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gerrit-public.fairphone.software / platform / external / webrtc / a4d873786f10eedd72de25ad0d94ad7c53c1f68a / modules / audio_coding / include / audio_coding_module_typedefs.h
  1. 8d8ffdb Expose new audio stats on the API by Ivo Creusen · 5 years ago
  2. 44125fa Reland "Piping audio interruption metrics to API layer" by Henrik Lundin · 5 years ago
  3. fc02a79 Revert "Piping audio interruption metrics to API layer" by Henrik Andreassson · 5 years ago
  4. 299c4e6 Piping audio interruption metrics to API layer by Henrik Lundin · 5 years ago
  5. c936cb6 Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h by Niels Möller · 5 years ago
  6. 232b3fd Expose relative packet arrival delay metric in stats API. by Jakob Ivarsson · 5 years ago
  7. 0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 6 years ago
  8. 352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
  9. 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
  10. 78e88fe Move NetworkStatistics and AudioDecodingCallStats from common_types.h by Fredrik Solenberg · 6 years ago
  11. a12c42a Delete root header file typedef.h. by Niels Möller · 6 years ago
  12. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  13. bbf21a3 Remove dependencies on modules:module_api from AudioProcessing. by Fredrik Solenberg · 6 years ago
  14. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  15. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  16. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h]
  17. 3e6db23 audio_coding: remove "main" directory by kjellander · 9 years ago