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 34 using namespace std; 35 36 namespace WebCore { 37 38 const size_t AudioResamplerKernel::MaxFramesToProcess = 128; 39 40 AudioResamplerKernel::AudioResamplerKernel(AudioResampler* resampler) 41 : m_resampler(resampler) 42 // The buffer size must be large enough to hold up to two extra sample frames for the linear interpolation. 43 , m_sourceBuffer(2 + static_cast<int>(MaxFramesToProcess * AudioResampler::MaxRate)) 44 , m_virtualReadIndex(0.0) 45 , m_fillIndex(0) 46 { 47 m_lastValues[0] = 0.0f; 48 m_lastValues[1] = 0.0f; 49 } 50 51 float* AudioResamplerKernel::getSourcePointer(size_t framesToProcess, size_t* numberOfSourceFramesNeededP) 52 { 53 ASSERT(framesToProcess <= MaxFramesToProcess); 54 55 // Calculate the next "virtual" index. After process() is called, m_virtualReadIndex will equal this value. 56 double nextFractionalIndex = m_virtualReadIndex + framesToProcess * rate(); 57 58 // Because we're linearly interpolating between the previous and next sample we need to round up so we include the next sample. 59 int endIndex = static_cast<int>(nextFractionalIndex + 1.0); // round up to next integer index 60 61 // Determine how many input frames we'll need. 62 // 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. 63 size_t framesNeeded = 1 + endIndex - m_fillIndex; 64 if (numberOfSourceFramesNeededP) 65 *numberOfSourceFramesNeededP = framesNeeded; 66 67 // Do bounds checking for the source buffer. 68 bool isGood = m_fillIndex < m_sourceBuffer.size() && m_fillIndex + framesNeeded <= m_sourceBuffer.size(); 69 ASSERT(isGood); 70 if (!isGood) 71 return 0; 72 73 return m_sourceBuffer.data() + m_fillIndex; 74 } 75 76 void AudioResamplerKernel::process(float* destination, size_t framesToProcess) 77 { 78 ASSERT(framesToProcess <= MaxFramesToProcess); 79 80 float* source = m_sourceBuffer.data(); 81 82 double rate = this->rate(); 83 rate = max(0.0, rate); 84 rate = min(AudioResampler::MaxRate, rate); 85 86 // Start out with the previous saved values (if any). 87 if (m_fillIndex > 0) { 88 source[0] = m_lastValues[0]; 89 source[1] = m_lastValues[1]; 90 } 91 92 // Make a local copy. 93 double virtualReadIndex = m_virtualReadIndex; 94 95 // Sanity check source buffer access. 96 ASSERT(framesToProcess > 0); 97 ASSERT(virtualReadIndex >= 0 && 1 + static_cast<unsigned>(virtualReadIndex + (framesToProcess - 1) * rate) < m_sourceBuffer.size()); 98 99 // Do the linear interpolation. 100 int n = framesToProcess; 101 while (n--) { 102 unsigned readIndex = static_cast<unsigned>(virtualReadIndex); 103 double interpolationFactor = virtualReadIndex - readIndex; 104 105 double sample1 = source[readIndex]; 106 double sample2 = source[readIndex + 1]; 107 108 double sample = (1.0 - interpolationFactor) * sample1 + interpolationFactor * sample2; 109 110 *destination++ = static_cast<float>(sample); 111 112 virtualReadIndex += rate; 113 } 114 115 // Save the last two sample-frames which will later be used at the beginning of the source buffer the next time around. 116 int readIndex = static_cast<int>(virtualReadIndex); 117 m_lastValues[0] = source[readIndex]; 118 m_lastValues[1] = source[readIndex + 1]; 119 m_fillIndex = 2; 120 121 // Wrap the virtual read index back to the start of the buffer. 122 virtualReadIndex -= readIndex; 123 124 // Put local copy back into member variable. 125 m_virtualReadIndex = virtualReadIndex; 126 } 127 128 void AudioResamplerKernel::reset() 129 { 130 m_virtualReadIndex = 0.0; 131 m_fillIndex = 0; 132 m_lastValues[0] = 0.0f; 133 m_lastValues[1] = 0.0f; 134 } 135 136 double AudioResamplerKernel::rate() const 137 { 138 return m_resampler->rate(); 139 } 140 141 } // namespace WebCore 142 143 #endif // ENABLE(WEB_AUDIO) 144
