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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
#define MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
#include <stddef.h>
#include <stdint.h>
namespace webrtc {
enum {
kAgcModeUnchanged,
kAgcModeAdaptiveAnalog,
kAgcModeAdaptiveDigital,
kAgcModeFixedDigital
};
enum { kAgcFalse = 0, kAgcTrue };
typedef struct {
int16_t targetLevelDbfs; // default 3 (-3 dBOv)
int16_t compressionGaindB; // default 9 dB
uint8_t limiterEnable; // default kAgcTrue (on)
} WebRtcAgcConfig;
/*
* This function analyses the number of samples passed to
* farend and produces any error code that could arise.
*
* Input:
* - agcInst : AGC instance.
* - samples : Number of samples in input vector.
*
* Return value:
* : 0 - Normal operation.
* : -1 - Error.
*/
int WebRtcAgc_GetAddFarendError(void* state, size_t samples);
/*
* This function processes a 10 ms frame of far-end speech to determine
* if there is active speech. The length of the input speech vector must be
* given in samples (80 when FS=8000, and 160 when FS=16000, FS=32000 or
* FS=48000).
*
* Input:
* - agcInst : AGC instance.
* - inFar : Far-end input speech vector
* - samples : Number of samples in input vector
*
* Return value:
* : 0 - Normal operation.
* : -1 - Error
*/
int WebRtcAgc_AddFarend(void* agcInst, const int16_t* inFar, size_t samples);
/*
* This function processes a 10 ms frame of microphone speech to determine
* if there is active speech. The length of the input speech vector must be
* given in samples (80 when FS=8000, and 160 when FS=16000, FS=32000 or
* FS=48000). For very low input levels, the input signal is increased in level
* by multiplying and overwriting the samples in inMic[].
*
* This function should be called before any further processing of the
* near-end microphone signal.
*
* Input:
* - agcInst : AGC instance.
* - inMic : Microphone input speech vector for each band
* - num_bands : Number of bands in input vector
* - samples : Number of samples in input vector
*
* Return value:
* : 0 - Normal operation.
* : -1 - Error
*/
int WebRtcAgc_AddMic(void* agcInst,
int16_t* const* inMic,
size_t num_bands,
size_t samples);
/*
* This function replaces the analog microphone with a virtual one.
* It is a digital gain applied to the input signal and is used in the
* agcAdaptiveDigital mode where no microphone level is adjustable. The length
* of the input speech vector must be given in samples (80 when FS=8000, and 160
* when FS=16000, FS=32000 or FS=48000).
*
* Input:
* - agcInst : AGC instance.
* - inMic : Microphone input speech vector for each band
* - num_bands : Number of bands in input vector
* - samples : Number of samples in input vector
* - micLevelIn : Input level of microphone (static)
*
* Output:
* - inMic : Microphone output after processing (L band)
* - inMic_H : Microphone output after processing (H band)
* - micLevelOut : Adjusted microphone level after processing
*
* Return value:
* : 0 - Normal operation.
* : -1 - Error
*/
int WebRtcAgc_VirtualMic(void* agcInst,
int16_t* const* inMic,
size_t num_bands,
size_t samples,
int32_t micLevelIn,
int32_t* micLevelOut);
/*
* This function analyses a 10 ms frame and produces the analog and digital
* gains required to normalize the signal. The gain adjustments are done only
* during active periods of speech. The length of the speech vectors must be
* given in samples (80 when FS=8000, and 160 when FS=16000, FS=32000 or
* FS=48000). The echo parameter can be used to ensure the AGC will not adjust
* upward in the presence of echo.
*
* This function should be called after processing the near-end microphone
* signal, in any case after any echo cancellation.
*
* Input:
* - agcInst : AGC instance
* - inNear : Near-end input speech vector for each band
* - num_bands : Number of bands in input/output vector
* - samples : Number of samples in input/output vector
* - inMicLevel : Current microphone volume level
* - echo : Set to 0 if the signal passed to add_mic is
* almost certainly free of echo; otherwise set
* to 1. If you have no information regarding echo
* set to 0.
*
* Output:
* - outMicLevel : Adjusted microphone volume level
* - saturationWarning : A returned value of 1 indicates a saturation event
* has occurred and the volume cannot be further
* reduced. Otherwise will be set to 0.
* - gains : Vector of gains to apply for digital normalization
*
* Return value:
* : 0 - Normal operation.
* : -1 - Error
*/
int WebRtcAgc_Analyze(void* agcInst,
const int16_t* const* inNear,
size_t num_bands,
size_t samples,
int32_t inMicLevel,
int32_t* outMicLevel,
int16_t echo,
uint8_t* saturationWarning,
int32_t gains[11]);
/*
* This function processes a 10 ms frame by applying precomputed digital gains.
*
* Input:
* - agcInst : AGC instance
* - gains : Vector of gains to apply for digital normalization
* - in_near : Near-end input speech vector for each band
* - num_bands : Number of bands in input/output vector
*
* Output:
* - out : Gain-adjusted near-end speech vector
* : May be the same vector as the input.
*
* Return value:
* : 0 - Normal operation.
* : -1 - Error
*/
int WebRtcAgc_Process(const void* agcInst,
const int32_t gains[11],
const int16_t* const* in_near,
size_t num_bands,
int16_t* const* out);
/*
* This function sets the config parameters (targetLevelDbfs,
* compressionGaindB and limiterEnable).
*
* Input:
* - agcInst : AGC instance
* - config : config struct
*
* Output:
*
* Return value:
* : 0 - Normal operation.
* : -1 - Error
*/
int WebRtcAgc_set_config(void* agcInst, WebRtcAgcConfig config);
/*
* This function returns the config parameters (targetLevelDbfs,
* compressionGaindB and limiterEnable).
*
* Input:
* - agcInst : AGC instance
*
* Output:
* - config : config struct
*
* Return value:
* : 0 - Normal operation.
* : -1 - Error
*/
int WebRtcAgc_get_config(void* agcInst, WebRtcAgcConfig* config);
/*
* This function creates and returns an AGC instance, which will contain the
* state information for one (duplex) channel.
*/
void* WebRtcAgc_Create(void);
/*
* This function frees the AGC instance created at the beginning.
*
* Input:
* - agcInst : AGC instance.
*/
void WebRtcAgc_Free(void* agcInst);
/*
* This function initializes an AGC instance.
*
* Input:
* - agcInst : AGC instance.
* - minLevel : Minimum possible mic level
* - maxLevel : Maximum possible mic level
* - agcMode : 0 - Unchanged
* : 1 - Adaptive Analog Automatic Gain Control -3dBOv
* : 2 - Adaptive Digital Automatic Gain Control -3dBOv
* : 3 - Fixed Digital Gain 0dB
* - fs : Sampling frequency
*
* Return value : 0 - Ok
* -1 - Error
*/
int WebRtcAgc_Init(void* agcInst,
int32_t minLevel,
int32_t maxLevel,
int16_t agcMode,
uint32_t fs);
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_