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gerrit-public.fairphone.software / platform / external / webrtc / 39cefdb3c5fb5b93f2681188871544a38a0c9efe / voice_engine / transmit_mixer.cc
  1. 39cefdb Revert "Reland "Remove WEBRTC_TRACE."" by Fredrik Solenberg · 7 years ago
  2. 68007e9 Reland "Remove WEBRTC_TRACE." by Fredrik Solenberg · 7 years ago
  3. 729b910 Revert "Remove WEBRTC_TRACE." by Fredrik Solenberg · 7 years ago
  4. 2209b90 Remove WEBRTC_TRACE. by Fredrik Solenberg · 7 years ago
  5. 1c239d4 Remove voe::Statistics. by solenberg · 7 years ago
  6. fc3a2e3 Remove the VoiceEngineObserver callback interface. by solenberg · 7 years ago
  7. b63310a Remove VoEFile and things it uses. by solenberg · 7 years ago
  8. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  9. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/voice_engine/transmit_mixer.cc]
  10. 3c45186 Move total audio energy and duration tracking to AudioLevel and protect with existing critial section. by zstein · 7 years ago
  11. e76bd3a Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy. by zstein · 7 years ago
  12. 950c1c9 TransmitMixer: Check GetSendCodec return value. by ossu · 7 years ago
  13. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  14. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  15. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  16. 36b1a5f Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide: by yujo · 7 years ago
  17. ec6fbd2 Moves channel-dependent audio input processing to separate encoder task queue. by henrika · 8 years ago
  18. fe7dd6d Remove VoEAudioProcessing interface. by solenberg · 8 years ago
  19. 8d73f8c Remove VoEVolumeControl interface. by solenberg · 8 years ago
  20. dea489f Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule. by tommi · 8 years ago
  21. ba08a14 Remove saturation warning support from TransmitMixer. by tommi · 8 years ago
  22. b1175bb Simplify webrtc::voe::MonitorModule and remove the .cc file. by tommi · 8 years ago
  23. 76377c5 Remove usage of VoEAudioProcessing from WVoE/MC. by solenberg · 8 years ago
  24. e374e01 Remove VoEExternalMedia interface. by solenberg · 8 years ago
  25. f00082d Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h by henrik.lundin · 8 years ago
  26. 6321b49 Move functionality out from AudioFrame and into AudioFrameOperations. by aleloi · 8 years ago
  27. 5b356f4 FilePlayer: Remove backwards compatibility stuff that we no longer need by kwiberg · 8 years ago
  28. 4ec01d9 Fix trivial lint errors in FileRecorder and FilePlayer by kwiberg · 8 years ago
  29. 5a25d95 FileRecorder + FilePlayer: Let Create functions return unique_ptr by kwiberg · 8 years ago
  30. 1c2af8e Avoid clicks when muting/unmuting a voe::Channel. by solenberg · 9 years ago
  31. 776593b Reland: Drop the 16kHz sample rate restriction on AECM and zero out higher bands by aluebs · 9 years ago
  32. b2a24ec Relanding https://codereview.webrtc.org/1715883002/ in pieces. by solenberg · 9 years ago
  33. dfc2870 Revert of Drop the 16kHz sample rate restriction on AECM and zero out higher bands (patchset #3 id:40001 of https://codereview.webrtc.org/1774553002/ ) by perkj · 9 years ago
  34. f687d53 Drop the 16kHz sample rate restriction on AECM and zero out higher bands by Alex Luebs · 9 years ago
  35. 3ecb5c8 Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ ) by solenberg · 9 years ago
  36. 8886c81 - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs. by solenberg · 9 years ago
  37. b7f89d6 Replace scoped_ptr with unique_ptr in webrtc/voice_engine/ by kwiberg · 9 years ago
  38. 31fc21f Swap use of CriticalSectionWrapper with rtc::CriticalSection in voice_engine/ 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. 302c978 Work around data race in TransmitMixer. by solenberg · 9 years ago
  42. ff761fb modules: more interface -> include renames by Henrik Kjellander · 9 years ago
  43. 98f5351 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  44. cdfe20b Fix the maximum native sample rate in AudioProcessing by Alejandro Luebs · 9 years ago
  45. d5c75b1 Reduce LS_INFO spam from voice_engine/. by Peter Boström · 9 years ago
  46. dce40cf Update a ton of audio code to use size_t more correctly and in general reduce by Peter Kasting · 9 years ago
  47. 3985f01 ProcessThread improvements. by tommi@webrtc.org · 10 years ago
  48. 00b8f6b Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away by kwiberg@webrtc.org · 10 years ago
  49. eec6ecd Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC. --- by tommi@webrtc.org · 10 years ago
  50. 6b02eea Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  51. e44a84d Only clamp to 16 kHz when AECM is enabled. by andrew@webrtc.org · 10 years ago
  52. 8f69330 Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
  53. 6680348 Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 11 years ago
