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gerrit-public.fairphone.software / platform / external / webrtc / a4d873786f10eedd72de25ad0d94ad7c53c1f68a / modules / audio_coding / codecs / opus
  1. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  2. e8fbc5d Refactor WebRtcOpus_PacketHasFec. by Minyue Li · 5 years ago
  3. 7eb0a5e AudioDecoderOpus: Add support for 16 kHz output sample rate by Karl Wiberg · 5 years ago
  4. a1d1a1e WebRTC Opus C interface: Add support for non-48 kHz decode sample rate by Karl Wiberg · 5 years ago
  5. 126f2b3 AudioEncoderOpus: Add support for 16 kHz input sample rate by Karl Wiberg · 5 years ago
  6. e0eb325 AudioEncoderOpusImpl: Remove unused static methods by Karl Wiberg · 5 years ago
  7. 7e7c5c3 WebRTC Opus C interface: Add support for non-48 kHz encode sample rate by Karl Wiberg · 5 years ago
  8. eb16697 AudioEncoderOpus: Don't mix up sample rate and RTP timestamp rate by Karl Wiberg · 5 years ago
  9. 44c21f4 Encoder side of Multistream Opus. by Alex Loiko · 5 years ago
  10. 40889f3 Removes TimeMicros interface from ThreadProcessingFakeClock. by Sebastian Jansson · 5 years ago
  11. 6a489f2 Fully qualify googletest symbols. by Mirko Bonadei · 5 years ago
  12. e5b9416 Decoder for multistream Opus. by Alex Loiko · 5 years ago
  13. 50b8c39 Generalize the C-language Opus interface. by Alex Loiko · 5 years ago
  14. 94b57c0 Cleanup BUILD.gn files from imports like foo:foo by Artem Titov · 5 years ago
  15. c4dd730 Fix -Wextra-semi warnings. by Mirko Bonadei · 5 years ago
  16. 6543881 2nd reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 5 years ago
  17. 22f9925 webrtc: Remove semicolons. by Nico Weber · 5 years ago
  18. 8b3db59 Revert "Reland of https://webrtc-review.googlesource.com/c/src/+/114883" by Alex Loiko · 5 years ago
  19. 5341aac Reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 5 years ago
  20. ffd1f93 Revert "Tests for multi-stream Opus." by Mirko Bonadei · 5 years ago
  21. 9c31ac2 Tests for multi-stream Opus. by Alex Loiko · 5 years ago
  22. e45c688 Remove webrtc::ProtoString. by Mirko Bonadei · 5 years ago
  23. 05cf6be [clang-tidy] Apply performance-move-const-arg fixes. by Mirko Bonadei · 5 years ago
  24. c84f661 Stop using Googletest legacy APIs. by Mirko Bonadei · 5 years ago
  25. 7a3e43a Reland of Opus multistream. by Alex Loiko · 5 years ago
  26. 9dac02d Adding text log on actual opus bitrate. by Minyue Li · 5 years ago
  27. 1fa51d6 Revert "Opus multistream." by Amit Hilbuch · 5 years ago
  28. 83ed89a Opus multistream. by Alex Loiko · 5 years ago
  29. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 5 years ago
  30. 41f3a43 Remove CodecInst pt.3 by Fredrik Solenberg · 6 years ago
  31. 8ac05cc Adds trial to use link capacity estimate in Opus encoder. by Sebastian Jansson · 6 years ago
  32. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  33. 2edab4c Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. by Niels Möller · 6 years ago
  34. 83bd37c Add field trials for configuring Opus encoder packet loss rate. by Jakob Ivarsson · 6 years ago
  35. 88b68ac Create field trial for setting a minimum value for Opus encoder packet loss rate by Jakob Ivarsson · 6 years ago
  36. d4161a3 Moving LappedTransform, Blocker and AudioRingBuffer. by Alessio Bazzica · 6 years ago
  37. a12c42a Delete root header file typedef.h. by Niels Möller · 6 years ago
  38. 682aac5 Enable clang::find_bad_constructs for audio_coding (part 1/2). by Mirko Bonadei · 6 years ago
  39. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  40. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  41. b602123 Replace rtc::Optional with absl::optional in modules/audio_coding by Danil Chapovalov · 6 years ago
  42. c7f09ad NetEq fix for repeated audio issue. by Ivo Creusen · 6 years ago
  43. b9fc650 Add min and max allowed bitrate in Opus bitrate tests by Gustaf Ullberg · 6 years ago
  44. 5f83cf0 Replacing rtc::TimeDelta with webrtc::TimeDelta. by Sebastian Jansson · 6 years ago
  45. 2a35c43 Removing shared_ptr in a unittest in audio coding. by Minyue Li · 6 years ago
  46. e40468b Move some numeric utility code from rtc_base/ to rtc_base/numerics/ by Karl Wiberg · 7 years ago
  47. eeb2765 Implement Opus bandwidth adjustment behind a FieldTrial by Alex Luebs · 7 years ago
  48. 36de62e Avoid flagging Opus DTX frames as speech. by Gustaf Ullberg · 7 years ago
  49. 12ab00b Optional: Use nullopt and implicit construction in /modules/audio_coding by Oskar Sundbom · 7 years ago
  50. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  51. 7275e18 Hide the internal AudioEncoderOpus class by giving it an "Impl" suffix by Karl Wiberg · 7 years ago
  52. c5ee987 Stop using std::tr1 by Edward Lemur · 7 years ago
  53. 737e073 Fixing warning C4267 on Win (more_configs). by Mirko Bonadei · 7 years ago
  54. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  55. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  56. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago