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webrtc / src.git / 99c519b3fdac0cbedaecead04a841e89a5f0f69b / audio / voip / BUILD.gn
  1. 99c519b Mass removal of absl_deps in all BUILD.gn files by Florent Castelli · 10 months ago
  2. f4673f9 Move webrtc::AudioDeviceModule include to api/ folder by Florent Castelli · 11 months ago
  3. 0afde76 Move webrtc::AudioProcessing include to api/ folder by Florent Castelli · 11 months ago
  4. b1799b0 Cleanup usage of the rtc::TaskQueue in audio/ by Danil Chapovalov · 1 year, 2 months ago
  5. 8267cf3 [Unwrap] Use RtpTimestampUnwrapper in audio_ingress by Evan Shrubsole · 2 years, 2 months ago
  6. 9e09a1f Replace Thread::Invoke with Thread::BlockingCall by Danil Chapovalov · 2 years, 6 months ago
  7. 0fd2ed5 Delete ProcessThread and related Module interface by Danil Chapovalov · 2 years, 8 months ago
  8. c3e6e3a Remove dependency on rtc_base_approved from most targets by Florent Castelli · 2 years, 11 months ago
  9. aa6d05d Move location out of rtc_base_approved by Florent Castelli · 2 years, 11 months ago
  10. d15a575 Use SequenceChecker from public API by Artem Titov · 4 years, 1 month ago
  11. 06159aa Remove deprecated thread checker by Artem Titov · 4 years, 1 month ago
  12. c8421c4 Replace rtc::ThreadChecker with webrtc::SequenceChecker by Artem Titov · 4 years, 1 month ago
  13. 507eacf Reland "ChannelStatistics used for RTP stats in VoipStatistics." by Tim Na · 4 years, 2 months ago
  14. 37827c9 Revert "ChannelStatistics used for RTP stats in VoipStatistics." by Alex Loiko · 4 years, 2 months ago
  15. 444e04b ChannelStatistics used for RTP stats in VoipStatistics. by Tim Na · 4 years, 2 months ago
  16. 2412602 Using absl::optional for round trip time return type handling. by Tim Na · 4 years, 3 months ago
  17. 20e4c80 Reland "Introduce RTC_NO_UNIQUE_ADDRESS." by Mirko Bonadei · 4 years, 4 months ago
  18. 0abd518 Revert "Introduce RTC_NO_UNIQUE_ADDRESS." by Mirko Bonadei · 4 years, 5 months ago
  19. f5e261a Introduce RTC_NO_UNIQUE_ADDRESS. by Mirko Bonadei · 4 years, 6 months ago
  20. 6287280 Migrate audio/ to use webrtc::Mutex by Markus Handell · 4 years, 8 months ago
  21. 2dcf348 Use absl_deps in order to preapre to the Abseil component build release. by Mirko Bonadei · 4 years, 9 months ago
  22. c0df5fc VoIP API implementation on top of AudioIngress/Egress by Tim Na · 4 years, 10 months ago
  23. 11f92bc Audio ingress implementation for voip api. by Tim Na · 5 years ago
  24. 8ab3c77 Audio egress implementation for initial voip api in api/voip. by Tim Na · 5 years ago