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 "AudioResamplerKernel.h"
30
31#include "AudioResampler.h"
32#include <algorithm>
33
34namespace WebCore {
35
36const size_t AudioResamplerKernel::MaxFramesToProcess = 128;
37
38AudioResamplerKernel::AudioResamplerKernel(AudioResampler* resampler)
39 : m_resampler(resampler)
40 // The buffer size must be large enough to hold up to two extra sample frames for the linear interpolation.
41 , m_sourceBuffer(2 + static_cast<int>(MaxFramesToProcess * AudioResampler::MaxRate))
42 , m_virtualReadIndex(0.0)
43 , m_fillIndex(0)
44{
45 m_lastValues[0] = 0.0f;
46 m_lastValues[1] = 0.0f;
47}
48
49float* AudioResamplerKernel::getSourcePointer(size_t framesToProcess, size_t* numberOfSourceFramesNeededP)
50{
51 ASSERT(framesToProcess <= MaxFramesToProcess);
52
53 // Calculate the next "virtual" index. After process() is called, m_virtualReadIndex will equal this value.
54 double nextFractionalIndex = m_virtualReadIndex + framesToProcess * rate();
55
56 // Because we're linearly interpolating between the previous and next sample we need to round up so we include the next sample.
57 int endIndex = static_cast<int>(nextFractionalIndex + 1.0); // round up to next integer index
58
59 // Determine how many input frames we'll need.
60 // We need to fill the buffer up to and including endIndex (so add 1) but we've already buffered m_fillIndex frames from last time.
61 size_t framesNeeded = 1 + endIndex - m_fillIndex;
62 if (numberOfSourceFramesNeededP)
63 *numberOfSourceFramesNeededP = framesNeeded;
64
65 // Do bounds checking for the source buffer.
66 bool isGood = m_fillIndex < m_sourceBuffer.size() && m_fillIndex + framesNeeded <= m_sourceBuffer.size();
67 ASSERT(isGood);
68 if (!isGood)
69 return 0;
70
71 return m_sourceBuffer.data() + m_fillIndex;
72}
73
74void AudioResamplerKernel::process(float* destination, size_t framesToProcess)
75{
76 ASSERT(framesToProcess <= MaxFramesToProcess);
77
78 float* source = m_sourceBuffer.data();
79
80 double rate = this->rate();
81 rate = std::max(0.0, rate);
82 rate = std::min(AudioResampler::MaxRate, rate);
83
84 // Start out with the previous saved values (if any).
85 if (m_fillIndex > 0) {
86 source[0] = m_lastValues[0];
87 source[1] = m_lastValues[1];
88 }
89
90 // Make a local copy.
91 double virtualReadIndex = m_virtualReadIndex;
92
93 // Sanity check source buffer access.
94 ASSERT(framesToProcess > 0);
95 ASSERT(virtualReadIndex >= 0 && 1 + static_cast<unsigned>(virtualReadIndex + (framesToProcess - 1) * rate) < m_sourceBuffer.size());
96
97 // Do the linear interpolation.
98 int n = framesToProcess;
99 while (n--) {
100 unsigned readIndex = static_cast<unsigned>(virtualReadIndex);
101 double interpolationFactor = virtualReadIndex - readIndex;
102
103 double sample1 = source[readIndex];
104 double sample2 = source[readIndex + 1];
105
106 double sample = (1.0 - interpolationFactor) * sample1 + interpolationFactor * sample2;
107
108 *destination++ = static_cast<float>(sample);
109
110 virtualReadIndex += rate;
111 }
112
113 // Save the last two sample-frames which will later be used at the beginning of the source buffer the next time around.
114 int readIndex = static_cast<int>(virtualReadIndex);
115 m_lastValues[0] = source[readIndex];
116 m_lastValues[1] = source[readIndex + 1];
117 m_fillIndex = 2;
118
119 // Wrap the virtual read index back to the start of the buffer.
120 virtualReadIndex -= readIndex;
121
122 // Put local copy back into member variable.
123 m_virtualReadIndex = virtualReadIndex;
124}
125
126void AudioResamplerKernel::reset()
127{
128 m_virtualReadIndex = 0.0;
129 m_fillIndex = 0;
130 m_lastValues[0] = 0.0f;
131 m_lastValues[1] = 0.0f;
132}
133
134double AudioResamplerKernel::rate() const
135{
136 return m_resampler->rate();
137}
138
139} // namespace WebCore
140
141#endif // ENABLE(WEB_AUDIO)
142