/* * 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 "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 { WebRtc_Word16 targetLevelDbfs; // default 3 (-3 dBOv) WebRtc_Word16 compressionGaindB; // default 9 dB WebRtc_UWord8 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 WebRtc_Word16* inFar, WebRtc_Word16 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, WebRtc_Word16* inMic, WebRtc_Word16* inMic_H, WebRtc_Word16 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, WebRtc_Word16* inMic, WebRtc_Word16* inMic_H, WebRtc_Word16 samples, WebRtc_Word32 micLevelIn, WebRtc_Word32* 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 WebRtc_Word16* inNear, const WebRtc_Word16* inNear_H, WebRtc_Word16 samples, WebRtc_Word16* out, WebRtc_Word16* out_H, WebRtc_Word32 inMicLevel, WebRtc_Word32* outMicLevel, WebRtc_Word16 echo, WebRtc_UWord8* 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, WebRtc_Word32 minLevel, WebRtc_Word32 maxLevel, WebRtc_Word16 agcMode, WebRtc_UWord32 fs); #if defined(__cplusplus) } #endif #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_