1 /*
2  *  Copyright (c) 2014 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 //
12 //  Specifies core class for intelligbility enhancement.
13 //
14 
15 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_
16 #define WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_
17 
18 #include <complex>
19 #include <vector>
20 
21 #include "webrtc/base/scoped_ptr.h"
22 #include "webrtc/common_audio/lapped_transform.h"
23 #include "webrtc/common_audio/channel_buffer.h"
24 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
25 
26 namespace webrtc {
27 
28 // Speech intelligibility enhancement module. Reads render and capture
29 // audio streams and modifies the render stream with a set of gains per
30 // frequency bin to enhance speech against the noise background.
31 // Note: assumes speech and noise streams are already separated.
32 class IntelligibilityEnhancer {
33  public:
34   struct Config {
35     // |var_*| are parameters for the VarianceArray constructor for the
36     // clear speech stream.
37     // TODO(bercic): the |var_*|, |*_rate| and |gain_limit| parameters should
38     // probably go away once fine tuning is done.
ConfigConfig39     Config()
40         : sample_rate_hz(16000),
41           num_capture_channels(1),
42           num_render_channels(1),
43           var_type(intelligibility::VarianceArray::kStepDecaying),
44           var_decay_rate(0.9f),
45           var_window_size(10),
46           analysis_rate(800),
47           gain_change_limit(0.1f),
48           rho(0.02f) {}
49     int sample_rate_hz;
50     int num_capture_channels;
51     int num_render_channels;
52     intelligibility::VarianceArray::StepType var_type;
53     float var_decay_rate;
54     size_t var_window_size;
55     int analysis_rate;
56     float gain_change_limit;
57     float rho;
58   };
59 
60   explicit IntelligibilityEnhancer(const Config& config);
61   IntelligibilityEnhancer();  // Initialize with default config.
62 
63   // Reads and processes chunk of noise stream in time domain.
64   void AnalyzeCaptureAudio(float* const* audio,
65                            int sample_rate_hz,
66                            int num_channels);
67 
68   // Reads chunk of speech in time domain and updates with modified signal.
69   void ProcessRenderAudio(float* const* audio,
70                           int sample_rate_hz,
71                           int num_channels);
72   bool active() const;
73 
74  private:
75   enum AudioSource {
76     kRenderStream = 0,  // Clear speech stream.
77     kCaptureStream,  // Noise stream.
78   };
79 
80   // Provides access point to the frequency domain.
81   class TransformCallback : public LappedTransform::Callback {
82    public:
83     TransformCallback(IntelligibilityEnhancer* parent, AudioSource source);
84 
85     // All in frequency domain, receives input |in_block|, applies
86     // intelligibility enhancement, and writes result to |out_block|.
87     void ProcessAudioBlock(const std::complex<float>* const* in_block,
88                            int in_channels,
89                            size_t frames,
90                            int out_channels,
91                            std::complex<float>* const* out_block) override;
92 
93    private:
94     IntelligibilityEnhancer* parent_;
95     AudioSource source_;
96   };
97   friend class TransformCallback;
98 #ifndef WEBRTC_AUDIO_PROCESSING_ONLY_BUILD
99   FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, TestErbCreation);
100   FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, TestSolveForGains);
101 #endif
102 
103   // Sends streams to ProcessClearBlock or ProcessNoiseBlock based on source.
104   void DispatchAudio(AudioSource source,
105                      const std::complex<float>* in_block,
106                      std::complex<float>* out_block);
107 
108   // Updates variance computation and analysis with |in_block_|,
109   // and writes modified speech to |out_block|.
110   void ProcessClearBlock(const std::complex<float>* in_block,
111                          std::complex<float>* out_block);
112 
113   // Computes and sets modified gains.
114   void AnalyzeClearBlock(float power_target);
115 
116   // Bisection search for optimal |lambda|.
117   void SolveForLambda(float power_target, float power_bot, float power_top);
118 
119   // Transforms freq gains to ERB gains.
120   void UpdateErbGains();
121 
122   // Updates variance calculation for noise input with |in_block|.
123   void ProcessNoiseBlock(const std::complex<float>* in_block,
124                          std::complex<float>* out_block);
125 
126   // Returns number of ERB filters.
127   static size_t GetBankSize(int sample_rate, size_t erb_resolution);
128 
129   // Initializes ERB filterbank.
130   void CreateErbBank();
131 
132   // Analytically solves quadratic for optimal gains given |lambda|.
133   // Negative gains are set to 0. Stores the results in |sols|.
134   void SolveForGainsGivenLambda(float lambda, size_t start_freq, float* sols);
135 
136   // Computes variance across ERB filters from freq variance |var|.
137   // Stores in |result|.
138   void FilterVariance(const float* var, float* result);
139 
140   // Returns dot product of vectors specified by size |length| arrays |a|,|b|.
141   static float DotProduct(const float* a, const float* b, size_t length);
142 
143   const size_t freqs_;         // Num frequencies in frequency domain.
144   const size_t window_size_;   // Window size in samples; also the block size.
145   const size_t chunk_length_;  // Chunk size in samples.
146   const size_t bank_size_;     // Num ERB filters.
147   const int sample_rate_hz_;
148   const int erb_resolution_;
149   const int num_capture_channels_;
150   const int num_render_channels_;
151   const int analysis_rate_;    // Num blocks before gains recalculated.
152 
153   const bool active_;          // Whether render gains are being updated.
154                                // TODO(ekm): Add logic for updating |active_|.
155 
156   intelligibility::VarianceArray clear_variance_;
157   intelligibility::VarianceArray noise_variance_;
158   rtc::scoped_ptr<float[]> filtered_clear_var_;
159   rtc::scoped_ptr<float[]> filtered_noise_var_;
160   std::vector<std::vector<float>> filter_bank_;
161   rtc::scoped_ptr<float[]> center_freqs_;
162   size_t start_freq_;
163   rtc::scoped_ptr<float[]> rho_;  // Production and interpretation SNR.
164                                   // for each ERB band.
165   rtc::scoped_ptr<float[]> gains_eq_;  // Pre-filter modified gains.
166   intelligibility::GainApplier gain_applier_;
167 
168   // Destination buffers used to reassemble blocked chunks before overwriting
169   // the original input array with modifications.
170   ChannelBuffer<float> temp_render_out_buffer_;
171   ChannelBuffer<float> temp_capture_out_buffer_;
172 
173   rtc::scoped_ptr<float[]> kbd_window_;
174   TransformCallback render_callback_;
175   TransformCallback capture_callback_;
176   rtc::scoped_ptr<LappedTransform> render_mangler_;
177   rtc::scoped_ptr<LappedTransform> capture_mangler_;
178   int block_count_;
179   int analysis_step_;
180 };
181 
182 }  // namespace webrtc
183 
184 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_
185