Google Git
Sign in
webrtc / src / d440358ccaeb9aa5dc4a50c0c37638416a15cac5 / api / audio_codecs
  1. 9d9b3a3 Add option for the audio encoder to allocate a bitrate range. by Jakob Ivarsson · 1 year, 1 month ago
  2. de17252 Reland "Unify access to SDP codec parameters" by Philipp Hancke · 1 year, 2 months ago
  3. 6c9c958 Revert "Unify access to SDP codec parameters" by Mirko Bonadei · 1 year, 3 months ago
  4. 63d03f5 Unify access to SDP codec parameters by Philipp Hancke · 1 year, 3 months ago
  5. 78f905e Move some users to use webrtc::RefCountInterface by Harald Alvestrand · 1 year, 4 months ago
  6. bceec84 Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ by Jared Siskin · 1 year, 11 months ago
  7. 6e1ae44 Don't use low complexity Opus on all ARM devices. by Jakob Ivarsson · 2 years, 3 months ago
  8. 8754a3c Update some audio modules with new OWNERS by Henrik Lundin · 2 years, 4 months ago
  9. b00f881 Remove xooglers from WATCHLISTS and OWNERS by Christoffer Jansson · 2 years, 4 months ago
  10. 17887eb Reland "[ACM] iSAC audio codec removed" by Alessio Bazzica · 2 years, 4 months ago
  11. fbeb76a Revert "[ACM] iSAC audio codec removed" by Alessio Bazzica · 2 years, 4 months ago
  12. b46c4bf [ACM] iSAC audio codec removed by Alessio Bazzica · 2 years, 4 months ago
  13. 5c9b7da Add missing dependencies. by Mirko Bonadei · 2 years, 5 months ago
  14. 9d9c2d5 Make header files self contained. by Mirko Bonadei · 2 years, 5 months ago
  15. 7c8c4db Add rtc::make_ref_counted to api/ by Niels Möller · 2 years, 9 months ago
  16. c3e6e3a Remove dependency on rtc_base_approved from most targets by Florent Castelli · 2 years, 11 months ago
  17. a30aef3 Move event_tracer out of rtc_base_approved by Florent Castelli · 2 years, 11 months ago
  18. f9c5984 Move buffer out of rtc_base_approved by Florent Castelli · 2 years, 11 months ago
  19. 57aa81b Remove //rtc_base:stringutils from public deps by Florent Castelli · 3 years ago
  20. e10a9f6 Remove //rtc_base:safe_conversions from public deps by Florent Castelli · 3 years ago
  21. f86f6f9 Remove //rtc_base:refcount from public deps by Florent Castelli · 3 years ago
  22. e62c2f2 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf by Jonas Oreland · 3 years ago
  23. 6e2b9e2 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 5/inf by Jonas Oreland · 3 years ago
  24. c065e73 Remove RTC_DISALLOW_COPY_AND_ASSIGN more. by Byoungchan Lee · 3 years, 2 months ago
  25. deb1b1b Always call IsOk() to ensure audio codec configuration is valid when negotiating. by Ivo Creusen · 3 years, 4 months ago
  26. d823259 Set the maximum number of audio channels to 24 by Ivo Creusen · 3 years, 4 months ago
  27. d325196 Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED by Artem Titov · 3 years, 4 months ago
  28. 0e61fdd Use backticks not vertical bars to denote variables in comments for /api by Artem Titov · 3 years, 8 months ago
  29. 190244b Remove all #include <assert.h>/<cassert> and usage in Obj-C code. by Mirko Bonadei · 3 years, 8 months ago
  30. 25ab322 Replace assert() with RTC_DCHECK(). by Mirko Bonadei · 3 years, 8 months ago
  31. c1d5891 Replace `new rtc::RefCountedObject` with `rtc::make_ref_counted` in a few files by Tomas Gunnarsson · 3 years, 11 months ago
  32. a3575cb Remove tautological 'unsigned expr < 0' comparisons by Anton Bikineev · 4 years ago
  33. e904161 Replace RTC_DEPRECATED with ABSL_DEPRECATED by Danil Chapovalov · 4 years, 1 month ago
  34. ff0e01f Implement audio_interruption metric for kCodecPlc by Pablo Barrera González · 4 years, 1 month ago
  35. 6c80aeb Remove kwiberg@webrtc.org from OWNERS files by Henrik Lundin · 4 years, 3 months ago
  36. c95b939 Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED() by Karl Wiberg · 4 years, 4 months ago
  37. d354d56 Audio codec factories: Implementations return null on unsupported formats by Karl Wiberg · 4 years, 7 months ago
  38. 2dcf348 Use absl_deps in order to preapre to the Abseil component build release. by Mirko Bonadei · 4 years, 9 months ago
  39. b261118 Fix a typo for decoder naming by Hua, Chunbo · 4 years, 11 months ago
  40. 90ecee1 Make AudioEncoder::GetFrameLengthRange() pure virtual. by Ali Tofigh · 5 years ago
  41. e52115a Remove inactive OWNERS. by Mirko Bonadei · 5 years ago
  42. 08b11ca iSAC config: target bitrate exposed for fixed impl by Alessio Bazzica · 5 years ago
  43. ccbe95f Reformat GN files. by Mirko Bonadei · 5 years ago
  44. 44d7ec0 Add Opus-only audio codec factories by Karl Wiberg · 5 years ago
  45. cd2a92f Removes RPLR based FEC controller. by Sebastian Jansson · 5 years ago
  46. c66e004 Adding missing RTC_EXPORT for component build. by Mirko Bonadei · 5 years ago
  47. 86d053c Use source_sets in component builds and static_library in release builds. by Mirko Bonadei · 5 years ago
