Inherits WebRTCAudioLib, and IProcessor< short >.
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| WebRTCAudioProcessor (ILogger logger, int frameSize, int samplingRate, int channels, int reverseSamplingRate, int reverseChannels) |
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short[] | Process (short[] buf) |
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void | OnAudioOutFrameFloat (float[] data) |
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void | Dispose () |
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static IntPtr | webrtc_audio_processor_create (int samplingRate, int channels, int frameSize, int revSamplingRate, int revChannels) |
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static int | webrtc_audio_processor_init (IntPtr proc) |
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static int | webrtc_audio_processor_set_param (IntPtr proc, int param, int v) |
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static int | webrtc_audio_processor_process (IntPtr proc, short[] buffer, int offset, out bool voiceDetected) |
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static int | webrtc_audio_processor_process_reverse (IntPtr proc, short[] buffer, int bufferSize) |
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static void | webrtc_audio_processor_destroy (IntPtr proc) |
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T[] | Process (T[] buf) |
| Process a frame of data. More...
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static readonly int[] | SupportedSamplingRates = { 8000, 16000, 32000, 48000 } |
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int | AECStreamDelayMs [set] |
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bool?? | AEC [set] |
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bool? | AECHighPass [set] |
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bool?? | AECMobile [set] |
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bool? | HighPass [set] |
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bool? | NoiseSuppression [set] |
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bool? | AGC [set] |
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int | AGCCompressionGain [set] |
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int | AGCTargetLevel [set] |
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bool? | AGC2 [set] |
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bool? | VAD [set] |
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bool | Bypass [set] |
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