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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
12#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
13
14#include <string.h>  // Access to size_t.
15
16#include "webrtc/base/constructormagic.h"
17#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
18#include "webrtc/typedefs.h"
19
20namespace webrtc {
21
22// This class contains various signal processing functions, all implemented as
23// static methods.
24class 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(AudioMultiVector* signal,
71                        size_t start_index,
72                        size_t length,
73                        int factor,
74                        int increment);
75
76  // Peak detection with parabolic fit. Looks for |num_peaks| maxima in |data|,
77  // having length |data_length| and sample rate multiplier |fs_mult|. The peak
78  // locations and values are written to the arrays |peak_index| and
79  // |peak_value|, respectively. Both arrays must hold at least |num_peaks|
80  // elements.
81  static void PeakDetection(int16_t* data, int data_length,
82                            int num_peaks, int fs_mult,
83                            int* peak_index, int16_t* peak_value);
84
85  // Estimates the height and location of a maximum. The three values in the
86  // array |signal_points| are used as basis for a parabolic fit, which is then
87  // used to find the maximum in an interpolated signal. The |signal_points| are
88  // assumed to be from a 4 kHz signal, while the maximum, written to
89  // |peak_index| and |peak_value| is given in the full sample rate, as
90  // indicated by the sample rate multiplier |fs_mult|.
91  static void ParabolicFit(int16_t* signal_points, int fs_mult,
92                           int* peak_index, int16_t* peak_value);
93
94  // Calculates the sum-abs-diff for |signal| when compared to a displaced
95  // version of itself. Returns the displacement lag that results in the minimum
96  // distortion. The resulting distortion is written to |distortion_value|.
97  // The values of |min_lag| and |max_lag| are boundaries for the search.
98  static int MinDistortion(const int16_t* signal, int min_lag,
99                           int max_lag, int length, int32_t* distortion_value);
100
101  // Mixes |length| samples from |input1| and |input2| together and writes the
102  // result to |output|. The gain for |input1| starts at |mix_factor| (Q14) and
103  // is decreased by |factor_decrement| (Q14) for each sample. The gain for
104  // |input2| is the complement 16384 - mix_factor.
105  static void CrossFade(const int16_t* input1, const int16_t* input2,
106                        size_t length, int16_t* mix_factor,
107                        int16_t factor_decrement, int16_t* output);
108
109  // Scales |input| with an increasing gain. Applies |factor| (Q14) to the first
110  // sample and increases the gain by |increment| (Q20) for each sample. The
111  // result is written to |output|. |length| samples are processed.
112  static void UnmuteSignal(const int16_t* input, size_t length, int16_t* factor,
113                           int16_t increment, int16_t* output);
114
115  // Starts at unity gain and gradually fades out |signal|. For each sample,
116  // the gain is reduced by |mute_slope| (Q14). |length| samples are processed.
117  static void MuteSignal(int16_t* signal, int16_t mute_slope, size_t length);
118
119  // Downsamples |input| from |sample_rate_hz| to 4 kHz sample rate. The input
120  // has |input_length| samples, and the method will write |output_length|
121  // samples to |output|. Compensates for the phase delay of the downsampling
122  // filters if |compensate_delay| is true. Returns -1 if the input is too short
123  // to produce |output_length| samples, otherwise 0.
124  static int DownsampleTo4kHz(const int16_t* input, size_t input_length,
125                              int output_length, int input_rate_hz,
126                              bool compensate_delay, int16_t* output);
127
128 private:
129  // Table of constants used in method DspHelper::ParabolicFit().
130  static const int16_t kParabolaCoefficients[17][3];
131
132  DISALLOW_COPY_AND_ASSIGN(DspHelper);
133};
134
135}  // namespace webrtc
136#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
137