mirror of
https://github.com/oxen-io/session-android.git
synced 2024-12-18 14:07:30 +00:00
260 lines
9.0 KiB
C
260 lines
9.0 KiB
C
/*
|
|
* 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 WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
|
|
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
|
|
|
|
#include "webrtc/typedefs.h"
|
|
|
|
// Errors
|
|
#define AGC_UNSPECIFIED_ERROR 18000
|
|
#define AGC_UNSUPPORTED_FUNCTION_ERROR 18001
|
|
#define AGC_UNINITIALIZED_ERROR 18002
|
|
#define AGC_NULL_POINTER_ERROR 18003
|
|
#define AGC_BAD_PARAMETER_ERROR 18004
|
|
|
|
// Warnings
|
|
#define AGC_BAD_PARAMETER_WARNING 18050
|
|
|
|
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)
|
|
} WebRtcAgc_config_t;
|
|
|
|
#if defined(__cplusplus)
|
|
extern "C"
|
|
{
|
|
#endif
|
|
|
|
/*
|
|
* This function processes a 10/20ms frame of far-end speech to determine
|
|
* if there is active speech. Far-end speech length can be either 10ms or
|
|
* 20ms. The length of the input speech vector must be given in samples
|
|
* (80/160 when FS=8000, and 160/320 when FS=16000 or FS=32000).
|
|
*
|
|
* Input:
|
|
* - agcInst : AGC instance.
|
|
* - inFar : Far-end input speech vector (10 or 20ms)
|
|
* - samples : Number of samples in input vector
|
|
*
|
|
* Return value:
|
|
* : 0 - Normal operation.
|
|
* : -1 - Error
|
|
*/
|
|
int WebRtcAgc_AddFarend(void* agcInst,
|
|
const int16_t* inFar,
|
|
int16_t samples);
|
|
|
|
/*
|
|
* This function processes a 10/20ms frame of microphone speech to determine
|
|
* if there is active speech. Microphone speech length can be either 10ms or
|
|
* 20ms. The length of the input speech vector must be given in samples
|
|
* (80/160 when FS=8000, and 160/320 when FS=16000 or FS=32000). 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 (10 or 20 ms) for
|
|
* L band
|
|
* - inMic_H : Microphone input speech vector (10 or 20 ms) for
|
|
* H band
|
|
* - samples : Number of samples in input vector
|
|
*
|
|
* Return value:
|
|
* : 0 - Normal operation.
|
|
* : -1 - Error
|
|
*/
|
|
int WebRtcAgc_AddMic(void* agcInst,
|
|
int16_t* inMic,
|
|
int16_t* inMic_H,
|
|
int16_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.
|
|
* Microphone speech length can be either 10ms or 20ms. The length of the
|
|
* input speech vector must be given in samples (80/160 when FS=8000, and
|
|
* 160/320 when FS=16000 or FS=32000).
|
|
*
|
|
* Input:
|
|
* - agcInst : AGC instance.
|
|
* - inMic : Microphone input speech vector for (10 or 20 ms)
|
|
* L band
|
|
* - inMic_H : Microphone input speech vector for (10 or 20 ms)
|
|
* H band
|
|
* - 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* inMic,
|
|
int16_t* inMic_H,
|
|
int16_t samples,
|
|
int32_t micLevelIn,
|
|
int32_t* micLevelOut);
|
|
|
|
/*
|
|
* This function processes a 10/20ms frame and adjusts (normalizes) the gain
|
|
* both analog and digitally. The gain adjustments are done only during
|
|
* active periods of speech. The input speech length can be either 10ms or
|
|
* 20ms and the output is of the same length. The length of the speech
|
|
* vectors must be given in samples (80/160 when FS=8000, and 160/320 when
|
|
* FS=16000 or FS=32000). 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 (10 or 20 ms) for
|
|
* L band
|
|
* - inNear_H : Near-end input speech vector (10 or 20 ms) for
|
|
* H band
|
|
* - 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
|
|
* - out : Gain-adjusted near-end speech vector (L band)
|
|
* : May be the same vector as the input.
|
|
* - out_H : Gain-adjusted near-end speech vector (H band)
|
|
* - 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.
|
|
*
|
|
* Return value:
|
|
* : 0 - Normal operation.
|
|
* : -1 - Error
|
|
*/
|
|
int WebRtcAgc_Process(void* agcInst,
|
|
const int16_t* inNear,
|
|
const int16_t* inNear_H,
|
|
int16_t samples,
|
|
int16_t* out,
|
|
int16_t* out_H,
|
|
int32_t inMicLevel,
|
|
int32_t* outMicLevel,
|
|
int16_t echo,
|
|
uint8_t* saturationWarning);
|
|
|
|
/*
|
|
* 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, WebRtcAgc_config_t 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, WebRtcAgc_config_t* config);
|
|
|
|
/*
|
|
* This function creates an AGC instance, which will contain the state
|
|
* information for one (duplex) channel.
|
|
*
|
|
* Return value : AGC instance if successful
|
|
* : 0 (i.e., a NULL pointer) if unsuccessful
|
|
*/
|
|
int WebRtcAgc_Create(void **agcInst);
|
|
|
|
/*
|
|
* This function frees the AGC instance created at the beginning.
|
|
*
|
|
* Input:
|
|
* - agcInst : AGC instance.
|
|
*
|
|
* Return value : 0 - Ok
|
|
* -1 - Error
|
|
*/
|
|
int 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);
|
|
|
|
#if defined(__cplusplus)
|
|
}
|
|
#endif
|
|
|
|
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
|