зеркало из https://github.com/mozilla/gecko-dev.git
Bug 815643 - Part 4: Add the Convolution processing implementation to the build system; r=roc
--HG-- extra : rebase_source : 2a5aa609334c67bb9b09090d9f681c5c3a940c5c
This commit is contained in:
Ehsan Akhgari
Родитель
e279e084bb
Коммит
573f70e559
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@ -39,20 +39,28 @@ WriteZeroesToAudioBlock(AudioChunk* aChunk, uint32_t aStart, uint32_t aLength)
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}
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}
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void AudioBufferAddWithScale(const float* aInput,
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float aScale,
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float* aOutput,
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uint32_t aSize)
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{
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if (aScale == 1.0f) {
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for (uint32_t i = 0; i < aSize; ++i) {
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aOutput[i] += aInput[i];
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}
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} else {
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for (uint32_t i = 0; i < aSize; ++i) {
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aOutput[i] += aInput[i]*aScale;
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}
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}
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}
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void
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AudioBlockAddChannelWithScale(const float aInput[WEBAUDIO_BLOCK_SIZE],
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float aScale,
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float aOutput[WEBAUDIO_BLOCK_SIZE])
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{
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if (aScale == 1.0f) {
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for (uint32_t i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
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aOutput[i] += aInput[i];
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}
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} else {
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for (uint32_t i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
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aOutput[i] += aInput[i]*aScale;
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}
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}
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AudioBufferAddWithScale(aInput, aScale, aOutput, WEBAUDIO_BLOCK_SIZE);
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}
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void
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@ -98,14 +106,23 @@ AudioBlockCopyChannelWithScale(const float aInput[WEBAUDIO_BLOCK_SIZE],
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}
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void
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AudioBlockInPlaceScale(float aBlock[WEBAUDIO_BLOCK_SIZE],
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uint32_t aChannelCount,
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float aScale)
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AudioBufferInPlaceScale(float aBlock[WEBAUDIO_BLOCK_SIZE],
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uint32_t aChannelCount,
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float aScale)
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{
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AudioBufferInPlaceScale(aBlock, aChannelCount, aScale, WEBAUDIO_BLOCK_SIZE);
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}
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void
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AudioBufferInPlaceScale(float* aBlock,
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uint32_t aChannelCount,
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float aScale,
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uint32_t aSize)
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{
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if (aScale == 1.0f) {
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return;
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}
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for (uint32_t i = 0; i < WEBAUDIO_BLOCK_SIZE * aChannelCount; ++i) {
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for (uint32_t i = 0; i < aSize * aChannelCount; ++i) {
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*aBlock++ *= aScale;
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}
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}
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@ -141,4 +158,16 @@ AudioBlockPanStereoToStereo(const float aInputL[WEBAUDIO_BLOCK_SIZE],
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}
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}
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}
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float
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AudioBufferSumOfSquares(const float* aInput, uint32_t aLength)
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{
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float sum = 0.0f;
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while (aLength--) {
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sum += *aInput * *aInput;
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++aInput;
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}
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return sum;
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}
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}
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@ -87,6 +87,14 @@ void AllocateAudioBlock(uint32_t aChannelCount, AudioChunk* aChunk);
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*/
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void WriteZeroesToAudioBlock(AudioChunk* aChunk, uint32_t aStart, uint32_t aLength);
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/**
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* Pointwise multiply-add operation. aScale == 1.0f should be optimized.
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*/
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void AudioBufferAddWithScale(const float* aInput,
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float aScale,
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float* aOutput,
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uint32_t aSize);
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/**
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* Pointwise multiply-add operation. aScale == 1.0f should be optimized.
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*/
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@ -121,9 +129,17 @@ void BufferComplexMultiply(const float* aInput,
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/**
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* In place gain. aScale == 1.0f should be optimized.
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*/
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void AudioBlockInPlaceScale(float aBlock[WEBAUDIO_BLOCK_SIZE],
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uint32_t aChannelCount,
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float aScale);
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void AudioBufferInPlaceScale(float aBlock[WEBAUDIO_BLOCK_SIZE],
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uint32_t aChannelCount,
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float aScale);
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/**
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* In place gain. aScale == 1.0f should be optimized.
