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Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/agc/legacy/gain_control.h')
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diff --git a/third_party/libwebrtc/modules/audio_processing/agc/legacy/gain_control.h b/third_party/libwebrtc/modules/audio_processing/agc/legacy/gain_control.h new file mode 100644 index 0000000000..6010a988fa --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/agc/legacy/gain_control.h @@ -0,0 +1,256 @@ +/* + * 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_ |