cos / mirrors / cros / chromiumos / third_party / webrtc-apm / 640d48b3222a28a7cb84f9df30d35d236582fc5c / . / modules / audio_coding / neteq / dsp_helper.h

/* | |

* 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_CODING_NETEQ_DSP_HELPER_H_ | |

#define MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_ | |

#include <stdint.h> | |

#include <string.h> | |

#include "modules/audio_coding/neteq/audio_multi_vector.h" | |

#include "modules/audio_coding/neteq/audio_vector.h" | |

#include "rtc_base/constructormagic.h" | |

namespace webrtc { | |

// This class contains various signal processing functions, all implemented as | |

// static methods. | |

class DspHelper { | |

public: | |

// Filter coefficients used when downsampling from the indicated sample rates | |

// (8, 16, 32, 48 kHz) to 4 kHz. Coefficients are in Q12. | |

static const int16_t kDownsample8kHzTbl[3]; | |

static const int16_t kDownsample16kHzTbl[5]; | |

static const int16_t kDownsample32kHzTbl[7]; | |

static const int16_t kDownsample48kHzTbl[7]; | |

// Constants used to mute and unmute over 5 samples. The coefficients are | |

// in Q15. | |

static const int kMuteFactorStart8kHz = 27307; | |

static const int kMuteFactorIncrement8kHz = -5461; | |

static const int kUnmuteFactorStart8kHz = 5461; | |

static const int kUnmuteFactorIncrement8kHz = 5461; | |

static const int kMuteFactorStart16kHz = 29789; | |

static const int kMuteFactorIncrement16kHz = -2979; | |

static const int kUnmuteFactorStart16kHz = 2979; | |

static const int kUnmuteFactorIncrement16kHz = 2979; | |

static const int kMuteFactorStart32kHz = 31208; | |

static const int kMuteFactorIncrement32kHz = -1560; | |

static const int kUnmuteFactorStart32kHz = 1560; | |

static const int kUnmuteFactorIncrement32kHz = 1560; | |

static const int kMuteFactorStart48kHz = 31711; | |

static const int kMuteFactorIncrement48kHz = -1057; | |

static const int kUnmuteFactorStart48kHz = 1057; | |

static const int kUnmuteFactorIncrement48kHz = 1057; | |

// Multiplies the signal with a gradually changing factor. | |

// The first sample is multiplied with |factor| (in Q14). For each sample, | |

// |factor| is increased (additive) by the |increment| (in Q20), which can | |

// be negative. Returns the scale factor after the last increment. | |

static int RampSignal(const int16_t* input, | |

size_t length, | |

int factor, | |

int increment, | |

int16_t* output); | |

// Same as above, but with the samples of |signal| being modified in-place. | |

static int RampSignal(int16_t* signal, | |

size_t length, | |

int factor, | |

int increment); | |

// Same as above, but processes |length| samples from |signal|, starting at | |

// |start_index|. | |

static int RampSignal(AudioVector* signal, | |

size_t start_index, | |

size_t length, | |

int factor, | |

int increment); | |

// Same as above, but for an AudioMultiVector. | |

static int RampSignal(AudioMultiVector* signal, | |

size_t start_index, | |

size_t length, | |

int factor, | |

int increment); | |

// Peak detection with parabolic fit. Looks for |num_peaks| maxima in |data|, | |

// having length |data_length| and sample rate multiplier |fs_mult|. The peak | |

// locations and values are written to the arrays |peak_index| and | |

// |peak_value|, respectively. Both arrays must hold at least |num_peaks| | |

// elements. | |

static void PeakDetection(int16_t* data, | |

size_t data_length, | |

size_t num_peaks, | |

int fs_mult, | |

size_t* peak_index, | |

int16_t* peak_value); | |

// Estimates the height and location of a maximum. The three values in the | |

// array |signal_points| are used as basis for a parabolic fit, which is then | |

// used to find the maximum in an interpolated signal. The |signal_points| are | |

// assumed to be from a 4 kHz signal, while the maximum, written to | |

// |peak_index| and |peak_value| is given in the full sample rate, as | |

// indicated by the sample rate multiplier |fs_mult|. | |

static void ParabolicFit(int16_t* signal_points, | |

int fs_mult, | |

size_t* peak_index, | |

int16_t* peak_value); | |

// Calculates the sum-abs-diff for |signal| when compared to a displaced | |

// version of itself. Returns the displacement lag that results in the minimum | |

// distortion. The resulting distortion is written to |distortion_value|. | |

// The values of |min_lag| and |max_lag| are boundaries for the search. | |

static size_t MinDistortion(const int16_t* signal, | |

size_t min_lag, | |

size_t max_lag, | |

size_t length, | |

int32_t* distortion_value); | |

// Mixes |length| samples from |input1| and |input2| together and writes the | |

// result to |output|. The gain for |input1| starts at |mix_factor| (Q14) and | |

// is decreased by |factor_decrement| (Q14) for each sample. The gain for | |

// |input2| is the complement 16384 - mix_factor. | |

static void CrossFade(const int16_t* input1, | |

const int16_t* input2, | |

size_t length, | |

int16_t* mix_factor, | |

int16_t factor_decrement, | |

int16_t* output); | |

// Scales |input| with an increasing gain. Applies |factor| (Q14) to the first | |

// sample and increases the gain by |increment| (Q20) for each sample. The | |

// result is written to |output|. |length| samples are processed. | |

static void UnmuteSignal(const int16_t* input, | |

size_t length, | |

int16_t* factor, | |

int increment, | |

int16_t* output); | |

// Starts at unity gain and gradually fades out |signal|. For each sample, | |

// the gain is reduced by |mute_slope| (Q14). |length| samples are processed. | |

static void MuteSignal(int16_t* signal, int mute_slope, size_t length); | |

// Downsamples |input| from |sample_rate_hz| to 4 kHz sample rate. The input | |

// has |input_length| samples, and the method will write |output_length| | |

// samples to |output|. Compensates for the phase delay of the downsampling | |

// filters if |compensate_delay| is true. Returns -1 if the input is too short | |

// to produce |output_length| samples, otherwise 0. | |

static int DownsampleTo4kHz(const int16_t* input, | |

size_t input_length, | |

size_t output_length, | |

int input_rate_hz, | |

bool compensate_delay, | |

int16_t* output); | |

private: | |

// Table of constants used in method DspHelper::ParabolicFit(). | |

static const int16_t kParabolaCoefficients[17][3]; | |

RTC_DISALLOW_COPY_AND_ASSIGN(DspHelper); | |

}; | |

} // namespace webrtc | |

#endif // MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_ |