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webrtc / src / 50e2054c5b33f4101c50d2ca8948ad4693b6888a / audio
  1. c941579 Move field trial check WebRTC-DisableRtxRateLimiter by Danil Chapovalov · 1 year, 5 months ago
  2. 0505115 Pass the correct abs_capture_timestamp while cloning audio frame by Palak Agarwal · 1 year, 6 months ago
  3. c951d1b audio: fix some typos by Alfred E. Heggestad · 1 year, 6 months ago
  4. ad12dc5 Change ChannelReceive::GetAudioFrameWithInfo to use new Converts method by Olov Brändström · 1 year, 6 months ago
  5. 156facb change from unsigned to signed function (since offset can be negative) by Olov Brändström · 1 year, 6 months ago
  6. 4c55621 Cleanup RTPSenderAudio::SendAudio by Danil Chapovalov · 1 year, 6 months ago
  7. 36500ab Move RTPTimestamp offset handling out of encoded transform delegate by Tony Herre · 1 year, 7 months ago
  8. 14e5d4c Support sending IncomingFrames in audio by Palak Agarwal · 1 year, 7 months ago
  9. f263e1e Support receiving cloned encoded audio frames by Palak Agarwal · 1 year, 7 months ago
  10. 392e471 Remove deprecated TransformableAudioFrameInterface::getHeader() method by Tony Herre · 1 year, 7 months ago
  11. d43af91 Remove internal overrides using old SendRtp and SendRtcp interfaces. by Harald Alvestrand · 1 year, 7 months ago
  12. 48a2af3 Connected jitter_buffer_min_delay_ms to DelayManager's min_delay_ms by setting the neteq_config by anurag · 1 year, 9 months ago
  13. fc68f1f Stop using TransformableAudioFrameInterface::GetHeader() within webrtc by Tony Herre · 1 year, 9 months ago
  14. c4e0254 Fix capture_clock_offset_updater_ data race. by Jeremy Leconte · 1 year, 9 months ago
  15. 097a4de Make all encodedaudioframes inherit from TransformableAudioFrameI'face by Tony Herre · 1 year, 9 months ago
  16. 48c44e3 Ensure RtpSenderEgress run on worker queue by Per K · 1 year, 9 months ago
  17. fc260a18 Add method SetTimestamp in TransformableAudioFrameInterface by Palak Agarwal · 1 year, 9 months ago
  18. 54e95bc Propagate time of the last received packet with Timestamp type by Danil Chapovalov · 1 year, 10 months ago
  19. 3d6e88e Remove low_bandwidth_audio_test. by Jeremy Leconte · 1 year, 10 months ago
  20. 3e39254 Pass rtcp message to RtpTransportController through newer interface by Danil Chapovalov · 1 year, 10 months ago
  21. a2cf8ee Simplify handling rtcp messages in audio send channel by Danil Chapovalov · 1 year, 10 months ago
  22. 8095d02 Add RtpRtcpInterface::LastRtt function to replace RtpRtcpInterface::RTT by Danil Chapovalov · 1 year, 10 months ago
  23. 00ff2bb Cleanup usasge of ReportBlockData::report_block accessor in audio by Danil Chapovalov · 1 year, 10 months ago
  24. a9b9d4e Delete audio specific struct ReportBlock in favor of ReportBlockData by Danil Chapovalov · 1 year, 11 months ago
  25. 8a9f3a8 Reland "Remove dependency of video_replay on TestADM." by Artem Titov · 1 year, 11 months ago
  26. cde4b67 [SourceTracker] Move state to the worker thread, remove mutex. by Tommi · 1 year, 11 months ago
  27. f9e3bdd Revert "Remove dependency of video_replay on TestADM." by Jeremy Leconte · 1 year, 11 months ago
  28. 6a7bf10 Replace "rcvd" with "received" for readability by Philipp Hancke · 1 year, 11 months ago
  29. 0171666 Remove dependency of video_replay on TestADM. by Artem Titov · 1 year, 11 months ago
  30. eba7cee Extract TestADM into a separate target by Artem Titov · 1 year, 11 months ago
  31. fb8e3de Use AudioDeviceModule instead of TestAudioDeviceModule. by Artem Titov · 1 year, 11 months ago
  32. ec2670e Cleanup ReportBlockData class: use Timestamp and TimeDelta by Danil Chapovalov · 1 year, 11 months ago