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*/
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void AudioBufferInPlaceScale(float* aBlock,
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uint32_t aChannelCount,
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float aScale,
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uint32_t aSize);
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/**
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* Upmix a mono input to a stereo output, scaling the two output channels by two
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@ -147,6 +163,12 @@ AudioBlockPanStereoToStereo(const float aInputL[WEBAUDIO_BLOCK_SIZE],
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float aOutputL[WEBAUDIO_BLOCK_SIZE],
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float aOutputR[WEBAUDIO_BLOCK_SIZE]);
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/**
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* Return the sum of squares of all of the samples in the input.
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*/
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float
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AudioBufferSumOfSquares(const float* aInput, uint32_t aLength);
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/**
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* All methods of this class and its subclasses are called on the
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* MediaStreamGraph thread.
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@ -288,8 +288,8 @@ AnalyserNode::AppendChunk(const AudioChunk& aChunk)
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mBuffer.Elements() + mWriteIndex);
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}
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if (channelCount > 1) {
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AudioBlockInPlaceScale(mBuffer.Elements() + mWriteIndex, 1,
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1.0f / aChunk.mChannelData.Length());
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AudioBufferInPlaceScale(mBuffer.Elements() + mWriteIndex, 1,
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1.0f / aChunk.mChannelData.Length());
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}
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mWriteIndex += chunkDuration;
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MOZ_ASSERT(mWriteIndex <= bufferSize);
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@ -48,6 +48,16 @@ public:
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mFFTSize / 2 + 1);
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}
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void PerformPaddedFFT(const float* aData, size_t dataSize)
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{
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MOZ_ASSERT(dataSize <= FFTSize());
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nsTArray<float> paddedData;
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paddedData.SetLength(FFTSize());
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PodCopy(paddedData.Elements(), aData, dataSize);
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PodZero(paddedData.Elements() + dataSize, mFFTSize - dataSize);
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PerformFFT(paddedData.Elements());
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}
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void SetFFTSize(uint32_t aSize)
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{
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mFFTSize = aSize;
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@ -55,7 +65,7 @@ public:
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PodZero(mOutputBuffer.Elements(), aSize / 2 + 1);
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}
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float FFTSize() const
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uint32_t FFTSize() const
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{
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return mFFTSize;
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}
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@ -18,3 +18,4 @@ endif # !_MSC_VER
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FORCE_STATIC_LIB := 1
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include $(topsrcdir)/config/rules.mk
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include $(topsrcdir)/ipc/chromium/chromium-config.mk
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@ -332,7 +332,7 @@ PannerNodeEngine::DistanceAndConeGain(AudioChunk* aChunk, float aGain)
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float* samples = static_cast<float*>(const_cast<void*>(*aChunk->mChannelData.Elements()));
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uint32_t channelCount = aChunk->mChannelData.Length();
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AudioBlockInPlaceScale(samples, channelCount, aGain);
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AudioBufferInPlaceScale(samples, channelCount, aGain);
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}
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// This algorithm is specicied in the webaudio spec.
