 4 months ago
  17. f8feedf Make field trial string DisableRtxRateLimiter enabled by default. by Ying Wang · 1 year, 5 months ago
  18. c941579 Move field trial check WebRTC-DisableRtxRateLimiter by Danil Chapovalov · 1 year, 5 months ago
  19. 4c55621 Cleanup RTPSenderAudio::SendAudio by Danil Chapovalov · 1 year, 6 months ago
  20. 36500ab Move RTPTimestamp offset handling out of encoded transform delegate by Tony Herre · 1 year, 7 months ago
  21. 3e39254 Pass rtcp message to RtpTransportController through newer interface by Danil Chapovalov · 1 year, 10 months ago
  22. a2cf8ee Simplify handling rtcp messages in audio send channel by Danil Chapovalov · 1 year, 10 months ago
  23. a9b9d4e Delete audio specific struct ReportBlock in favor of ReportBlockData by Danil Chapovalov · 1 year, 10 months ago
  24. 6a7bf10 Replace "rcvd" with "received" for readability by Philipp Hancke · 1 year, 11 months ago
  25. ec2670e Cleanup ReportBlockData class: use Timestamp and TimeDelta by Danil Chapovalov · 1 year, 11 months ago
  26. 84f7569 Break apart AudioCodingModule and AcmReceiver by Henrik Lundin · 2 years, 1 month ago
  27. 1f206b8 Use ArrayView in the IncomingRtcpPacket function. by Harald Alvestrand · 2 years, 1 month ago
  28. db20831 Update RTP timestamp based on capture timestamp when audio send stream is resumed. by Jakob Ivarsson · 2 years, 2 months ago
  29. dcb09ff Reset encoder when audio send stream is stopped. by Jakob Ivarsson · 2 years, 2 months ago
  30. 478f3b7 Avoid waking up encoder thread when audio send stream is stopped. by Jakob Ivarsson · 2 years, 2 months ago
  31. 1b11b58 Remove pending packets from the pacer when an RTP module is removed. by Erik Språng · 2 years, 3 months ago
  32. aebba7b [Stats] Expose totalPacketSendDelay for audio as well. by Henrik Boström · 2 years, 5 months ago
  33. 2d0ba28 Audio stack traces by Olga Sharonova · 2 years, 6 months ago
  34. 0cf140d Rewrite AudioState null poller to use TaskQueueBase interface by Danil Chapovalov · 2 years, 7 months ago
  35. ee3ad9f Make ChannelSend::OnUplinkPacketLossRate public by Niels Möller · 2 years, 8 months ago
  36. 6939f63 Update old TODO comments by Niels Möller · 2 years, 8 months ago
  37. ea1e6f4 Delete rtc_base/format_macros.h by Niels Möller · 2 years, 10 months ago
  38. 8a1a0af In audio ChannelSend move task queue as last class member by Danil Chapovalov · 2 years, 11 months ago
  39. e62c2f2 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf by Jonas Oreland · 3 years ago
  40. a943e73 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 7/inf by Jonas Oreland · 3 years ago
  41. 7336422 Delete some unneeded references to ProcessThread. by Niels Möller · 3 years, 2 months ago
  42. bf08745 Implement RTCOutboundRtpStreamStats.targetBitrate for audio. by Jakob Ivarsson · 3 years, 4 months ago
  43. d0321c5 Deduplicate set of the rtp header extension uri constants by Danil Chapovalov · 3 years, 6 months ago
  44. ac09f0d Remove last traces of deferred sequencing. by Erik Språng · 3 years, 7 months ago
  45. 69dd142 Register audio send stream in packet router on Start(). by Erik Språng · 3 years, 7 months ago
  46. 2373bb9 Default-enable deferred sequence numbering for audio. by Erik Språng · 3 years, 7 months ago
  47. e91c992 Implement nack_count metric for outbound audio rtp streams. by Jakob Ivarsson · 3 years, 8 months ago
  48. eb61b7f ModuleRtcRtcpImpl2: remove Module inheritance. by Markus Handell · 3 years, 9 months ago
  49. c1d5891 Replace `new rtc::RefCountedObject` with `rtc::make_ref_counted` in a few files by Tomas Gunnarsson · 3 years, 11 months ago
  50. 5eda59c Replace legacy RtpRtcp::GetRemoteStat function with GetLatestReportBlockData by Danil Chapovalov · 4 years ago
  51. d15a575 Use SequenceChecker from public API by Artem Titov · 4 years, 1 month ago
  52. c8421c4 Replace rtc::ThreadChecker with webrtc::SequenceChecker by Artem Titov · 4 years, 1 month ago
  53. 49dbad0 Fixing audio timestamp stall during inactivation (under a kill switch) by Minyue Li · 4 years, 2 months ago
  54. 2accc7d Revert "Add task queue to RtpRtcpInterface::Configuration." by Niels Moller · 4 years, 2 months ago
