| /* |
| * 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_ |