  54. 40ee3d0 Consolidate audio conversion from Channel and TransmitMixer. by andrew@webrtc.org · 11 years ago
  55. 620d444 Extends max sample rate from 96kHz to 192kHz on the input side. by henrika@webrtc.org · 11 years ago
  56. 944cbeb Resolves TSan v2 warnings in voe_auto_test. by henrika@webrtc.org · 11 years ago
  57. 75dd288 Add an interface for accepting keypress signals to AudioProcessing. by andrew@webrtc.org · 11 years ago
  58. c693704 Move out typing detection to its own class. by henrikg@webrtc.org · 11 years ago
  59. 023cc5a Minor voice engine improvements around AGC. by andrew@webrtc.org · 11 years ago
  60. 60730cf Remove the requirement to call set_sample_rate_hz and friends. by andrew@webrtc.org · 11 years ago
  61. 6c264cc Clean up AudioProcessing defaults and errors. by andrew@webrtc.org · 11 years ago
  62. bf00740 Adds a new voice engine warning for the typing noise off state. by jiayl@webrtc.org · 11 years ago
  63. 676ff1e Ref-counted rewrite of ChannelManager. by pbos@webrtc.org · 11 years ago
  64. 8fff1f0 Merge r4394 from stable to trunk. by xians@webrtc.org · 11 years ago
  65. 2f84afa Merge r4326 from stable to trunk. by xians@webrtc.org · 11 years ago
  66. d900e8b Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
  67. 9213521 Remove const for plain data types in voice_engine/ by pbos@webrtc.org · 11 years ago
  68. 3be565b Refactoring for typing detection by niklas.enbom@webrtc.org · 11 years ago
  69. 28e82bf Replace Resampler with PushResampler in transmit_mixer. by andrew@webrtc.org · 11 years ago
  70. 6141e13 WebRtc_Word32 -> int32_t in voice_engine/ by pbos@webrtc.org · 12 years ago
  71. bb8ada6 TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC by henrika@webrtc.org · 12 years ago
  72. 2412085 Don't upsample the capture signal early. by andrew@webrtc.org · 12 years ago
  73. 6be1e93 Properly error check calls to AudioProcessing. by andrew@webrtc.org · 12 years ago
  74. ae1a58b Replace AudioFrame's operator= with CopyFrom(). by andrew@webrtc.org · 12 years ago
  75. a5e7e76 Use %d for signed value in trace. by andrew@webrtc.org · 12 years ago
  76. c862f49 Move capture level computation after all processing. by andrew@webrtc.org · 12 years ago
  77. 14b43be Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago[Renamed from src/voice_engine/transmit_mixer.cc]
  78. 21ab3ba Add external media hook for preprocessed audio. by andrew@webrtc.org · 12 years ago
  79. a7d8387 Opus integration by tina.legrand@webrtc.org · 12 years ago
  80. bdb03d4 Fix for issue 420 in TransmitMixer::SetTypingDetectionParameters. by vikasmarwaha@webrtc.org · 12 years ago
  81. 07ebdb9 Handle 96 kHz when downmixing the capture path. by andrew@webrtc.org · 12 years ago
  82. 6f8db36 Reorganize voice_engine/. by andrew@webrtc.org · 12 years ago[Renamed from src/voice_engine/main/source/transmit_mixer.cc]
  83. a9da4c5 Landing for thakis. Original review here: by tommi@webrtc.org · 12 years ago
  84. 4ecea3e Downmix before resampling in capture and render paths. by andrew@webrtc.org · 12 years ago
  85. 4de777b Refine the error processing of StopRecordingMicrophone. by braveyao@webrtc.org · 12 years ago
  86. ee646c3 I know this is ugly, but it helps a lot to quickly update webRTC in Chrome and libJingle. by niklas.enbom@webrtc.org · 12 years ago
  87. 7fbfc4c Use correct variable in trace. by andrew@webrtc.org · 12 years ago
  88. f6edfef Adding one parameter to typing detection tuning by niklas.enbom@webrtc.org · 12 years ago
  89. e59a0ac Fix AudioFrame types. by andrew@webrtc.org · 12 years ago
  90. 63a5098 Rename AudioFrame members. by andrew@webrtc.org · 12 years ago
  91. 02d7174 Add API to swap stereo channels. by andrew@webrtc.org · 12 years ago
  92. 06e722a Adding parameter setting for typing detection by niklas.enbom@webrtc.org · 13 years ago
  93. 3dc8865 Adding time since last typing by niklas.enbom@webrtc.org · 13 years ago
  94. d713143 To support playing mono file with stereo codec as mixing with microphone capture by braveyao@webrtc.org · 13 years ago
  95. 907bc55 Removes WebRtc_Word8 dependecy in the AudioDeviceModule. by henrika@webrtc.org · 13 years ago
  96. 9a065d1 VoiceEngine now uses pointer constructor of CriticalSectionScoped, instead of reference. by mflodman@webrtc.org · 13 years ago
  97. 813e4b0 Correct WebRtc_Word8 in voice engine by leozwang@webrtc.org · 13 years ago
  98. 87885e8 This CL will look a bit strange. Essentially I've removed sanity checks for > 1 channel and then fixed the bugs that remained. Will add testing in a separate CL. by niklas.enbom@webrtc.org · 13 years ago
  99. 3192d65 Fix for devices lacking stereo support. by andrew@webrtc.org · 13 years ago
  100. af71f0e Fixes two minor issues reported by the Coverty Integration Manager. by henrika@webrtc.org · 13 years ago