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 * 1. Redistributions of source code must retain the above copyright
8 * notice, this list of conditions and the following disclaimer.
9 * 2. Redistributions in binary form must reproduce the above copyright
10 * notice, this list of conditions and the following disclaimer in the
11 * documentation and/or other materials provided with the distribution.
12 *
13 * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
14 * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
15 * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
16 * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
17 * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
18 * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
19 * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
20 * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
21 * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
22 * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
23 */
24
25 #include "config.h"
26
27 #if ENABLE(WEB_AUDIO)
28
29 #include "RealtimeAnalyser.h"
30
31 #include "AudioBus.h"
32 #include "AudioUtilities.h"
33 #include "FFTFrame.h"
34
35 #if ENABLE(WEBGL)
36 #include "Float32Array.h"
37 #include "Uint8Array.h"
38 #endif
39
40 #include <algorithm>
41 #include <limits.h>
42 #include <wtf/Complex.h>
43 #include <wtf/MathExtras.h>
44 #include <wtf/Threading.h>
45
46 using namespace std;
47
48 namespace WebCore {
49
50 const double RealtimeAnalyser::DefaultSmoothingTimeConstant = 0.8;
51 const double RealtimeAnalyser::DefaultMinDecibels = -100.0;
52 const double RealtimeAnalyser::DefaultMaxDecibels = -30.0;
53
54 const unsigned RealtimeAnalyser::DefaultFFTSize = 2048;
55 const unsigned RealtimeAnalyser::MaxFFTSize = 2048;
56 const unsigned RealtimeAnalyser::InputBufferSize = RealtimeAnalyser::MaxFFTSize * 2;
57
RealtimeAnalyser()58 RealtimeAnalyser::RealtimeAnalyser()
59 : m_inputBuffer(InputBufferSize)
60 , m_writeIndex(0)
61 , m_fftSize(DefaultFFTSize)
62 , m_magnitudeBuffer(DefaultFFTSize / 2)
63 , m_smoothingTimeConstant(DefaultSmoothingTimeConstant)
64 , m_minDecibels(DefaultMinDecibels)
65 , m_maxDecibels(DefaultMaxDecibels)
66 {
67 m_analysisFrame = adoptPtr(new FFTFrame(DefaultFFTSize));
68 }
69
~RealtimeAnalyser()70 RealtimeAnalyser::~RealtimeAnalyser()
71 {
72 }
73
reset()74 void RealtimeAnalyser::reset()
75 {
76 m_writeIndex = 0;
77 m_inputBuffer.zero();
78 m_magnitudeBuffer.zero();
79 }
80
setFftSize(size_t size)81 void RealtimeAnalyser::setFftSize(size_t size)
82 {
83 ASSERT(isMainThread());
84
85 // Only allow powers of two.
86 unsigned log2size = static_cast<unsigned>(log2(size));
87 bool isPOT(1UL << log2size == size);
88
89 if (!isPOT || size > MaxFFTSize) {
90 // FIXME: It would be good to also set an exception.
91 return;
92 }
93
94 if (m_fftSize != size) {
95 m_analysisFrame = adoptPtr(new FFTFrame(m_fftSize));
96 m_magnitudeBuffer.resize(size);
97 m_fftSize = size;
98 }
99 }
100
writeInput(AudioBus * bus,size_t framesToProcess)101 void RealtimeAnalyser::writeInput(AudioBus* bus, size_t framesToProcess)
102 {
103 bool isBusGood = bus && bus->numberOfChannels() > 0 && bus->channel(0)->length() >= framesToProcess;
104 ASSERT(isBusGood);
105 if (!isBusGood)
106 return;
107
108 // FIXME : allow to work with non-FFTSize divisible chunking
109 bool isDestinationGood = m_writeIndex < m_inputBuffer.size() && m_writeIndex + framesToProcess <= m_inputBuffer.size();
110 ASSERT(isDestinationGood);
111 if (!isDestinationGood)
112 return;
113
114 // Perform real-time analysis
115 // FIXME : for now just use left channel (must mix if stereo source)
116 float* source = bus->channel(0)->data();
117
118 // The source has already been sanity checked with isBusGood above.
119
120 memcpy(m_inputBuffer.data() + m_writeIndex, source, sizeof(float) * framesToProcess);
121
122 m_writeIndex += framesToProcess;
123 if (m_writeIndex >= InputBufferSize)
124 m_writeIndex = 0;
125 }
126
127 namespace {
128
applyWindow(float * p,size_t n)129 void applyWindow(float* p, size_t n)
130 {
131 ASSERT(isMainThread());
132
133 // Blackman window
134 double alpha = 0.16;
135 double a0 = 0.5 * (1.0 - alpha);
136 double a1 = 0.5;
137 double a2 = 0.5 * alpha;
138
139 for (unsigned i = 0; i < n; ++i) {
140 double x = static_cast<double>(i) / static_cast<double>(n);
141 double window = a0 - a1 * cos(2.0 * piDouble * x) + a2 * cos(4.0 * piDouble * x);
142 p[i] *= float(window);
143 }
144 }
145
146 } // namespace
147
doFFTAnalysis()148 void RealtimeAnalyser::doFFTAnalysis()
149 {
150 ASSERT(isMainThread());
151
152 // Unroll the input buffer into a temporary buffer, where we'll apply an analysis window followed by an FFT.
153 size_t fftSize = this->fftSize();
154
155 AudioFloatArray temporaryBuffer(fftSize);
156 float* inputBuffer = m_inputBuffer.data();
157 float* tempP = temporaryBuffer.data();
158
159 // Take the previous fftSize values from the input buffer and copy into the temporary buffer.
