Google Git
Sign in
webrtc / src.git / e9aa8675d74321aa06d870df008fc6b9690148af / audio
  1. b3e5969 stats: use uint64_t for RTCSentRtpStreamStats.packetsSent by Philipp Hancke · 2 years ago
  2. 0c126ed De-flake NonSenderRttStats and make it faster to run on average. by Henrik Boström · 2 years ago
  3. 1e2d951 Add a clone method to the audio frame transformer API. by Tove Petersson · 2 years, 1 month ago
  4. a1ceae2 Implement support for Chrome task origin tracing. #3.5/4 by Markus Handell · 2 years, 1 month ago
  5. 9f39721 Delete RtpRtcpInterface::RemoteNtp as redundant to GetSenderReportStats by Danil Chapovalov · 2 years, 1 month ago
  6. 95d12ad Create unit test for the population of capture_start_ntp_time by Harald Alvestrand · 2 years, 2 months ago
  7. ba846cc Add a test that shows when channel_receive fires RR by Harald Alvestrand · 2 years, 2 months ago
  8. 84f7569 Break apart AudioCodingModule and AcmReceiver by Henrik Lundin · 2 years, 2 months ago
  9. 1f206b8 Use ArrayView in the IncomingRtcpPacket function. by Harald Alvestrand · 2 years, 2 months ago
  10. 217b384 Remove rtp header extension from config of Call audio and video receivers by Per K · 2 years, 2 months ago
  11. db20831 Update RTP timestamp based on capture timestamp when audio send stream is resumed. by Jakob Ivarsson · 2 years, 2 months ago
  12. dcb09ff Reset encoder when audio send stream is stopped. by Jakob Ivarsson · 2 years, 2 months ago
  13. 73e0cc8 Delete unused Audio Bwe integration test. by Per K · 2 years, 2 months ago
  14. e15b9ff Add a basic unittest for webrtc::voe::ChannelReceive by Harald Alvestrand · 2 years, 2 months ago
  15. 22821de Make capture timestamp optional in ADM. by Jakob Ivarsson · 2 years, 2 months ago
  16. 89870ff Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" by Per Kjellander · 2 years, 2 months ago
  17. 3e61f88 Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" by Per Kjellander · 2 years, 2 months ago
  18. 9ece54f Delete unnecssary AudioReceiveStreamInterface::GetRtpExtensions by Per K · 2 years, 2 months ago
  19. 3b96f2c Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp by Per K · 2 years, 2 months ago
  20. 478f3b7 Avoid waking up encoder thread when audio send stream is stopped. by Jakob Ivarsson · 2 years, 2 months ago
  21. 612872b2 Add RtcEvent to store when MinimumSetDelay is set on NetEq by Lionel Koenig · 2 years, 2 months ago
  22. 57e5562 [Unwrap] Use RtpTimestampUnwrapper in audio/channel_receive by Evan Shrubsole · 2 years, 2 months ago
  23. 6d5fa00 Flush buffers when stopping audio receive stream. by Jakob Ivarsson · 2 years, 2 months ago
  24. 8267cf3 [Unwrap] Use RtpTimestampUnwrapper in audio_ingress by Evan Shrubsole · 2 years, 2 months ago
  25. 9253240 Reland "Remove use of ReceiveStreamRtpConfig:transport_cc" by Per K · 2 years, 3 months ago
  26. be5c713 Revert "Remove use of ReceiveStreamRtpConfig:transport_cc" by Olga Sharonova · 2 years, 3 months ago
  27. 97ba853 Remove use of ReceiveStreamRtpConfig:transport_cc by Per K · 2 years, 3 months ago
  28. 68a7c41 Revert "Enforce stream id uniqueness in RtpSender::set_stream_ids" by Ilya Nikolaevskiy · 2 years, 3 months ago
  29. 315b95c Enforce stream id uniqueness in RtpSender::set_stream_ids by Philipp Hancke · 2 years, 3 months ago
  30. 794d599 Split media_channel and its dependencies from the rtc_media_base target by Harald Alvestrand · 2 years, 3 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. e0b4cab Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead by Per Kjellander · 2 years, 4 months ago
