| /* |
| * Copyright (C) 2010, Google Inc. All rights reserved. |
| * |
| * Redistribution and use in source and binary forms, with or without |
| * modification, are permitted provided that the following conditions |
| * are met: |
| * 1. Redistributions of source code must retain the above copyright |
| * notice, this list of conditions and the following disclaimer. |
| * 2. Redistributions in binary form must reproduce the above copyright |
| * notice, this list of conditions and the following disclaimer in the |
| * documentation and/or other materials provided with the distribution. |
| * |
| * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY |
| * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED |
| * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE |
| * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY |
| * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES |
| * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; |
| * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON |
| * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT |
| * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS |
| * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
| */ |
| |
| #include "config.h" |
| |
| #if ENABLE(WEB_AUDIO) |
| |
| #include "platform/audio/AudioResamplerKernel.h" |
| |
| #include <algorithm> |
| #include "platform/audio/AudioResampler.h" |
| |
| using namespace std; |
| |
| namespace WebCore { |
| |
| const size_t AudioResamplerKernel::MaxFramesToProcess = 128; |
| |
| AudioResamplerKernel::AudioResamplerKernel(AudioResampler* resampler) |
| : m_resampler(resampler) |
| // The buffer size must be large enough to hold up to two extra sample frames for the linear interpolation. |
| , m_sourceBuffer(2 + static_cast<int>(MaxFramesToProcess * AudioResampler::MaxRate)) |
| , m_virtualReadIndex(0.0) |
| , m_fillIndex(0) |
| { |
| m_lastValues[0] = 0.0f; |
| m_lastValues[1] = 0.0f; |
| } |
| |
| float* AudioResamplerKernel::getSourcePointer(size_t framesToProcess, size_t* numberOfSourceFramesNeededP) |
| { |
| ASSERT(framesToProcess <= MaxFramesToProcess); |
| |
| // Calculate the next "virtual" index. After process() is called, m_virtualReadIndex will equal this value. |
| double nextFractionalIndex = m_virtualReadIndex + framesToProcess * rate(); |
| |
| // Because we're linearly interpolating between the previous and next sample we need to round up so we include the next sample. |
| int endIndex = static_cast<int>(nextFractionalIndex + 1.0); // round up to next integer index |
| |
| // Determine how many input frames we'll need. |
| // 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. |
| size_t framesNeeded = 1 + endIndex - m_fillIndex; |
| if (numberOfSourceFramesNeededP) |
| *numberOfSourceFramesNeededP = framesNeeded; |
| |
| // Do bounds checking for the source buffer. |
| bool isGood = m_fillIndex < m_sourceBuffer.size() && m_fillIndex + framesNeeded <= m_sourceBuffer.size(); |
| ASSERT(isGood); |
| if (!isGood) |
| return 0; |
| |
| return m_sourceBuffer.data() + m_fillIndex; |
| } |
| |
| void AudioResamplerKernel::process(float* destination, size_t framesToProcess) |
| { |
| ASSERT(framesToProcess <= MaxFramesToProcess); |
| |
| float* source = m_sourceBuffer.data(); |
| |
| double rate = this->rate(); |
| rate = max(0.0, rate); |
| rate = min(AudioResampler::MaxRate, rate); |
| |
| // Start out with the previous saved values (if any). |
| if (m_fillIndex > 0) { |
| source[0] = m_lastValues[0]; |
| source[1] = m_lastValues[1]; |
| } |
| |
| // Make a local copy. |
| double virtualReadIndex = m_virtualReadIndex; |
| |
| // Sanity check source buffer access. |
| ASSERT(framesToProcess > 0); |
| ASSERT(virtualReadIndex >= 0 && 1 + static_cast<unsigned>(virtualReadIndex + (framesToProcess - 1) * rate) < m_sourceBuffer.size()); |
| |
| // Do the linear interpolation. |
| int n = framesToProcess; |
| while (n--) { |
| unsigned readIndex = static_cast<unsigned>(virtualReadIndex); |
| double interpolationFactor = virtualReadIndex - readIndex; |
| |
| double sample1 = source[readIndex]; |
| double sample2 = source[readIndex + 1]; |
| |
| double sample = (1.0 - interpolationFactor) * sample1 + interpolationFactor * sample2; |
| |
| *destination++ = static_cast<float>(sample); |
| |
| virtualReadIndex += rate; |
| } |
| |
| // Save the last two sample-frames which will later be used at the beginning of the source buffer the next time around. |
| int readIndex = static_cast<int>(virtualReadIndex); |
| m_lastValues[0] = source[readIndex]; |
| m_lastValues[1] = source[readIndex + 1]; |
| m_fillIndex = 2; |
| |
| // Wrap the virtual read index back to the start of the buffer. |
| virtualReadIndex -= readIndex; |
| |
| // Put local copy back into member variable. |
| m_virtualReadIndex = virtualReadIndex; |
| } |
| |
| void AudioResamplerKernel::reset() |
| { |
| m_virtualReadIndex = 0.0; |
| m_fillIndex = 0; |
| m_lastValues[0] = 0.0f; |
| m_lastValues[1] = 0.0f; |
| } |
| |
| double AudioResamplerKernel::rate() const |
| { |
| return m_resampler->rate(); |
| } |
| |
| } // namespace WebCore |
| |
| #endif // ENABLE(WEB_AUDIO) |