160 // FIXME : optimize with memcpy().
161 unsigned writeIndex = m_writeIndex;
162 for (unsigned i = 0; i < fftSize; ++i)
163 tempP[i] = inputBuffer[(i + writeIndex - fftSize + InputBufferSize) % InputBufferSize];
164
165 // Window the input samples.
166 applyWindow(tempP, fftSize);
167
168 // Do the analysis.
169 m_analysisFrame->doFFT(tempP);
170
171 size_t n = DefaultFFTSize / 2;
172
173 float* realP = m_analysisFrame->realData();
174 float* imagP = m_analysisFrame->imagData();
175
176 // Blow away the packed nyquist component.
177 imagP[0] = 0.0f;
178
179 // Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT scaling factor).
180 const double MagnitudeScale = 1.0 / DefaultFFTSize;
181
182 // A value of 0 does no averaging with the previous result. Larger values produce slower, but smoother changes.
183 double k = m_smoothingTimeConstant;
184 k = max(0.0, k);
185 k = min(1.0, k);
186
187 // Convert the analysis data from complex to magnitude and average with the previous result.
188 float* destination = magnitudeBuffer().data();
189 for (unsigned i = 0; i < n; ++i) {
190 Complex c(realP[i], imagP[i]);
191 double scalarMagnitude = abs(c) * MagnitudeScale;
192 destination[i] = float(k * destination[i] + (1.0 - k) * scalarMagnitude);
193 }
194 }
195
196 #if ENABLE(WEBGL)
197
getFloatFrequencyData(Float32Array * destinationArray)198 void RealtimeAnalyser::getFloatFrequencyData(Float32Array* destinationArray)
199 {
200 ASSERT(isMainThread());
201
202 if (!destinationArray)
203 return;
204
205 doFFTAnalysis();
206
207 // Convert from linear magnitude to floating-point decibels.
208 const double MinDecibels = m_minDecibels;
209 unsigned sourceLength = magnitudeBuffer().size();
210 size_t len = min(sourceLength, destinationArray->length());
211 if (len > 0) {
212 const float* source = magnitudeBuffer().data();
213 float* destination = destinationArray->data();
214
215 for (unsigned i = 0; i < len; ++i) {
216 float linearValue = source[i];
217 double dbMag = !linearValue ? MinDecibels : AudioUtilities::linearToDecibels(linearValue);
218 destination[i] = float(dbMag);
219 }
220 }
221 }
222
getByteFrequencyData(Uint8Array * destinationArray)223 void RealtimeAnalyser::getByteFrequencyData(Uint8Array* destinationArray)
224 {
225 ASSERT(isMainThread());
226
227 if (!destinationArray)
228 return;
229
230 doFFTAnalysis();
231
232 // Convert from linear magnitude to unsigned-byte decibels.
233 unsigned sourceLength = magnitudeBuffer().size();
234 size_t len = min(sourceLength, destinationArray->length());
235 if (len > 0) {
236 const double RangeScaleFactor = m_maxDecibels == m_minDecibels ? 1.0 : 1.0 / (m_maxDecibels - m_minDecibels);
237
238 const float* source = magnitudeBuffer().data();
239 unsigned char* destination = destinationArray->data();
240
241 for (unsigned i = 0; i < len; ++i) {
242 float linearValue = source[i];
243 double dbMag = !linearValue ? m_minDecibels : AudioUtilities::linearToDecibels(linearValue);
244
245 // The range m_minDecibels to m_maxDecibels will be scaled to byte values from 0 to UCHAR_MAX.
246 double scaledValue = UCHAR_MAX * (dbMag - m_minDecibels) * RangeScaleFactor;
247
248 // Clip to valid range.
249 if (scaledValue < 0.0)
250 scaledValue = 0.0;
251 if (scaledValue > UCHAR_MAX)
252 scaledValue = UCHAR_MAX;
253
254 destination[i] = static_cast<unsigned char>(scaledValue);
255 }
256 }
257 }
258
getByteTimeDomainData(Uint8Array * destinationArray)259 void RealtimeAnalyser::getByteTimeDomainData(Uint8Array* destinationArray)
260 {
261 ASSERT(isMainThread());
262
263 if (!destinationArray)
264 return;
265
266 unsigned fftSize = this->fftSize();
267 size_t len = min(fftSize, destinationArray->length());
268 if (len > 0) {
269 bool isInputBufferGood = m_inputBuffer.size() == InputBufferSize && m_inputBuffer.size() > fftSize;
270 ASSERT(isInputBufferGood);
271 if (!isInputBufferGood)
272 return;
273
274 float* inputBuffer = m_inputBuffer.data();
275 unsigned char* destination = destinationArray->data();
276
277 unsigned writeIndex = m_writeIndex;
278
279 for (unsigned i = 0; i < len; ++i) {
280 // Buffer access is protected due to modulo operation.
281 float value = inputBuffer[(i + writeIndex - fftSize + InputBufferSize) % InputBufferSize];
282
283 // Scale from nominal -1.0 -> +1.0 to unsigned byte.
284 double scaledValue = 128.0 * (value + 1.0);
285
286 // Clip to valid range.
287 if (scaledValue < 0.0)
288 scaledValue = 0.0;
289 if (scaledValue > UCHAR_MAX)
290 scaledValue = UCHAR_MAX;
291
292 destination[i] = static_cast<unsigned char>(scaledValue);
293 }
294 }
295 }
296
297 #endif // WEBGL
298
299 } // namespace WebCore
300
301 #endif // ENABLE(WEB_AUDIO)
302