  33. 50b0a76 Reland "[WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream" by Per Kjellander · 1 year, 11 months ago
  34. 73f048d Revert "[WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream" by Tomas Gunnarsson · 1 year, 11 months ago
  35. dd557fd [WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream by Per K · 1 year, 11 months ago
  36. 40a0e31 Remove AudioConfig::Mode. by Jeremy Leconte · 2 years ago
  37. c848268 Use SequenceChecker(SequenceChecker::kDetached) in a few places. by Tommi · 2 years ago
  38. b3e5969 stats: use uint64_t for RTCSentRtpStreamStats.packetsSent by Philipp Hancke · 2 years ago
  39. 0c126ed De-flake NonSenderRttStats and make it faster to run on average. by Henrik Boström · 2 years ago
  40. 1e2d951 Add a clone method to the audio frame transformer API. by Tove Petersson · 2 years ago
  41. a1ceae2 Implement support for Chrome task origin tracing. #3.5/4 by Markus Handell · 2 years ago
  42. 9f39721 Delete RtpRtcpInterface::RemoteNtp as redundant to GetSenderReportStats by Danil Chapovalov · 2 years ago
  43. 95d12ad Create unit test for the population of capture_start_ntp_time by Harald Alvestrand · 2 years, 1 month ago
  44. ba846cc Add a test that shows when channel_receive fires RR by Harald Alvestrand · 2 years, 1 month ago
  45. 84f7569 Break apart AudioCodingModule and AcmReceiver by Henrik Lundin · 2 years, 1 month ago
  46. 1f206b8 Use ArrayView in the IncomingRtcpPacket function. by Harald Alvestrand · 2 years, 1 month ago
  47. 217b384 Remove rtp header extension from config of Call audio and video receivers by Per K · 2 years, 1 month ago
  48. db20831 Update RTP timestamp based on capture timestamp when audio send stream is resumed. by Jakob Ivarsson · 2 years, 2 months ago
  49. dcb09ff Reset encoder when audio send stream is stopped. by Jakob Ivarsson · 2 years, 2 months ago
  50. 73e0cc8 Delete unused Audio Bwe integration test. by Per K · 2 years, 2 months ago
  51. e15b9ff Add a basic unittest for webrtc::voe::ChannelReceive by Harald Alvestrand · 2 years, 2 months ago
  52. 22821de Make capture timestamp optional in ADM. by Jakob Ivarsson · 2 years, 2 months ago
  53. 89870ff Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" by Per Kjellander · 2 years, 2 months ago
  54. 3e61f88 Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" by Per Kjellander · 2 years, 2 months ago
  55. 9ece54f Delete unnecssary AudioReceiveStreamInterface::GetRtpExtensions by Per K · 2 years, 2 months ago
  56. 3b96f2c Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp by Per K · 2 years, 2 months ago
  57. 478f3b7 Avoid waking up encoder thread when audio send stream is stopped. by Jakob Ivarsson · 2 years, 2 months ago
  58. 612872b2 Add RtcEvent to store when MinimumSetDelay is set on NetEq by Lionel Koenig · 2 years, 2 months ago
  59. 57e5562 [Unwrap] Use RtpTimestampUnwrapper in audio/channel_receive by Evan Shrubsole · 2 years, 2 months ago
  60. 6d5fa00 Flush buffers when stopping audio receive stream. by Jakob Ivarsson · 2 years, 2 months ago
  61. 8267cf3 [Unwrap] Use RtpTimestampUnwrapper in audio_ingress by Evan Shrubsole · 2 years, 2 months ago
  62. 9253240 Reland "Remove use of ReceiveStreamRtpConfig:transport_cc" by Per K · 2 years, 2 months ago
  63. be5c713 Revert "Remove use of ReceiveStreamRtpConfig:transport_cc" by Olga Sharonova · 2 years, 2 months ago
  64. 97ba853 Remove use of ReceiveStreamRtpConfig:transport_cc by Per K · 2 years, 2 months ago
  65. 68a7c41 Revert "Enforce stream id uniqueness in RtpSender::set_stream_ids" by Ilya Nikolaevskiy · 2 years, 3 months ago
  66. 315b95c Enforce stream id uniqueness in RtpSender::set_stream_ids by Philipp Hancke · 2 years, 3 months ago
