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gerrit-public.fairphone.software / platform / external / webrtc / 8f319a3472b19172c1d3d6849be373338ccd85b4 / api / audio / audio_frame.h
  1. 8f319a3 Reland "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" by Alessio Bazzica · 5 years ago
  2. fab3460 Revert "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" by Alessio Bazzica · 5 years ago
  3. 9973933 Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." by Chen Xing · 5 years ago
  4. 24192c2 Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." by Ivo Creusen · 5 years ago
  5. 2250b05 Adding support for channel mixing between different channel layouts. by henrika · 5 years ago
  6. 3e8ef94 Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker. by Chen Xing · 5 years ago
  7. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  8. 2a49065 Increases max size of webrtc::AudioFrame from 60ms to 120ms @32kHz. by henrika · 6 years ago
  9. a12c42a Delete root header file typedef.h. by Niels Möller · 6 years ago
  10. 0bc58cf Replace rtc::Optional with absl::optional in api by Danil Chapovalov · 6 years ago
  11. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  12. bbf21a3 Remove dependencies on modules:module_api from AudioProcessing. by Fredrik Solenberg · 7 years ago
  13. 03bfc73 Remove deprecated methods from AudioFrame. by Fredrik Solenberg · 7 years ago
  14. d377f04 Move AudioFrame to its own header file and target in api/. by Niels Möller · 7 years ago