  33. f0c33c4 Ensure audio quality tools are downloaded on Fuchsia by Christoffer Jansson · 2 years, 4 months ago
  34. acabb36 pc: Add asynchronous RtpSender::SetParameters() call by Florent Castelli · 2 years, 4 months ago
  35. a3e51df Add a new PeerConnectionE2EQualityTestFixture::AddPeer method. by Jeremy Leconte · 2 years, 4 months ago
  36. af51228 audio: make packets lost a signed integer by Philipp Hancke · 2 years, 5 months ago
  37. aebba7b [Stats] Expose totalPacketSendDelay for audio as well. by Henrik Boström · 2 years, 5 months ago
  38. 15dfc5a Add GetContributionSources to TransformableIncomingAudioFrame by Sergio Garcia Murillo · 2 years, 5 months ago
  39. 828ef91 Replace TaskQueue with MaybeWorkerThread in RtpTransportControllerInterface by Per Kjellander · 2 years, 6 months ago
  40. 9d9c2d5 Make header files self contained. by Mirko Bonadei · 2 years, 6 months ago
  41. 8a31b75 More audio stack traces by Olga Sharonova · 2 years, 6 months ago
  42. 2d0ba28 Audio stack traces by Olga Sharonova · 2 years, 6 months ago
  43. 718d7b3 Add missing export to the perf output file by Artem Titov · 2 years, 6 months ago
  44. e2f2cae Cleanup: Deduplicate static functions that create network links by Byoungchan Lee · 2 years, 6 months ago
  45. c45f4e4 [PCLF] Fully switch to new metrics export API by Artem Titov · 2 years, 6 months ago
  46. 56b96ffe Surface `local_capture_clock_offset` from `RtpSource` by Alessio Bazzica · 2 years, 6 months ago
  47. 53e5e28 Replace `ChannelReceive::GetRTT()` with `ModuleRtpRtcpImpl2::RTT()` by Alessio Bazzica · 2 years, 7 months ago
  48. 9e09a1f Replace Thread::Invoke with Thread::BlockingCall by Danil Chapovalov · 2 years, 7 months ago
  49. c92338a Remove `CallReceiveStatistics::rttMs` by Alessio Bazzica · 2 years, 7 months ago
  50. 7cc631e8 Add alessiob@webrtc.org in audio/OWNERS by Alessio Bazzica · 2 years, 7 months ago
  51. 2cfc1af Update rtc::Event::Wait call sites to use TimeDelta. by Markus Handell · 2 years, 7 months ago
  52. 0cf140d Rewrite AudioState null poller to use TaskQueueBase interface by Danil Chapovalov · 2 years, 7 months ago
  53. f4f2287 CallTest: migrate timeouts to TimeDelta. by Markus Handell · 2 years, 7 months ago
  54. e519f38 Remove rtc::Location from SendTask test helper by Danil Chapovalov · 2 years, 8 months ago
  55. 3bd444f Clarify and extend test support for certain sample rates in audio processing by Sam Zackrisson · 2 years, 8 months ago
  56. 6e7c268 Allow recursive check for RTC_DCHECK_RUN_ON macro by Danil Chapovalov · 2 years, 8 months ago
  57. 1a84b56 Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay by Ivo Creusen · 2 years, 8 months ago
  58. c05a1be Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable by Danil Chapovalov · 2 years, 8 months ago
  59. 253f36f Delete rtp_sender_ check in ModuleRtpRtcpImpl2::SetSendingMediaStatus by Niels Möller · 2 years, 8 months ago
  60. ee3ad9f Make ChannelSend::OnUplinkPacketLossRate public by Niels Möller · 2 years, 8 months ago
  61. d78789e Delete old TODOs. by Niels Möller · 2 years, 8 months ago
  62. aeb4412 Video and flexfec receive stream config changes without recreate. by Tommi · 2 years, 8 months ago
  63. 6939f63 Update old TODO comments by Niels Möller · 2 years, 9 months ago
  64. 0fd2ed5 Delete ProcessThread and related Module interface by Danil Chapovalov · 2 years, 9 months ago
  65. c374d11 Move to_queued_task.h and pending_task_safety_flag.h into public API by Artem Titov · 2 years, 9 months ago
