1. 714e3cb Adopt absl::string_view in modules/audio_coding/ by Ali Tofigh · 3 years ago
  2. 190244b Remove all #include <assert.h>/<cassert> and usage in Obj-C code. by Mirko Bonadei · 4 years ago
  3. 25ab322 Replace assert() with RTC_DCHECK(). by Mirko Bonadei · 4 years ago
  4. b4100ad Avoid using legacy rtp parser in neteq test::Packet by Danil Chapovalov · 4 years, 1 month ago
  5. ff0e4db Reland "Send absolute capture time through audio coding module." by Minyue Li · 5 years ago
  6. 4175914 Revert "Send absolute capture time through audio coding module." by Minyue Li · 5 years ago
  7. 48655cf Send absolute capture time through audio coding module. by Minyue Li · 5 years ago
  8. c35b6e6 Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData by Niels Möller · 6 years ago
  9. c936cb6 Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h by Niels Möller · 6 years ago
  10. 87e2d78 Prepare for splitting FrameType into AudioFrameType and VideoFrameType by Niels Möller · 6 years ago
  11. 6543881 2nd reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 6 years ago
  12. 8b3db59 Revert "Reland of https://webrtc-review.googlesource.com/c/src/+/114883" by Alex Loiko · 6 years ago
  13. 5341aac Reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 6 years ago
  14. ffd1f93 Revert "Tests for multi-stream Opus." by Mirko Bonadei · 6 years ago
  15. 9c31ac2 Tests for multi-stream Opus. by Alex Loiko · 6 years ago
  16. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 7 years ago
  17. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 7 years ago
  18. 801500c Audio encoder tests: Create audio encoders the new way by Karl Wiberg · 7 years ago
  19. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  20. 5817d3d AudioCodingModule::Create(): Require caller to supply an AudioDecoderFactory by Karl Wiberg · 7 years ago
  21. c7b4a45 Remove various IDs: by solenberg · 8 years ago
  22. e423a9de Revert of Remove various IDs (patchset #7 id:120001 of https://codereview.webrtc.org/3019543002/ ) by solenberg · 8 years ago
  23. 2d0f775 Remove various IDs: by solenberg · 8 years ago
  24. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  25. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/modules/audio_coding/acm2/acm_send_test.cc]
  26. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  27. dca1e09 Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)" by Henrik Kjellander · 8 years ago
  28. c8fa692 Update includes for webrtc/{base => rtc_base} rename (1/3) by kjellander · 8 years ago
  29. 36b1a5f Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide: by yujo · 8 years ago
  30. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 8 years ago
  31. 2504c0a Drop _oldapi from ACM test file names by henrik.lundin · 9 years ago[Renamed (98%) from webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc]
  32. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 9 years ago
  33. 77eab70 Enable the -Wundef warning for clang by kwiberg · 9 years ago
  34. 65fc8b9 Fix Chromium clang plugin warnings by kwiberg · 9 years ago
  35. 46ba49c Let PacketSource::NextPacket() return an std::unique_ptr by henrik.lundin · 9 years ago
  36. 3e6db23 audio_coding: remove "main" directory by kjellander · 10 years ago[Renamed (97%) from webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc]
  37. 7464089 audio_coding: rename interface -> include by Henrik Kjellander · 10 years ago
  38. 91d6ede Add RTC_ prefix to (D)CHECKs and related macros. by henrikg · 10 years ago
  39. 12cfc9b Fold AudioEncoderMutable into AudioEncoder by kwiberg · 10 years ago
  40. dce40cf Update a ton of audio code to use size_t more correctly and in general reduce by Peter Kasting · 10 years ago
  41. b7e5054 Match existing type usage better. by Peter Kasting · 10 years ago
  42. 7e0c7d4 Add support for external encoders in ACM by Karl Wiberg · 10 years ago
  43. f56c162 Remove AudioCodingModule::Process() by henrik.lundin@webrtc.org · 10 years ago
  44. 0e81fdf Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting. by pkasting@chromium.org · 10 years ago
  45. 4591fbd Use size_t more consistently for packet/payload lengths. by pkasting@chromium.org · 11 years ago
  46. 0e6e4d2 Reland "Converting five tests to use new AudioCoding interface" (r7258) by henrik.lundin@webrtc.org · 11 years ago
  47. 99e404c Revert "Converting five tests to use new AudioCoding interface" (rev 7258). by andresp@webrtc.org · 11 years ago
  48. eb1de5c Converting five tests to use new AudioCoding interface by henrik.lundin@webrtc.org · 11 years ago