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_SPECTRAL_REVERB_H_ |
18 | #define RESONANCE_AUDIO_DSP_SPECTRAL_REVERB_H_ |
19 | |
20 | #include <memory> |
21 | #include <vector> |
22 | |
23 | #include "base/audio_buffer.h" |
24 | #include "dsp/circular_buffer.h" |
25 | #include "dsp/fft_manager.h" |
26 | |
27 | namespace vraudio { |
28 | |
29 | // Implements a spectral reverb producing a decorrelated stereo output. See: |
30 | // [1] E. Vickers, J-L Wu, P.G. Krishnan, R. N. K. Sadanandam, "Frequency Domain |
31 | // Artificial Reverberation using Spectral Magnitude Decay", |
32 | // https://goo.gl/hv1pdJ. |
33 | class SpectralReverb { |
34 | public: |
35 | // Constructs a spectral reverb. |
36 | // |
37 | // @param sample_rate The system sample rate. |
38 | // @param frames_per_buffer System frames per buffer of input and output. |
39 | // Note that this class expects power of two buffers of input and output. |
40 | SpectralReverb(int sample_rate, size_t frames_per_buffer); |
41 | |
42 | // Sets the overall gain to be applied to the output of the reverb. |
43 | // |
44 | // @param gain Gain to be applied to the reverb output, min value 0.0f. |
45 | void SetGain(float gain); |
46 | |
47 | // Sets the |SpectralReverb|'s reverberation times in different frequency |
48 | // bands. Supports times between: |
49 | // (0.15 * 48000 / |sample_rate|)s and (25 * 48000 / |sample_rate|)s. |
50 | // |
51 | // @param rt60_values |kNumReverbOctaveBands| values denoting the |
52 | // reverberation decay time to -60dB in octave bands starting at |
53 | // |kLowestOctaveBand|. |
54 | void SetRt60PerOctaveBand(const float* rt60_values); |
55 | |
56 | // Applies reverb to an input channel of audio data and produces a stereo |
57 | // output. |
58 | // |
59 | // @param input Mono inpu data. |
60 | // @param left_out Left channel of reverberated output. |
61 | // @param right_out Right channel of reverberated output. |
62 | void Process(const AudioBuffer::Channel& input, |
63 | AudioBuffer::Channel* left_out, AudioBuffer::Channel* right_out); |
64 | |
65 | private: |
66 | // Uses an AudioBuffer with four channels to overlap add and insert the final |
67 | // reverb into the output circular buffers. |
68 | // |
69 | // @param channel_index Denotes the (left or right) channel to output to. |
70 | // @param buffer The buffer to be added onto the pre-existing reverb output. |
71 | void AccumulateOverlap(size_t channel_index, |
72 | const AudioBuffer::Channel& buffer); |
73 | |
74 | // Generates a window function which is a normalized sum of three overlapping |
75 | // (50%) hann windows of length (|kFftSize| / 2) that also incorporates the |
76 | // inverse fft scaling. |
77 | void GenerateAnalysisWindow(); |
78 | |
79 | // Generates a large buffer of sines and cosines of random noise between 0 and |
80 | // pi to be randomly indexed into in order to cheaply generate highly |
81 | // decorrelated phase buffers, |
82 | void GenerateRandomPhaseBuffer(); |
83 | |
84 | // Obtains the next stero pair of time domain reverb blocks which can then be |
85 | // summed together in an overlap add fashion to provide the reverb output. |
86 | // |
87 | // @param delay_index An index into the frequency domain magnitude ring |
88 | // buffer. |
89 | // @param left_channel Channel to contain the left partial reverb output. |
90 | // @param right_channel Channel to contain the right partial reverb output. |
91 | void GetNextReverbBlock(size_t delay_index, |
92 | AudioBuffer::Channel* left_channel, |
93 | AudioBuffer::Channel* right_channel); |
94 | |
95 | // Initializes the output circular buffers such that they contain zeros if the |
96 | // value of |frames_per_buffer_| is sufficiently smaller than that of |
97 | // |kOverlapLength| that buffering of input will be required prior to |
98 | // processing. Also allocates memory for the output accumulators. |
99 | void InitializeCircularBuffersAndAccumulators(); |
100 | |
101 | // System sample rate. |
102 | const int sample_rate_; |
103 | |
104 | // System frames per buffer. |
105 | const size_t frames_per_buffer_; |
106 | |
107 | // Indices into the magnitude and overlap add delay lines, modulo of their |
108 | // respective lengths. |
109 | size_t magnitude_delay_index_; |
110 | size_t overlap_add_index_; |
111 | |
112 | // Manages the time-frequency transforms and phase/magnitude-frequency |
113 | // transforms. |
114 | FftManager fft_manager_; |
115 | |
116 | // Buffer containing sines and cosines of random values between 0 and pi to be |
117 | // used for phase. |
118 | AudioBuffer sin_cos_random_phase_buffer_; |
119 | |
120 | // Buffer containing a triple overlapping hann window for windowing time |
121 | // domain data. |
122 | AudioBuffer unscaled_window_; |
123 | |
124 | // Buffer containing a triple overlapping hann window for windowing time |
125 | // domain data, this window has been scaled by the output gain factor. |
126 | AudioBuffer window_; |
127 | |
128 | // Buffer containing RT60 tuned feedback values. |
129 | AudioBuffer feedback_; |
130 | |
131 | // Buffer used to store scaling values which account for the different initial |
132 | // peak magnitudes for different RT60s. |
133 | AudioBuffer magnitude_compensation_; |
134 | |
135 | // Buffer that acts as the frequency domain magnitde delay. |
136 | AudioBuffer magnitude_delay_; |
137 | |
138 | // Buffer to contain a linear |kFftSize| chunk of input data. |
139 | AudioBuffer fft_size_input_; |
140 | |
141 | // Circular buffers to sit at the input and output of the |Process()| method |
142 | // to allow |frames_per_buffer_| to differ from |kFftSize|. |
143 | CircularBuffer input_circular_buffer_; |
144 | std::vector<std::unique_ptr<CircularBuffer>> output_circular_buffers_; |
145 | |
146 | // Time domain buffer used to store reverb before the overlap add operation. |
147 | AudioBuffer out_time_buffer_; |
148 | |
149 | // Temporary frequency domain buffer, used to store frequency domain data when |
150 | // transforming between Pffft and Canonical format frequency domain data. |
151 | AudioBuffer temp_freq_buffer_; |
152 | |
153 | // Buffer used to store feedback scaled magnitude values. |
154 | AudioBuffer scaled_magnitude_buffer_; |
155 | |
156 | // Buffer used for the accumulation of scaled magnitude buffers. |
157 | AudioBuffer temp_magnitude_buffer_; |
158 | |
159 | // Buffer used to store randomized phase. |
160 | AudioBuffer temp_phase_buffer_; |
161 | |
162 | // Buffers used to calculate the overlap add at the output. |
163 | std::vector<AudioBuffer> output_accumulator_; |
164 | |
165 | // Processing of the spectral reverb is bypassed when the feedback values are |
166 | // all approximately zero OR when the gain is set to near zero. |
167 | bool is_gain_near_zero_; |
168 | bool is_feedback_near_zero_; |
169 | }; |
170 | |
171 | } // namespace vraudio |
172 | |
173 | #endif // RESONANCE_AUDIO_DSP_SPECTRAL_REVERB_H_ |
174 | |