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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 2851916f57d2cedae9d99a0c3ab2769df55512d6 / modules / audio_processing / audio_processing_impl.cc
  1. 3fbe666 Add a Config parameter to AudioProcessing::Create(). by andrew@webrtc.org · 11 years ago
  2. e95dc25 Remove the requirement to call set_sample_rate_hz and friends. by andrew@webrtc.org · 11 years ago
  3. 7b72264 Allow opening an AEC dump from an existing file handle. by henrikg@webrtc.org · 11 years ago
  4. 09b40ec Add experimental noise suppression dummy API. by aluebs@webrtc.org · 11 years ago
  5. d7e9041 WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN. by andrew@webrtc.org · 11 years ago
  6. 80142aa Small refactoring of AudioProcessing use in channel.cc. by andrew@webrtc.org · 11 years ago
  7. 24add92 Fix some chromium-style warnings in webrtc/modules/audio_processing/ by pbos@webrtc.org · 11 years ago
  8. 25613ea Add a Config class interface to AudioProcessing for passing options. by andrew@webrtc.org · 11 years ago
  9. 3f6d5e0 WebRtc_Word32 -> int32_t in audio_processing/ by pbos@webrtc.org · 11 years ago
  10. 1d25eac Resolves TSan v2 reports data races in voe_auto_test. by henrika@webrtc.org · 11 years ago
  11. f1de5e9 Return an error when greater than 16 kHz is used with AECM. by andrew@webrtc.org · 11 years ago
  12. 7838d79 Only reinitialize AudioProcessing when needed. by andrew@webrtc.org · 12 years ago
  13. 5e512ae Qickly fixed android platform build breakage by leozwang@webrtc.org · 12 years ago
  14. b015cbe Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago