254 lines
8.9 KiB
C++
254 lines
8.9 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 MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
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#define MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
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namespace webrtc {
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enum {
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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 { kAgcFalse = 0, kAgcTrue };
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typedef struct {
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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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} WebRtcAgcConfig;
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/*
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* This function analyses the number of samples passed to
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* farend and produces any error code that could arise.
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*
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* Input:
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* - agcInst : AGC instance.
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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_GetAddFarendError(void* state, size_t samples);
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/*
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* This function processes a 10 ms frame of far-end speech to determine
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* if there is active speech. The length of the input speech vector must be
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* given in samples (80 when FS=8000, and 160 when FS=16000, FS=32000 or
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* FS=48000).
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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
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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, const int16_t* inFar, size_t samples);
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/*
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* This function processes a 10 ms frame of microphone speech to determine
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* if there is active speech. The length of the input speech vector must be
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* given in samples (80 when FS=8000, and 160 when FS=16000, FS=32000 or
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* FS=48000). For very low input levels, the input signal is increased in level
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* by multiplying and 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 for each band
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* - num_bands : Number of bands in input vector
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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* const* inMic,
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size_t num_bands,
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size_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. The length
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* of the input speech vector must be given in samples (80 when FS=8000, and 160
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* when FS=16000, FS=32000 or FS=48000).
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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 each band
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* - num_bands : Number of bands in input vector
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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* const* inMic,
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size_t num_bands,
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size_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 analyses a 10 ms frame and produces the analog and digital
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* gains required to normalize the signal. The gain adjustments are done only
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* during active periods of speech. The length of the speech vectors must be
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* given in samples (80 when FS=8000, and 160 when FS=16000, FS=32000 or
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* FS=48000). The echo parameter can be used to ensure the AGC will not adjust
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* 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 for each band
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* - num_bands : Number of bands in input/output vector
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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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* - 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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* - gains : Vector of gains to apply for digital normalization
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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_Analyze(void* agcInst,
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const int16_t* const* inNear,
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size_t num_bands,
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size_t samples,
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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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int32_t gains[11]);
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/*
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* This function processes a 10 ms frame by applying precomputed digital gains.
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*
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* Input:
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* - agcInst : AGC instance
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* - gains : Vector of gains to apply for digital normalization
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* - in_near : Near-end input speech vector for each band
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* - num_bands : Number of bands in input/output vector
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*
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* Output:
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* - out : Gain-adjusted near-end speech vector
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* : May be the same vector as the input.
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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(const void* agcInst,
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const int32_t gains[11],
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const int16_t* const* in_near,
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size_t num_bands,
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int16_t* const* out);
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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, WebRtcAgcConfig 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, WebRtcAgcConfig* config);
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/*
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* This function creates and returns an AGC instance, which will contain the
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* state information for one (duplex) channel.
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*/
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void* WebRtcAgc_Create(void);
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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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void 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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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
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