  66. 105711e Move rtc::make_ref_counted to api/ by Niels Möller · 2 years, 9 months ago
  67. 83e34ee Migrate some scripts to python3 by Björn Terelius · 2 years, 10 months ago
  68. a136ed4 Add SetTransportCc to ReceiveStreamInterface. by Tommi · 2 years, 10 months ago
  69. 3176ef7 Rename AudioReceiveStream to AudioReceiveStreamInterface by Tommi · 2 years, 10 months ago
  70. dddbbeb Rename internal::AudioReceiveStream to AudioReceiveStreamImpl by Tommi · 2 years, 10 months ago
  71. 1def899 Remove legacy (unused) config param: jitter_buffer_enable_rtx_handling by Tommi · 2 years, 10 months ago
  72. 83830f3 Delete TestListener and top-level thread wrapping. by Niels Möller · 2 years, 10 months ago
  73. dab50c6 Revert "Use ADM internal state for init state check." by Mirko Bonadei · 2 years, 10 months ago
  74. 0e2221e Use ADM internal state for init state check. by Tim Na · 2 years, 11 months ago
  75. cf4ed15 Add GetRtpExtensionMap to ReceiveStream and remove GetRtpExtensions. by Tommi · 2 years, 11 months ago
  76. edcb25b Migrate RemoteNtpTimeEstimator to more precise time representations by Danil Chapovalov · 2 years, 11 months ago
  77. 7a15ff3 Add a transport_cc() getter and remove rtp_config(). by Tommi · 2 years, 11 months ago
  78. 6be3e78 Add getter for rtp header extensions for receiver classes. by Tommi · 2 years, 11 months ago
  79. 853a407 Revert "Migrate RemoteNtpTimeEstimator to more precise time representations" by Danil Chapovalov · 2 years, 11 months ago
  80. cb7c736 Separate reading remote_ssrc from using the rtp_config() getter. by Tommi · 2 years, 11 months ago
  81. a154a15 Migrate RemoteNtpTimeEstimator to more precise time representations by Danil Chapovalov · 2 years, 11 months ago
  82. ea1e6f4 Delete rtc_base/format_macros.h by Niels Möller · 2 years, 11 months ago
  83. cc50b04 Remove config() getter from AudioReceiveStream(). by Tommi · 2 years, 11 months ago
  84. c3e6e3a Remove dependency on rtc_base_approved from most targets by Florent Castelli · 2 years, 11 months ago
  85. f9c5984 Move buffer out of rtc_base_approved by Florent Castelli · 3 years ago
  86. f4db351 Move race_checker out of rtc_base_approved by Florent Castelli · 3 years ago
  87. ba2de58 Update audio/, media/, and video/ to not use implicit conversion by Niels Möller · 3 years ago
  88. 0c68a7a Use WebRTC's Java VM initialization in tests. by Björn Terelius · 3 years ago
  89. aa6d05d Move location out of rtc_base_approved by Florent Castelli · 3 years ago
  90. 8a1a0af In audio ChannelSend move task queue as last class member by Danil Chapovalov · 3 years ago
  91. dd837e2 Remove //rtc_base:timeutils from public deps by Florent Castelli · 3 years ago
  92. 57aa81b Remove //rtc_base:stringutils from public deps by Florent Castelli · 3 years ago
  93. e10a9f6 Remove //rtc_base:safe_conversions from public deps by Florent Castelli · 3 years ago
  94. 33d31fb Remove //rtc_base:rtc_event from public deps by Florent Castelli · 3 years ago
  95. f86f6f9 Remove //rtc_base:refcount from public deps by Florent Castelli · 3 years ago
  96. 4467ad7 Remove //rtc_base:macromagic from public deps by Florent Castelli · 3 years ago
  97. 0af55ba Remove //rtc_base:logging from public deps by Florent Castelli · 3 years ago
  98. e62c2f2 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf by Jonas Oreland · 3 years ago
  99. bddfa1d Prepare the code to inherit from chromium's mb.py (3rd attempt). by Jeremy Leconte · 3 years ago
  100. 1b51b11 Fix low_bandwidth_audio_perf_test gn group definition on windows. by Jeremy Leconte · 3 years ago