  55. f23e214 Add task queue to RtpRtcpInterface::Configuration. by Niels Möller · 4 years, 2 months ago
  56. 6287280 Migrate audio/ to use webrtc::Mutex by Markus Handell · 4 years, 8 months ago
  57. 2b4d2f3 Removes locking in TransportFeedbackProxy. by Erik Språng · 4 years, 9 months ago
  58. f25761d Remove dependency from RtpRtcp on the Module interface. by Tomas Gunnarsson · 4 years, 9 months ago
  59. fae0562 Deprecate the static RtpRtcp::Create() method. by Tomas Gunnarsson · 4 years, 9 months ago
  60. 04e1bab Replaces OverheadObserver with simple getter. by Erik Språng · 4 years, 10 months ago
  61. b9d4685 insertable streams: include rtp_timestamp offset for audio by Philipp Hancke · 5 years ago
  62. 65674d8 Transform encoded frames in ChannelSend. by Marina Ciocea · 5 years ago
  63. d2aa8f9 Insert audio frame transformer between encoder and packetizer. by Marina Ciocea · 5 years ago
  64. ff0e4db Reland "Send absolute capture time through audio coding module." by Minyue Li · 5 years ago
  65. 4175914 Revert "Send absolute capture time through audio coding module." by Minyue Li · 5 years ago
  66. 48655cf Send absolute capture time through audio coding module. by Minyue Li · 5 years ago
  67. 6298b56 Cleanup: Using RtpRtcp directly from AudioSendStream by Sebastian Jansson · 5 years ago
  68. b2b2031 Concatenate string literals at compile time. by Jonas Olsson · 5 years ago
  69. f2c0818 Minor fixes to ChannelSend. by Mirko Bonadei · 5 years ago
  70. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 5 years ago
  71. cd2a92f Removes RPLR based FEC controller. by Sebastian Jansson · 5 years ago
  72. f39c815 Cleanup: Replacing set extension status bool with CHECK. by Sebastian Jansson · 5 years ago
  73. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
  74. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
  75. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
  76. ea55b08 Adds support for passing a vector of packets to the paced sender. by Erik Språng · 5 years ago
  77. ee5ec9a Replacing local closure classes with C++14 moving capture lambdas. by Sebastian Jansson · 5 years ago
  78. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
  79. b6220d9 Delete unused logic for audio RtcpMode::kOff by Niels Möller · 6 years ago
  80. fac7e31 Removes TransportSequenceNumberAllocator by Erik Språng · 6 years ago
  81. 4208a13 Removes deprecated InsertPacket/TimeToSendPacket/TimeToSendPadding by Erik Språng · 6 years ago
  82. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 6 years ago
  83. 54d5d2c Rename RtpRtcp::Configuration::media_send_ssrc to local_media_ssrc by Erik Språng · 6 years ago
  84. 83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 6 years ago
  85. da4f093 Reland "Only include payload in bytes sent/received." by Bjorn A Mellem · 6 years ago
  86. bcd068d Revert "Only include payload in bytes sent/received." by Bjorn Mellem · 6 years ago
  87. aa59eca Move RtpPacketSender and merge it with RtpPacketPacer. by Erik Språng · 6 years ago
  88. 74a1b4b Only include payload in bytes sent/received. by Bjorn A Mellem · 6 years ago
  89. 4c2c412 Set local ssrc at construction (audio) by Erik Språng · 6 years ago
  90. d0679bd Enables usage of ChannelMixer in WebRTC's output mixer. by henrika · 6 years ago
  91. f48bca7 Avoid triggering a false error logging when using encryptor and sending DTX. by Minyue Li · 6 years ago
  92. 59b8654 Switch from RtpPacketSender to RtpPacketPacer interface usage. by Erik Språng · 6 years ago
  93. 9ab520e Reland "Avoid encrypting empty audio packet." by Minyue Li · 6 years ago
  94. 6e436d1 [audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 6 years ago
  95. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 6 years ago
  96. d703cd0 Revert "Avoid encrypting empty audio packet." by Minyue Li · 6 years ago
  97. b0ac943 Avoid encrypting empty audio packet. by Minyue Li · 6 years ago
  98. c35b6e6 Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData by Niels Möller · 6 years ago
  99. 30a276b Add RTP sequence number to TransportFeedbackObserver::AddPacket() by Erik Språng · 6 years ago
  100. cf96e0f Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent. by Henrik Boström · 6 years ago