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@ -26,76 +26,45 @@
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* THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#include "config.h"
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#include "DirectConvolver.h"
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#include "mozilla/PodOperations.h"
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#if ENABLE(WEB_AUDIO)
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#include "core/platform/audio/DirectConvolver.h"
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#if OS(DARWIN)
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#include <Accelerate/Accelerate.h>
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#endif
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#include "core/platform/audio/VectorMath.h"
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using namespace mozilla;
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namespace WebCore {
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using namespace VectorMath;
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DirectConvolver::DirectConvolver(size_t inputBlockSize)
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: m_inputBlockSize(inputBlockSize)
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#if USE(WEBAUDIO_IPP)
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, m_overlayBuffer(inputBlockSize)
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#endif // USE(WEBAUDIO_IPP)
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, m_buffer(inputBlockSize * 2)
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{
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m_buffer.SetLength(inputBlockSize * 2);
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PodZero(m_buffer.Elements(), inputBlockSize * 2);
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}
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void DirectConvolver::process(AudioFloatArray* convolutionKernel, const float* sourceP, float* destP, size_t framesToProcess)
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void DirectConvolver::process(const nsTArray<float>* convolutionKernel, const float* sourceP, float* destP, size_t framesToProcess)
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{
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ASSERT(framesToProcess == m_inputBlockSize);
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MOZ_ASSERT(framesToProcess == m_inputBlockSize);
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if (framesToProcess != m_inputBlockSize)
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return;
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// Only support kernelSize <= m_inputBlockSize
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size_t kernelSize = convolutionKernel->size();
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ASSERT(kernelSize <= m_inputBlockSize);
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size_t kernelSize = convolutionKernel->Length();
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MOZ_ASSERT(kernelSize <= m_inputBlockSize);
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if (kernelSize > m_inputBlockSize)
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return;
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float* kernelP = convolutionKernel->data();
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const float* kernelP = convolutionKernel->Elements();
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// Sanity check
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bool isCopyGood = kernelP && sourceP && destP && m_buffer.data();
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ASSERT(isCopyGood);
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bool isCopyGood = kernelP && sourceP && destP && m_buffer.Elements();
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MOZ_ASSERT(isCopyGood);
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if (!isCopyGood)
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return;
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#if USE(WEBAUDIO_IPP)
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float* outputBuffer = m_buffer.data();
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float* overlayBuffer = m_overlayBuffer.data();
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bool isCopyGood2 = overlayBuffer && m_overlayBuffer.size() >= kernelSize && m_buffer.size() == m_inputBlockSize * 2;
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ASSERT(isCopyGood2);
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if (!isCopyGood2)
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return;
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ippsConv_32f(static_cast<const Ipp32f*>(sourceP), framesToProcess, static_cast<Ipp32f*>(kernelP), kernelSize, static_cast<Ipp32f*>(outputBuffer));
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vadd(outputBuffer, 1, overlayBuffer, 1, destP, 1, framesToProcess);
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memcpy(overlayBuffer, outputBuffer + m_inputBlockSize, sizeof(float) * kernelSize);
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#else
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float* inputP = m_buffer.data() + m_inputBlockSize;
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float* inputP = m_buffer.Elements() + m_inputBlockSize;
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// Copy samples to 2nd half of input buffer.
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memcpy(inputP, sourceP, sizeof(float) * framesToProcess);
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#if OS(DARWIN)
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#if defined(__ppc__) || defined(__i386__)
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conv(inputP - kernelSize + 1, 1, kernelP + kernelSize - 1, -1, destP, 1, framesToProcess, kernelSize);
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#else
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vDSP_conv(inputP - kernelSize + 1, 1, kernelP + kernelSize - 1, -1, destP, 1, framesToProcess, kernelSize);
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#endif // defined(__ppc__) || defined(__i386__)
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#else
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// FIXME: The macro can be further optimized to avoid pipeline stalls. One possibility is to maintain 4 separate sums and change the macro to CONVOLVE_FOUR_SAMPLES.
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#define CONVOLVE_ONE_SAMPLE \
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sum += inputP[i - j] * kernelP[j]; \
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@ -365,21 +334,14 @@ void DirectConvolver::process(AudioFloatArray* convolutionKernel, const float* s
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}
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destP[i++] = sum;
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}
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#endif // OS(DARWIN)
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// Copy 2nd half of input buffer to 1st half.
