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webrtc / src / d440358ccaeb9aa5dc4a50c0c37638416a15cac5 / api / audio_codecs / opus
  1. bceec84 Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ by Jared Siskin · 1 year, 11 months ago
  2. 6e1ae44 Don't use low complexity Opus on all ARM devices. by Jakob Ivarsson · 2 years, 3 months ago
  3. 9d9c2d5 Make header files self contained. by Mirko Bonadei · 2 years, 5 months ago
  4. c3e6e3a Remove dependency on rtc_base_approved from most targets by Florent Castelli · 2 years, 11 months ago
  5. e62c2f2 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf by Jonas Oreland · 3 years ago
  6. 6e2b9e2 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 5/inf by Jonas Oreland · 3 years ago
  7. deb1b1b Always call IsOk() to ensure audio codec configuration is valid when negotiating. by Ivo Creusen · 3 years, 4 months ago
  8. d823259 Set the maximum number of audio channels to 24 by Ivo Creusen · 3 years, 4 months ago
  9. 0e61fdd Use backticks not vertical bars to denote variables in comments for /api by Artem Titov · 3 years, 8 months ago
  10. a3575cb Remove tautological 'unsigned expr < 0' comparisons by Anton Bikineev · 4 years ago
  11. 2dcf348 Use absl_deps in order to preapre to the Abseil component build release. by Mirko Bonadei · 4 years, 9 months ago
  12. ccbe95f Reformat GN files. by Mirko Bonadei · 5 years ago
  13. 86d053c Use source_sets in component builds and static_library in release builds. by Mirko Bonadei · 5 years ago
  14. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
  15. 7eb0a5e AudioDecoderOpus: Add support for 16 kHz output sample rate by Karl Wiberg · 6 years ago
  16. 126f2b3 AudioEncoderOpus: Add support for 16 kHz input sample rate by Karl Wiberg · 6 years ago
  17. 44c21f4 Encoder side of Multistream Opus. by Alex Loiko · 6 years ago
  18. e5b9416 Decoder for multistream Opus. by Alex Loiko · 6 years ago
  19. 6543881 2nd reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 6 years ago
  20. 8b3db59 Revert "Reland of https://webrtc-review.googlesource.com/c/src/+/114883" by Alex Loiko · 6 years ago
  21. 5341aac Reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 6 years ago
  22. ffd1f93 Revert "Tests for multi-stream Opus." by Mirko Bonadei · 6 years ago
  23. 9c31ac2 Tests for multi-stream Opus. by Alex Loiko · 6 years ago
  24. 05cf6be [clang-tidy] Apply performance-move-const-arg fixes. by Mirko Bonadei · 6 years ago
  25. 2edab4c Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. by Niels Möller · 6 years ago
  26. e482ff8 Audio codecs API: Remove some weasel words in the docs by Karl Wiberg · 6 years ago
  27. 3b56ee7 Export symbols needed by the Chromium component build (part 2). by Mirko Bonadei · 6 years ago
  28. 3d25530 Reland "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
  29. 16fe3f2 Revert "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
  30. 99eea42 Reland "Reland "Export symbols needed by the Chromium component build (part 1)."" by Mirko Bonadei · 6 years ago
  31. b49520b Revert "Reland "Export symbols needed by the Chromium component build (part 1)."" by Mirko Bonadei · 6 years ago
  32. 588f464 Reland "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
  33. 2ea9af2 Revert "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
  34. 9e24dcf Export symbols needed by the Chromium component build (part 1). by Mirko Bonadei · 6 years ago
  35. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 7 years ago
  36. 0bc58cf Replace rtc::Optional with absl::optional in api by Danil Chapovalov · 7 years ago
  37. bb23c83 GN hack to tag targets as poisonous (and use it with audio codecs) by Karl Wiberg · 7 years ago
  38. 17668ec Audio codec implementations: Take optional codec ID argument by Karl Wiberg · 7 years ago
  39. 6114c24 Stop using public_deps in api. by Mirko Bonadei · 7 years ago
  40. a7f2d84 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" by Per Kjellander · 7 years ago
  41. c73e1f4 Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" by Per Kjellander · 7 years ago
  42. 588c548 GN rtc_* templates: Set default visibility to webrtc_root + "/*" by Karl Wiberg · 7 years ago
  43. 9065730 Optional: Use nullopt and implicit construction in /api/audio_codecs by Oskar Sundbom · 7 years ago
  44. 7275e18 Hide the internal AudioEncoderOpus class by giving it an "Impl" suffix by Karl Wiberg · 7 years ago
  45. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 8 years ago
  46. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  47. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago