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260 lines
9.0 KiB
C
260 lines
9.0 KiB
C
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
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#include "webrtc/typedefs.h"
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// Errors
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#define AGC_UNSPECIFIED_ERROR 18000
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#define AGC_UNSUPPORTED_FUNCTION_ERROR 18001
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#define AGC_UNINITIALIZED_ERROR 18002
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#define AGC_NULL_POINTER_ERROR 18003
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#define AGC_BAD_PARAMETER_ERROR 18004
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// Warnings
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#define AGC_BAD_PARAMETER_WARNING 18050
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enum
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{
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kAgcModeUnchanged,
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kAgcModeAdaptiveAnalog,
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kAgcModeAdaptiveDigital,
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kAgcModeFixedDigital
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};
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enum
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{
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kAgcFalse = 0,
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kAgcTrue
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};
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typedef struct
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{
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int16_t targetLevelDbfs; // default 3 (-3 dBOv)
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int16_t compressionGaindB; // default 9 dB
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uint8_t limiterEnable; // default kAgcTrue (on)
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} WebRtcAgc_config_t;
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#if defined(__cplusplus)
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extern "C"
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{
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#endif
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/*
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* This function processes a 10/20ms frame of far-end speech to determine
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* if there is active speech. Far-end speech length can be either 10ms or
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* 20ms. The length of the input speech vector must be given in samples
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* (80/160 when FS=8000, and 160/320 when FS=16000 or FS=32000).
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*
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* Input:
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* - agcInst : AGC instance.
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* - inFar : Far-end input speech vector (10 or 20ms)
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* - samples : Number of samples in input vector
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*
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* Return value:
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* : 0 - Normal operation.
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* : -1 - Error
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*/
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int WebRtcAgc_AddFarend(void* agcInst,
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const int16_t* inFar,
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int16_t samples);
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/*
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* This function processes a 10/20ms frame of microphone speech to determine
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* if there is active speech. Microphone speech length can be either 10ms or
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* 20ms. The length of the input speech vector must be given in samples
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* (80/160 when FS=8000, and 160/320 when FS=16000 or FS=32000). For very low
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* input levels, the input signal is increased in level by multiplying and
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* overwriting the samples in inMic[].
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*
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* This function should be called before any further processing of the
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* near-end microphone signal.
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*
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* Input:
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* - agcInst : AGC instance.
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* - inMic : Microphone input speech vector (10 or 20 ms) for
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* L band
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* - inMic_H : Microphone input speech vector (10 or 20 ms) for
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* H band
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* - samples : Number of samples in input vector
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*
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* Return value:
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* : 0 - Normal operation.
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* : -1 - Error
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*/
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int WebRtcAgc_AddMic(void* agcInst,
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int16_t* inMic,
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int16_t* inMic_H,
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int16_t samples);
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/*
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* This function replaces the analog microphone with a virtual one.
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* It is a digital gain applied to the input signal and is used in the
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* agcAdaptiveDigital mode where no microphone level is adjustable.
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* Microphone speech length can be either 10ms or 20ms. The length of the
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* input speech vector must be given in samples (80/160 when FS=8000, and
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* 160/320 when FS=16000 or FS=32000).
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*
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* Input:
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* - agcInst : AGC instance.
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* - inMic : Microphone input speech vector for (10 or 20 ms)
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* L band
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* - inMic_H : Microphone input speech vector for (10 or 20 ms)
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* H band
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* - samples : Number of samples in input vector
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* - micLevelIn : Input level of microphone (static)
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*
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* Output:
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* - inMic : Microphone output after processing (L band)
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* - inMic_H : Microphone output after processing (H band)
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* - micLevelOut : Adjusted microphone level after processing
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*
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* Return value:
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* : 0 - Normal operation.
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* : -1 - Error
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*/
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int WebRtcAgc_VirtualMic(void* agcInst,
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int16_t* inMic,
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int16_t* inMic_H,
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int16_t samples,
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int32_t micLevelIn,
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int32_t* micLevelOut);
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/*
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* This function processes a 10/20ms frame and adjusts (normalizes) the gain
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* both analog and digitally. The gain adjustments are done only during
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* active periods of speech. The input speech length can be either 10ms or
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* 20ms and the output is of the same length. The length of the speech
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* vectors must be given in samples (80/160 when FS=8000, and 160/320 when
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* FS=16000 or FS=32000). The echo parameter can be used to ensure the AGC will
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* not adjust upward in the presence of echo.
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*
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* This function should be called after processing the near-end microphone
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* signal, in any case after any echo cancellation.
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*
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* Input:
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* - agcInst : AGC instance
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* - inNear : Near-end input speech vector (10 or 20 ms) for
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* L band
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* - inNear_H : Near-end input speech vector (10 or 20 ms) for
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* H band
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* - samples : Number of samples in input/output vector
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* - inMicLevel : Current microphone volume level
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* - echo : Set to 0 if the signal passed to add_mic is
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* almost certainly free of echo; otherwise set
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* to 1. If you have no information regarding echo
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* set to 0.
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*
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* Output:
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* - outMicLevel : Adjusted microphone volume level
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* - out : Gain-adjusted near-end speech vector (L band)
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* : May be the same vector as the input.
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* - out_H : Gain-adjusted near-end speech vector (H band)
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* - saturationWarning : A returned value of 1 indicates a saturation event
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* has occurred and the volume cannot be further
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* reduced. Otherwise will be set to 0.
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*
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* Return value:
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* : 0 - Normal operation.
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* : -1 - Error
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*/
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int WebRtcAgc_Process(void* agcInst,
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const int16_t* inNear,
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const int16_t* inNear_H,
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int16_t samples,
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int16_t* out,
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int16_t* out_H,
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int32_t inMicLevel,
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int32_t* outMicLevel,
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int16_t echo,
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uint8_t* saturationWarning);
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/*
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* This function sets the config parameters (targetLevelDbfs,
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* compressionGaindB and limiterEnable).
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*
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* Input:
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* - agcInst : AGC instance
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* - config : config struct
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*
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* Output:
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*
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* Return value:
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* : 0 - Normal operation.
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* : -1 - Error
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*/
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int WebRtcAgc_set_config(void* agcInst, WebRtcAgc_config_t config);
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/*
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* This function returns the config parameters (targetLevelDbfs,
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* compressionGaindB and limiterEnable).
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*
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* Input:
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* - agcInst : AGC instance
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*
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* Output:
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* - config : config struct
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*
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* Return value:
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* : 0 - Normal operation.
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* : -1 - Error
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*/
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int WebRtcAgc_get_config(void* agcInst, WebRtcAgc_config_t* config);
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/*
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* This function creates an AGC instance, which will contain the state
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* information for one (duplex) channel.
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*
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* Return value : AGC instance if successful
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* : 0 (i.e., a NULL pointer) if unsuccessful
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*/
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int WebRtcAgc_Create(void **agcInst);
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/*
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* This function frees the AGC instance created at the beginning.
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*
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* Input:
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* - agcInst : AGC instance.
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*
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* Return value : 0 - Ok
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* -1 - Error
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*/
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int WebRtcAgc_Free(void *agcInst);
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/*
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* This function initializes an AGC instance.
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*
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* Input:
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* - agcInst : AGC instance.
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* - minLevel : Minimum possible mic level
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* - maxLevel : Maximum possible mic level
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* - agcMode : 0 - Unchanged
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* : 1 - Adaptive Analog Automatic Gain Control -3dBOv
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* : 2 - Adaptive Digital Automatic Gain Control -3dBOv
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* : 3 - Fixed Digital Gain 0dB
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* - fs : Sampling frequency
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*
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* Return value : 0 - Ok
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* -1 - Error
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*/
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int WebRtcAgc_Init(void *agcInst,
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int32_t minLevel,
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int32_t maxLevel,
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int16_t agcMode,
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uint32_t fs);
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#if defined(__cplusplus)
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}
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#endif
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
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