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webrtc / src / 19c51ea5370d5e78b384e6ef38cbebc48f23a425 / audio / remix_resample.cc
  1. 19c51ea Use std::array<> consistently for reusable audio buffers. by Tommi · 10 months ago
  2. 5d3e680 Add audio view classes by Tommi · 10 months ago
  3. 57b09ec Update AudioFrameOperations to require ArrayView by Tommi · 11 months ago
  4. 1f36798 Start using ArrayView in AudioFrame, update PushResampler by Tommi · 11 months ago
  5. 01719fb Reland "Rename FATAL() into RTC_FATAL()." by Mirko Bonadei · 4 years, 4 months ago
  6. a4fd641 Revert "Rename FATAL() into RTC_FATAL()." by Mirko Bonadei · 4 years, 4 months ago
  7. 9653d26 Rename FATAL() into RTC_FATAL(). by Mirko Bonadei · 4 years, 4 months ago
  8. 8f319a3 Reland "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" by Alessio Bazzica · 6 years ago
  9. fab3460 Revert "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" by Alessio Bazzica · 6 years ago
  10. 9973933 Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." by Chen Xing · 6 years ago
  11. 24192c2 Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." by Ivo Creusen · 6 years ago
  12. 3e8ef94 Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker. by Chen Xing · 6 years ago
  13. b4977de Receive-side ready for multiple channels. by Alex Loiko · 6 years ago
  14. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  15. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  16. bbf21a3 Remove dependencies on modules:module_api from AudioProcessing. by Fredrik Solenberg · 7 years ago
  17. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago[Renamed (98%) from voice_engine/utility.cc]
  18. 55900fd Move APM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  19. b63310a Remove VoEFile and things it uses. by solenberg · 7 years ago
  20. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 8 years ago
  21. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  22. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/voice_engine/utility.cc]
  23. de5ff8e Fix a variable naming typo by henrik.lundin · 8 years ago
  24. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  25. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  26. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  27. 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
  28. fb4f8b6 VoE Utility: Fix a naming nit in RemixAndResample by henrik.lundin · 8 years ago
  29. 228c268 Support 4 channel mic in Windows Core Audio by jens.nielsen · 8 years ago
  30. 6321b49 Move functionality out from AudioFrame and into AudioFrameOperations. by aleloi · 8 years ago
  31. af476c7 RTC_[D]CHECK_op: Remove "u" suffix on integer constants by kwiberg · 8 years ago
  32. 54e1c6a Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2018553002/ ) by Tommi · 9 years ago
  33. b4ff7a7 Revert of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2014183003/ ) by tommi · 9 years ago
  34. b7318f1 Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2006313009/ ) by tommi · 9 years ago
  35. 6895d8c Revert of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2009253004/ ) by kjellander · 9 years ago
  36. ba189cc Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2006243002/ ) by tommi · 9 years ago
  37. fb98b9e Revert of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #6 id:100001 of https://codereview.webrtc.org/2007563002/ ) by tommi · 9 years ago
  38. d36df89 Adding a some checks and switching out a few assert for RTC_[D]CHECK. by tommi · 9 years ago
  39. 6955870 Convert channel counts to size_t. by Peter Kasting · 9 years ago
  40. ad85622 Use webrtc/base/logging.h for voice_engine. by pbos · 9 years ago
  41. ff761fb modules: more interface -> include renames by Henrik Kjellander · 9 years ago
  42. 98f5351 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  43. cdfe20b Fix the maximum native sample rate in AudioProcessing by Alejandro Luebs · 9 years ago
  44. dce40cf Update a ton of audio code to use size_t more correctly and in general reduce by Peter Kasting · 10 years ago
  45. aada86b Add a simple AudioConverter class. by andrew@webrtc.org · 10 years ago
  46. 94454b7 Fix the chain that propagates the audio frame's rtp and ntp timestamp including: by wu@webrtc.org · 11 years ago
  47. 1fddd61 Add a Reset() method to AudioFrame. by andrew@webrtc.org · 11 years ago
  48. f5a33f1 Resampler modifications in preparation for arbitrary audioproc rates. by andrew@webrtc.org · 11 years ago
  49. 40ee3d0 Consolidate audio conversion from Channel and TransmitMixer. by andrew@webrtc.org · 11 years ago
  50. d900e8b Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 12 years ago
  51. 956aa7e Include files from webrtc/.. paths in voice_engine/ by pbos@webrtc.org · 12 years ago
  52. 6141e13 WebRtc_Word32 -> int32_t in voice_engine/ by pbos@webrtc.org · 12 years ago
  53. 14b43be Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago[Renamed from src/voice_engine/utility.cc]
  54. 6f8db36 Reorganize voice_engine/. by andrew@webrtc.org · 13 years ago[Renamed from src/voice_engine/main/source/utility.cc]
  55. d713143 To support playing mono file with stereo codec as mixing with microphone capture by braveyao@webrtc.org · 13 years ago
  56. c450a19 Removed Version function from all modules. by pwestin@webrtc.org · 13 years ago
  57. 470e71d by niklase@google.com · 14 years ago