1 /*
2 * Copyright (C) 2010 Google Inc. All rights reserved.
3 *
4 * Redistribution and use in source and binary forms, with or without
5 * modification, are permitted provided that the following conditions
6 * are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright
9 * notice, this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright
11 * notice, this list of conditions and the following disclaimer in the
12 * documentation and/or other materials provided with the distribution.
13 * 3. Neither the name of Apple Computer, Inc. ("Apple") nor the names of
14 * its contributors may be used to endorse or promote products derived
15 * from this software without specific prior written permission.
16 *
17 * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
18 * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
19 * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
20 * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
21 * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
22 * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
23 * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
24 * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
25 * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
26 * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
27 */
28
29 #include "config.h"
30
31 #if ENABLE(WEB_AUDIO)
32
33 #include "Reverb.h"
34
35 #include "AudioBus.h"
36 #include "AudioFileReader.h"
37 #include "ReverbConvolver.h"
38 #include <math.h>
39 #include <wtf/MathExtras.h>
40 #include <wtf/OwnPtr.h>
41 #include <wtf/PassOwnPtr.h>
42
43 #if OS(DARWIN)
44 using namespace std;
45 #endif
46
47 namespace WebCore {
48
49 // Empirical gain calibration tested across many impulse responses to ensure perceived volume is same as dry (unprocessed) signal
50 const double GainCalibration = -58.0;
51
52 // A minimum power value to when normalizing a silent (or very quiet) impulse response
53 const double MinPower = 0.000125;
54
calculateNormalizationScale(AudioBus * response)55 static double calculateNormalizationScale(AudioBus* response)
56 {
57 // Normalize by RMS power
58 size_t numberOfChannels = response->numberOfChannels();
59 size_t length = response->length();
60
61 double power = 0.0;
62
63 for (size_t i = 0; i < numberOfChannels; ++i) {
64 int n = length;
65 float* p = response->channel(i)->data();
66
67 while (n--) {
68 float sample = *p++;
69 power += sample * sample;
70 }
71 }
72
73 power = sqrt(power / (numberOfChannels * length));
74
75 // Protect against accidental overload
76 if (isinf(power) || isnan(power) || power < MinPower)
77 power = MinPower;
78
79 double scale = 1.0 / power;
80
81 scale *= pow(10.0, GainCalibration * 0.05); // calibrate to make perceived volume same as unprocessed
82
83 // True-stereo compensation
84 if (response->numberOfChannels() == 4)
85 scale *= 0.5;
86
87 return scale;
88 }
89
Reverb(AudioBus * impulseResponse,size_t renderSliceSize,size_t maxFFTSize,size_t numberOfChannels,bool useBackgroundThreads)90 Reverb::Reverb(AudioBus* impulseResponse, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads)
91 {
92 double scale = calculateNormalizationScale(impulseResponse);
93 if (scale)
94 impulseResponse->scale(scale);
95
96 initialize(impulseResponse, renderSliceSize, maxFFTSize, numberOfChannels, useBackgroundThreads);
97
98 // Undo scaling since this shouldn't be a destructive operation on impulseResponse
99 if (scale)
100 impulseResponse->scale(1.0 / scale);
101 }
102
initialize(AudioBus * impulseResponseBuffer,size_t renderSliceSize,size_t maxFFTSize,size_t numberOfChannels,bool useBackgroundThreads)103 void Reverb::initialize(AudioBus* impulseResponseBuffer, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads)
104 {
105 m_impulseResponseLength = impulseResponseBuffer->length();
106
107 // The reverb can handle a mono impulse response and still do stereo processing
108 size_t numResponseChannels = impulseResponseBuffer->numberOfChannels();
109 m_convolvers.reserveCapacity(numberOfChannels);
110
111 int convolverRenderPhase = 0;
112 for (size_t i = 0; i < numResponseChannels; ++i) {
113 AudioChannel* channel = impulseResponseBuffer->channel(i);
114
115 OwnPtr<ReverbConvolver> convolver = adoptPtr(new ReverbConvolver(channel, renderSliceSize, maxFFTSize, convolverRenderPhase, useBackgroundThreads));
116 m_convolvers.append(convolver.release());
117
118 convolverRenderPhase += renderSliceSize;
119 }
120
121 // For "True" stereo processing we allocate a temporary buffer to avoid repeatedly allocating it in the process() method.
122 // It can be bad to allocate memory in a real-time thread.
123 if (numResponseChannels == 4)
124 m_tempBuffer = new AudioBus(2, MaxFrameSize);
125 }
126
process(AudioBus * sourceBus,AudioBus * destinationBus,size_t framesToProcess)127 void Reverb::process(AudioBus* sourceBus, AudioBus* destinationBus, size_t framesToProcess)
128 {
129 // Do a fairly comprehensive sanity check.
