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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 "platform/audio/AudioResamplerKernel.h"
30 
31 #include <algorithm>
32 #include "platform/audio/AudioResampler.h"
33 
34 namespace blink {
35 
36 const size_t AudioResamplerKernel::MaxFramesToProcess = 128;
37 
AudioResamplerKernel(AudioResampler * resampler)38 AudioResamplerKernel::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 
getSourcePointer(size_t framesToProcess,size_t * numberOfSourceFramesNeededP)49 float* 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 
process(float * destination,size_t framesToProcess)74 void 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 
reset()126 void 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 
rate() const134 double AudioResamplerKernel::rate() const
135 {
136     return m_resampler->rate();
137 }
138 
139 } // namespace blink
140 
141 #endif // ENABLE(WEB_AUDIO)
142