  67. 794d599 Split media_channel and its dependencies from the rtc_media_base target by Harald Alvestrand · 2 years, 3 months ago
  68. 1b11b58 Remove pending packets from the pacer when an RTP module is removed. by Erik Språng · 2 years, 3 months ago
  69. e0b4cab Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead by Per Kjellander · 2 years, 3 months ago
  70. f0c33c4 Ensure audio quality tools are downloaded on Fuchsia by Christoffer Jansson · 2 years, 3 months ago
  71. acabb36 pc: Add asynchronous RtpSender::SetParameters() call by Florent Castelli · 2 years, 4 months ago
  72. a3e51df Add a new PeerConnectionE2EQualityTestFixture::AddPeer method. by Jeremy Leconte · 2 years, 4 months ago
  73. af51228 audio: make packets lost a signed integer by Philipp Hancke · 2 years, 4 months ago
  74. aebba7b [Stats] Expose totalPacketSendDelay for audio as well. by Henrik Boström · 2 years, 5 months ago
  75. 15dfc5a Add GetContributionSources to TransformableIncomingAudioFrame by Sergio Garcia Murillo · 2 years, 5 months ago
  76. 828ef91 Replace TaskQueue with MaybeWorkerThread in RtpTransportControllerInterface by Per Kjellander · 2 years, 5 months ago
  77. 9d9c2d5 Make header files self contained. by Mirko Bonadei · 2 years, 5 months ago
  78. 8a31b75 More audio stack traces by Olga Sharonova · 2 years, 5 months ago
  79. 2d0ba28 Audio stack traces by Olga Sharonova · 2 years, 6 months ago
  80. 718d7b3 Add missing export to the perf output file by Artem Titov · 2 years, 6 months ago
  81. e2f2cae Cleanup: Deduplicate static functions that create network links by Byoungchan Lee · 2 years, 6 months ago
  82. c45f4e4 [PCLF] Fully switch to new metrics export API by Artem Titov · 2 years, 6 months ago
  83. 56b96ffe Surface `local_capture_clock_offset` from `RtpSource` by Alessio Bazzica · 2 years, 6 months ago
  84. 53e5e28 Replace `ChannelReceive::GetRTT()` with `ModuleRtpRtcpImpl2::RTT()` by Alessio Bazzica · 2 years, 6 months ago
  85. 9e09a1f Replace Thread::Invoke with Thread::BlockingCall by Danil Chapovalov · 2 years, 6 months ago
  86. c92338a Remove `CallReceiveStatistics::rttMs` by Alessio Bazzica · 2 years, 6 months ago
  87. 7cc631e8 Add alessiob@webrtc.org in audio/OWNERS by Alessio Bazzica · 2 years, 6 months ago
  88. 2cfc1af Update rtc::Event::Wait call sites to use TimeDelta. by Markus Handell · 2 years, 7 months ago
  89. 0cf140d Rewrite AudioState null poller to use TaskQueueBase interface by Danil Chapovalov · 2 years, 7 months ago
  90. f4f2287 CallTest: migrate timeouts to TimeDelta. by Markus Handell · 2 years, 7 months ago
  91. e519f38 Remove rtc::Location from SendTask test helper by Danil Chapovalov · 2 years, 7 months ago
  92. 3bd444f Clarify and extend test support for certain sample rates in audio processing by Sam Zackrisson · 2 years, 7 months ago
  93. 6e7c268 Allow recursive check for RTC_DCHECK_RUN_ON macro by Danil Chapovalov · 2 years, 8 months ago
  94. 1a84b56 Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay by Ivo Creusen · 2 years, 8 months ago
  95. c05a1be Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable by Danil Chapovalov · 2 years, 8 months ago
  96. 253f36f Delete rtp_sender_ check in ModuleRtpRtcpImpl2::SetSendingMediaStatus by Niels Möller · 2 years, 8 months ago
  97. ee3ad9f Make ChannelSend::OnUplinkPacketLossRate public by Niels Möller · 2 years, 8 months ago
  98. d78789e Delete old TODOs. by Niels Möller · 2 years, 8 months ago
  99. aeb4412 Video and flexfec receive stream config changes without recreate. by Tommi · 2 years, 8 months ago
  100. 6939f63 Update old TODO comments by Niels Möller · 2 years, 8 months ago