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gerrit-public.fairphone.software / platform / external / webrtc / a4d873786f10eedd72de25ad0d94ad7c53c1f68a / modules / audio_coding / include
  1. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  2. 8d8ffdb Expose new audio stats on the API by Ivo Creusen · 5 years ago
  3. 44125fa Reland "Piping audio interruption metrics to API layer" by Henrik Lundin · 5 years ago
  4. fc02a79 Revert "Piping audio interruption metrics to API layer" by Henrik Andreassson · 5 years ago
  5. 4babc68 Delete deprecated version of AudioPacketizationCallback::SendData. by Niels Möller · 5 years ago
  6. 299c4e6 Piping audio interruption metrics to API layer by Henrik Lundin · 5 years ago
  7. c35b6e6 Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData by Niels Möller · 5 years ago
  8. 741daaf Move rtc::FunctionView to the public API by Artem Titov · 5 years ago
  9. c936cb6 Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h by Niels Möller · 5 years ago
  10. 87e2d78 Prepare for splitting FrameType into AudioFrameType and VideoFrameType by Niels Möller · 5 years ago
  11. 232b3fd Expose relative packet arrival delay metric in stats API. by Jakob Ivarsson · 5 years ago
  12. 1925b5a Delete deprecated version of AudioCodingModule::IncomingPacket by Niels Möller · 5 years ago
  13. afb5dbb Update ACM to use RTPHeader instead of WebRtcRTPHeader by Niels Möller · 5 years ago
  14. 3b50f9f Propagate base minimum delay to audio_receiver_stream by Ruslan Burakov · 5 years ago
  15. 0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 5 years ago
  16. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
  17. 352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
  18. 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
  19. 78e88fe Move NetworkStatistics and AudioDecodingCallStats from common_types.h by Fredrik Solenberg · 6 years ago
  20. 49c33ce AudioCodingModule: Remove support for creating encoders by Karl Wiberg · 6 years ago
  21. eddd366 Delete unused method AudioCodingModuleImpl::SetOpusApplication. by Niels Möller · 6 years ago
  22. 764c14c Delete unused AudioCodingModule methods. by Niels Möller · 6 years ago
  23. 18f1adc Delete AudioCodingModule::LeastRequiredDelayMs and related NetEq code. by Niels Möller · 6 years ago
  24. ec93075 Delete deprecated methods on AudioCodingModule by Niels Möller · 6 years ago
  25. a12c42a Delete root header file typedef.h. by Niels Möller · 6 years ago
  26. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  27. b602123 Replace rtc::Optional with absl::optional in modules/audio_coding by Danil Chapovalov · 6 years ago
  28. bbf21a3 Remove dependencies on modules:module_api from AudioProcessing. by Fredrik Solenberg · 6 years ago
  29. 5817d3d AudioCodingModule::Create(): Require caller to supply an AudioDecoderFactory by Karl Wiberg · 6 years ago
  30. abbff89 Add new UMA metric for NetEq target buffer delay by Henrik Lundin · 7 years ago
  31. d4a790f Remove AudioCodingModule::IncomingPayload by Henrik Lundin · 7 years ago
  32. c7b4a45 Remove various IDs: by solenberg · 7 years ago
  33. e423a9de Revert of Remove various IDs (patchset #7 id:120001 of https://codereview.webrtc.org/3019543002/ ) by solenberg · 7 years ago
  34. 2d0f775 Remove various IDs: by solenberg · 7 years ago
  35. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  36. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  37. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago