1 | /* |
2 | Copyright 2018 Google Inc. All Rights Reserved. |
3 | |
4 | Licensed under the Apache License, Version 2.0 (the "License"); |
5 | you may not use this file except in compliance with the License. |
6 | You may obtain a copy of the License at |
7 | |
8 | http://www.apache.org/licenses/LICENSE-2.0 |
9 | |
10 | Unless required by applicable law or agreed to in writing, software |
11 | distributed under the License is distributed on an "AS-IS" BASIS, |
12 | WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
13 | See the License for the specific language governing permissions and |
14 | limitations under the License. |
15 | */ |
16 | |
17 | #ifndef RESONANCE_AUDIO_DSP_RESAMPLER_H_ |
18 | #define RESONANCE_AUDIO_DSP_RESAMPLER_H_ |
19 | |
20 | #include "base/audio_buffer.h" |
21 | |
22 | namespace vraudio { |
23 | |
24 | // Class that provides rational resampling of audio data. |
25 | class Resampler { |
26 | public: |
27 | Resampler(); |
28 | |
29 | // Resamples an |AudioBuffer| of input data sampled at |source_frequency| to |
30 | // |destination_frequency|. |
31 | // |
32 | // @param input Input data to be resampled. |
33 | // @param output Resampled output data. |
34 | void Process(const AudioBuffer& input, AudioBuffer* output); |
35 | |
36 | // Returns the maximum length which the output buffer will be, given the |
37 | // current source and destination frequencies and input length. The actual |
38 | // output length will either be this or one less. |
39 | // |
40 | // @param input_length Length of the input. |
41 | // @return Maximum length of the output. |
42 | size_t GetMaxOutputLength(size_t input_length) const; |
43 | |
44 | // Returns the next length which the output buffer will be, given the |
45 | // current source and destination frequencies and input length. |
46 | // |
47 | // @param input_length Length of the input. |
48 | // @return Next length of the output. |
49 | size_t GetNextOutputLength(size_t input_length) const; |
50 | |
51 | // Sets the source and destination sampling rate as well as the number of |
52 | // channels. Note this method only resets the filter state number of channel |
53 | // changes. |
54 | // |
55 | // @param source_frequency Sampling rate of input data. |
56 | // @param destination_frequency Desired output sampling rate. |
57 | // @param num_channels Number of channels to process. |
58 | void SetRateAndNumChannels(int source_frequency, int destination_frequency, |
59 | size_t num_channels); |
60 | |
61 | // Returns whether the sampling rates provided are supported by the resampler. |
62 | // |
63 | // @param source Source sampling rate. |
64 | // @param destination Destination sampling rate. |
65 | // @return True if the sampling rate pair are supported. |
66 | static bool AreSampleRatesSupported(int source, int destination); |
67 | |
68 | // Resets the inner state of the |Resampler| allowing its use repeatedly on |
69 | // different data streams. |
70 | void ResetState(); |
71 | |
72 | private: |
73 | friend class PolyphaseFilterTest; |
74 | // Initializes the |state_| buffer. Called when sampling rate is changed or |
75 | // the state is reset. |
76 | // |
77 | // @param size_t old_state_num_frames Number of frames in the |state_| buffer |
78 | // previous to the most recent call to |GenerateInterpolatingFilter|. |
79 | void InitializeStateBuffer(size_t old_state_num_frames); |
80 | |
81 | // Generates a windowed sinc to act as the interpolating/anti-aliasing filter. |
82 | // |
83 | // @param sample_rate The system sampling rate. |
84 | void GenerateInterpolatingFilter(int sample_rate); |
85 | |
86 | // Arranges the anti aliasing filter coefficients in polyphase filter format. |
87 | // |
88 | // @param filter_length Number of frames in |filter| containing filter |
89 | // coefficients. |
90 | // @param filter Vector of filter coefficients. |
91 | void ArrangeFilterAsPolyphase(size_t filter_length, |
92 | const AudioBuffer::Channel& filter); |
93 | |
94 | // Generates Hann windowed sinc function anti aliasing filters. |
95 | // |
96 | // @param cutoff_frequency Transition band (-3dB) frequency of the filter. |
97 | // @param sample_rate The system sampling rate. |
98 | // @param filter_length Number of frames in |buffer| containing filter |
99 | // coefficients. |
100 | // @param buffer |AudioBuffer::Channel| to contain the filter coefficients. |
101 | void GenerateSincFilter(float cutoff_frequency, float sample_rate, |
102 | size_t filter_length, AudioBuffer::Channel* buffer); |
103 | |
104 | // Rate of the interpolator section of the rational sampling rate converter. |
105 | size_t up_rate_; |
106 | |
107 | // Rate of the decimator section of the rational sampling rate convereter. |
108 | size_t down_rate_; |
109 | |
110 | // Time variable for the polyphase filter. |
111 | size_t time_modulo_up_rate_; |
112 | |
113 | // Marks the last processed sample of the input. |
114 | size_t last_processed_sample_; |
115 | |
116 | // Number of channels in the |AudioBuffer|s processed. |
117 | size_t num_channels_; |
118 | |
119 | // Number of filter coefficients in each phase of the polyphase filter. |
120 | size_t coeffs_per_phase_; |
121 | |
122 | // Filter coefficients stored in polyphase form. |
123 | AudioBuffer transposed_filter_coeffs_; |
124 | |
125 | // Filter coefficients in planar form, used for calculating the transposed |
126 | // filter. |
127 | AudioBuffer temporary_filter_coeffs_; |
128 | |
129 | // Buffer holding the samples of input required between input buffers. |
130 | AudioBuffer state_; |
131 | }; |
132 | |
133 | } // namespace vraudio |
134 | |
135 | #endif // RESONANCE_AUDIO_DSP_RESAMPLER_H_ |
136 | |