1 /* 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_ 12 #define MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_ 13 14 #include <string.h> // Access to size_t. 15 16 #include "modules/audio_coding/neteq/audio_multi_vector.h" 17 #include "rtc_base/constructormagic.h" 18 #include "typedefs.h" // NOLINT(build/include) 19 20 namespace webrtc { 21 22 // This class contains various signal processing functions, all implemented as 23 // static methods. 24 class DspHelper { 25 public: 26 // Filter coefficients used when downsampling from the indicated sample rates 27 // (8, 16, 32, 48 kHz) to 4 kHz. Coefficients are in Q12. 28 static const int16_t kDownsample8kHzTbl[3]; 29 static const int16_t kDownsample16kHzTbl[5]; 30 static const int16_t kDownsample32kHzTbl[7]; 31 static const int16_t kDownsample48kHzTbl[7]; 32 33 // Constants used to mute and unmute over 5 samples. The coefficients are 34 // in Q15. 35 static const int kMuteFactorStart8kHz = 27307; 36 static const int kMuteFactorIncrement8kHz = -5461; 37 static const int kUnmuteFactorStart8kHz = 5461; 38 static const int kUnmuteFactorIncrement8kHz = 5461; 39 static const int kMuteFactorStart16kHz = 29789; 40 static const int kMuteFactorIncrement16kHz = -2979; 41 static const int kUnmuteFactorStart16kHz = 2979; 42 static const int kUnmuteFactorIncrement16kHz = 2979; 43 static const int kMuteFactorStart32kHz = 31208; 44 static const int kMuteFactorIncrement32kHz = -1560; 45 static const int kUnmuteFactorStart32kHz = 1560; 46 static const int kUnmuteFactorIncrement32kHz = 1560; 47 static const int kMuteFactorStart48kHz = 31711; 48 static const int kMuteFactorIncrement48kHz = -1057; 49 static const int kUnmuteFactorStart48kHz = 1057; 50 static const int kUnmuteFactorIncrement48kHz = 1057; 51 52 // Multiplies the signal with a gradually changing factor. 53 // The first sample is multiplied with |factor| (in Q14). For each sample, 54 // |factor| is increased (additive) by the |increment| (in Q20), which can 55 // be negative. Returns the scale factor after the last increment. 56 static int RampSignal(const int16_t* input, 57 size_t length, 58 int factor, 59 int increment, 60 int16_t* output); 61 62 // Same as above, but with the samples of |signal| being modified in-place. 63 static int RampSignal(int16_t* signal, 64 size_t length, 65 int factor, 66 int increment); 67 68 // Same as above, but processes |length| samples from |signal|, starting at 69 // |start_index|. 70 static int RampSignal(AudioVector* signal, 71 size_t start_index, 72 size_t length, 73 int factor, 74 int increment); 75 76 // Same as above, but for an AudioMultiVector. 77 static int RampSignal(AudioMultiVector* signal, 78 size_t start_index, 79 size_t length, 80 int factor, 81 int increment); 82 83 // Peak detection with parabolic fit. Looks for |num_peaks| maxima in |data|, 84 // having length |data_length| and sample rate multiplier |fs_mult|. The peak 85 // locations and values are written to the arrays |peak_index| and 86 // |peak_value|, respectively. Both arrays must hold at least |num_peaks| 87 // elements. 88 static void PeakDetection(int16_t* data, size_t data_length, 89 size_t num_peaks, int fs_mult, 90 size_t* peak_index, int16_t* peak_value); 91 92 // Estimates the height and location of a maximum. The three values in the 93 // array |signal_points| are used as basis for a parabolic fit, which is then 94 // used to find the maximum in an interpolated signal. The |signal_points| are 95 // assumed to be from a 4 kHz signal, while the maximum, written to 96 // |peak_index| and |peak_value| is given in the full sample rate, as 97 // indicated by the sample rate multiplier |fs_mult|. 98 static void ParabolicFit(int16_t* signal_points, int fs_mult, 99 size_t* peak_index, int16_t* peak_value); 100 101 // Calculates the sum-abs-diff for |signal| when compared to a displaced 102 // version of itself. Returns the displacement lag that results in the minimum 103 // distortion. The resulting distortion is written to |distortion_value|. 104 // The values of |min_lag| and |max_lag| are boundaries for the search. 105 static size_t MinDistortion(const int16_t* signal, size_t min_lag, 106 size_t max_lag, size_t length, 107 int32_t* distortion_value); 108 109 // Mixes |length| samples from |input1| and |input2| together and writes the 110 // result to |output|. The gain for |input1| starts at |mix_factor| (Q14) and 111 // is decreased by |factor_decrement| (Q14) for each sample. The gain for 112 // |input2| is the complement 16384 - mix_factor. 113 static void CrossFade(const int16_t* input1, const int16_t* input2, 114 size_t length, int16_t* mix_factor, 115 int16_t factor_decrement, int16_t* output); 116 117 // Scales |input| with an increasing gain. Applies |factor| (Q14) to the first 118 // sample and increases the gain by |increment| (Q20) for each sample. The 119 // result is written to |output|. |length| samples are processed. 120 static void UnmuteSignal(const int16_t* input, size_t length, int16_t* factor, 121 int increment, int16_t* output); 122 123 // Starts at unity gain and gradually fades out |signal|. For each sample, 124 // the gain is reduced by |mute_slope| (Q14). |length| samples are processed. 125 static void MuteSignal(int16_t* signal, int mute_slope, size_t length); 126 127 // Downsamples |input| from |sample_rate_hz| to 4 kHz sample rate. The input 128 // has |input_length| samples, and the method will write |output_length| 129 // samples to |output|. Compensates for the phase delay of the downsampling 130 // filters if |compensate_delay| is true. Returns -1 if the input is too short 131 // to produce |output_length| samples, otherwise 0. 132 static int DownsampleTo4kHz(const int16_t* input, size_t input_length, 133 size_t output_length, int input_rate_hz, 134 bool compensate_delay, int16_t* output); 135 136 private: 137 // Table of constants used in method DspHelper::ParabolicFit(). 138 static const int16_t kParabolaCoefficients[17][3]; 139 140 RTC_DISALLOW_COPY_AND_ASSIGN(DspHelper); 141 }; 142 143 } // namespace webrtc 144 #endif // MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_ 145