  48. 62aee93 Adds trial to calculate audio overhead based on available data. by Sebastian Jansson · 5 years ago
  49. ef14f07 Delete AudioDecoder method IncomingPacket by Niels Möller · 5 years ago
  50. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
  51. 608e6ba Add AudioDecoderIsacT::Config to include sampling rate and BWInfo object by Jiawei Ou · 6 years ago
  52. a4d8737 Format almost everything. by Jonas Olsson · 6 years ago
  53. 7eb0a5e AudioDecoderOpus: Add support for 16 kHz output sample rate by Karl Wiberg · 6 years ago
  54. 126f2b3 AudioEncoderOpus: Add support for 16 kHz input sample rate by Karl Wiberg · 6 years ago
  55. 44c21f4 Encoder side of Multistream Opus. by Alex Loiko · 6 years ago
  56. 6a489f2 Fully qualify googletest symbols. by Mirko Bonadei · 6 years ago
  57. e5b9416 Decoder for multistream Opus. by Alex Loiko · 6 years ago
  58. 9d8eaac Delete unneeded direct includes of common_types.h by Niels Möller · 6 years ago
  59. 3aa584f Fixes ClangTidy issues in api/ by Benjamin Wright · 6 years ago
  60. 6543881 2nd reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 6 years ago
  61. 8b3db59 Revert "Reland of https://webrtc-review.googlesource.com/c/src/+/114883" by Alex Loiko · 6 years ago
  62. 5341aac Reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 6 years ago
  63. ffd1f93 Revert "Tests for multi-stream Opus." by Mirko Bonadei · 6 years ago
  64. 9c31ac2 Tests for multi-stream Opus. by Alex Loiko · 6 years ago
  65. 1c54605 [clang-tidy] Apply performance-move-const-arg fixes (misc). by Mirko Bonadei · 6 years ago
  66. 05cf6be [clang-tidy] Apply performance-move-const-arg fixes. by Mirko Bonadei · 6 years ago
  67. d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 6 years ago
  68. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  69. ad858d1 Improve the audio codec factory documentation. by Karl Wiberg · 6 years ago
  70. b7180c0 Replace RegisterExternalDecoder in NetEq test VerifyTimestampPropagation. by Niels Möller · 6 years ago
  71. 540ef28 Adds OnReceivedUplinkAllocation method to AudioEncoder. by Sebastian Jansson · 6 years ago
  72. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  73. 2edab4c Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. by Niels Möller · 6 years ago
  74. ac19414 Export symbols needed by the Chromium component build (part 6). by Mirko Bonadei · 6 years ago
  75. e482ff8 Audio codecs API: Remove some weasel words in the docs by Karl Wiberg · 6 years ago
  76. 276827c Export symbols needed by the Chromium component build (part 3). by Mirko Bonadei · 6 years ago
  77. 3b56ee7 Export symbols needed by the Chromium component build (part 2). by Mirko Bonadei · 6 years ago
  78. 3d25530 Reland "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
  79. edc49c1 [Cleanup] Remove unused swap function. by Yves Gerey · 6 years ago
  80. 16fe3f2 Revert "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
  81. 99eea42 Reland "Reland "Export symbols needed by the Chromium component build (part 1)."" by Mirko Bonadei · 6 years ago
  82. 2e00abc Reland "[cleanup] Remove useless includes." by Yves Gerey · 6 years ago
  83. 96a0f61 Revert "[cleanup] Remove useless includes." by Oleh Prypin · 6 years ago
  84. be8b534 [cleanup] Remove useless includes. by Yves Gerey · 6 years ago
  85. b49520b Revert "Reland "Export symbols needed by the Chromium component build (part 1)."" by Mirko Bonadei · 6 years ago
  86. 588f464 Reland "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
  87. 2ea9af2 Revert "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
  88. 9e24dcf Export symbols needed by the Chromium component build (part 1). by Mirko Bonadei · 6 years ago
  89. 00eb12a Let NetEq use the PLC output from a decoder by Henrik Lundin · 7 years ago
  90. 801500c Audio encoder tests: Create audio encoders the new way by Karl Wiberg · 7 years ago
  91. 24744a9 Use string_view instead of overloading for const string& and const char* by Karl Wiberg · 7 years ago
  92. 133cff0 AudioCodingModuleTest.TestAllCodecs: Create audio encoders the new way by Karl Wiberg · 7 years ago
  93. a12c42a Delete root header file typedef.h. by Niels Möller · 7 years ago
  94. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 7 years ago
  95. 0bc58cf Replace rtc::Optional with absl::optional in api by Danil Chapovalov · 7 years ago
  96. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  97. f1c470e Remove the audio codec factory methods that don't take AudioCodecPairId by Karl Wiberg · 7 years ago
  98. c7f09ad NetEq fix for repeated audio issue. by Ivo Creusen · 7 years ago
  99. 3e18c82 Reland "Reland "Remove our stream << overloads from non-test build targets."" by Jonas Olsson · 7 years ago
  100. bb23c83 GN hack to tag targets as poisonous (and use it with audio codecs) by Karl Wiberg · 7 years ago