130 // If these conditions are satisfied, all of the source and destination pointers will be valid for the various matrixing cases.
131 bool isSafeToProcess = sourceBus && destinationBus && sourceBus->numberOfChannels() > 0 && destinationBus->numberOfChannels() > 0
132 && framesToProcess <= MaxFrameSize && framesToProcess <= sourceBus->length() && framesToProcess <= destinationBus->length();
133
134 ASSERT(isSafeToProcess);
135 if (!isSafeToProcess)
136 return;
137
138 // For now only handle mono or stereo output
139 if (destinationBus->numberOfChannels() > 2) {
140 destinationBus->zero();
141 return;
142 }
143
144 AudioChannel* destinationChannelL = destinationBus->channel(0);
145 AudioChannel* sourceChannelL = sourceBus->channel(0);
146
147 // Handle input -> output matrixing...
148 size_t numInputChannels = sourceBus->numberOfChannels();
149 size_t numOutputChannels = destinationBus->numberOfChannels();
150 size_t numReverbChannels = m_convolvers.size();
151
152 if (numInputChannels == 2 && numReverbChannels == 2 && numOutputChannels == 2) {
153 // 2 -> 2 -> 2
154 AudioChannel* sourceChannelR = sourceBus->channel(1);
155 AudioChannel* destinationChannelR = destinationBus->channel(1);
156 m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
157 m_convolvers[1]->process(sourceChannelR, destinationChannelR, framesToProcess);
158 } else if (numInputChannels == 1 && numOutputChannels == 2 && numReverbChannels == 2) {
159 // 1 -> 2 -> 2
160 for (int i = 0; i < 2; ++i) {
161 AudioChannel* destinationChannel = destinationBus->channel(i);
162 m_convolvers[i]->process(sourceChannelL, destinationChannel, framesToProcess);
163 }
164 } else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 2) {
165 // 1 -> 1 -> 2
166 m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
167
168 // simply copy L -> R
169 AudioChannel* destinationChannelR = destinationBus->channel(1);
170 bool isCopySafe = destinationChannelL->data() && destinationChannelR->data() && destinationChannelL->length() >= framesToProcess && destinationChannelR->length() >= framesToProcess;
171 ASSERT(isCopySafe);
172 if (!isCopySafe)
173 return;
174 memcpy(destinationChannelR->data(), destinationChannelL->data(), sizeof(float) * framesToProcess);
175 } else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 1) {
176 // 1 -> 1 -> 1
177 m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
178 } else if (numInputChannels == 2 && numReverbChannels == 4 && numOutputChannels == 2) {
179 // 2 -> 4 -> 2 ("True" stereo)
180 AudioChannel* sourceChannelR = sourceBus->channel(1);
181 AudioChannel* destinationChannelR = destinationBus->channel(1);
182
183 AudioChannel* tempChannelL = m_tempBuffer->channel(0);
184 AudioChannel* tempChannelR = m_tempBuffer->channel(1);
185
186 // Process left virtual source
187 m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
188 m_convolvers[1]->process(sourceChannelL, destinationChannelR, framesToProcess);
189
190 // Process right virtual source
191 m_convolvers[2]->process(sourceChannelR, tempChannelL, framesToProcess);
192 m_convolvers[3]->process(sourceChannelR, tempChannelR, framesToProcess);
193
194 destinationBus->sumFrom(*m_tempBuffer);
195 } else if (numInputChannels == 1 && numReverbChannels == 4 && numOutputChannels == 2) {
196 // 1 -> 4 -> 2 (Processing mono with "True" stereo impulse response)
197 // This is an inefficient use of a four-channel impulse response, but we should handle the case.
198 AudioChannel* destinationChannelR = destinationBus->channel(1);
199
200 AudioChannel* tempChannelL = m_tempBuffer->channel(0);
201 AudioChannel* tempChannelR = m_tempBuffer->channel(1);
202
203 // Process left virtual source
204 m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
205 m_convolvers[1]->process(sourceChannelL, destinationChannelR, framesToProcess);
206
207 // Process right virtual source
208 m_convolvers[2]->process(sourceChannelL, tempChannelL, framesToProcess);
209 m_convolvers[3]->process(sourceChannelL, tempChannelR, framesToProcess);
210
211 destinationBus->sumFrom(*m_tempBuffer);
212 } else {
213 // Handle gracefully any unexpected / unsupported matrixing
214 // FIXME: add code for 5.1 support...
215 destinationBus->zero();
216 }
217 }
218
reset()219 void Reverb::reset()
220 {
221 for (size_t i = 0; i < m_convolvers.size(); ++i)
222 m_convolvers[i]->reset();
223 }
224
225 } // namespace WebCore
226
227 #endif // ENABLE(WEB_AUDIO)
228