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memcpy(m_buffer.data(), inputP, sizeof(float) * framesToProcess);
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#endif
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memcpy(m_buffer.Elements(), inputP, sizeof(float) * framesToProcess);
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}
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void DirectConvolver::reset()
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{
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m_buffer.zero();
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#if USE(WEBAUDIO_IPP)
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m_overlayBuffer.zero();
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#endif // USE(WEBAUDIO_IPP)
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PodZero(m_buffer.Elements(), m_buffer.Length());
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}
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} // namespace WebCore
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#endif // ENABLE(WEB_AUDIO)
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@ -29,11 +29,7 @@
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#ifndef DirectConvolver_h
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#define DirectConvolver_h
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#include "core/platform/audio/AudioArray.h"
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#if USE(WEBAUDIO_IPP)
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#include <ipps.h>
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#endif // USE(WEBAUDIO_IPP)
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#include "nsTArray.h"
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namespace WebCore {
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@ -41,17 +37,14 @@ class DirectConvolver {
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public:
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DirectConvolver(size_t inputBlockSize);
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void process(AudioFloatArray* convolutionKernel, const float* sourceP, float* destP, size_t framesToProcess);
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void process(const nsTArray<float>* convolutionKernel, const float* sourceP, float* destP, size_t framesToProcess);
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void reset();
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private:
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size_t m_inputBlockSize;
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#if USE(WEBAUDIO_IPP)
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AudioFloatArray m_overlayBuffer;
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#endif // USE(WEBAUDIO_IPP)
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AudioFloatArray m_buffer;
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nsTArray<float> m_buffer;
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};
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} // namespace WebCore
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@ -26,35 +26,33 @@
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* THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#include "config.h"
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#include "FFTConvolver.h"
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#include "mozilla/PodOperations.h"
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#if ENABLE(WEB_AUDIO)
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#include "core/platform/audio/FFTConvolver.h"
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#include "core/platform/audio/VectorMath.h"
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using namespace mozilla;
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namespace WebCore {
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using namespace VectorMath;
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FFTConvolver::FFTConvolver(size_t fftSize)
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: m_frame(fftSize)
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, m_readWriteIndex(0)
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, m_inputBuffer(fftSize) // 2nd half of buffer is always zeroed
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, m_outputBuffer(fftSize)
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, m_lastOverlapBuffer(fftSize / 2)
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{
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m_inputBuffer.SetLength(fftSize);
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PodZero(m_inputBuffer.Elements(), fftSize);
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m_outputBuffer.SetLength(fftSize);
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PodZero(m_outputBuffer.Elements(), fftSize);
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m_lastOverlapBuffer.SetLength(fftSize / 2);
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PodZero(m_lastOverlapBuffer.Elements(), fftSize / 2);
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}
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void FFTConvolver::process(FFTFrame* fftKernel, const float* sourceP, float* destP, size_t framesToProcess)
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void FFTConvolver::process(FFTBlock* fftKernel, const float* sourceP, float* destP, size_t framesToProcess)
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{
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size_t halfSize = fftSize() / 2;
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// framesToProcess must be an exact multiple of halfSize,
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// or halfSize is a multiple of framesToProcess when halfSize > framesToProcess.
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bool isGood = !(halfSize % framesToProcess && framesToProcess % halfSize);
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ASSERT(isGood);
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MOZ_ASSERT(isGood);
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if (!isGood)
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return;
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@ -63,22 +61,22 @@ void FFTConvolver::process(FFTFrame* fftKernel, const float* sourceP, float* des
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for (size_t i = 0; i < numberOfDivisions; ++i, sourceP += divisionSize, destP += divisionSize) {
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// Copy samples to input buffer (note contraint above!)
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float* inputP = m_inputBuffer.data();
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float* inputP = m_inputBuffer.Elements();
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// Sanity check
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bool isCopyGood1 = sourceP && inputP && m_readWriteIndex + divisionSize <= m_inputBuffer.size();
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ASSERT(isCopyGood1);
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bool isCopyGood1 = sourceP && inputP && m_readWriteIndex + divisionSize <= m_inputBuffer.Length();
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MOZ_ASSERT(isCopyGood1);
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if (!isCopyGood1)
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return;
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memcpy(inputP + m_readWriteIndex, sourceP, sizeof(float) * divisionSize);
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// Copy samples from output buffer
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float* outputP = m_outputBuffer.data();
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float* outputP = m_outputBuffer.Elements();
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// Sanity check
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bool isCopyGood2 = destP && outputP && m_readWriteIndex + divisionSize <= m_outputBuffer.size();
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ASSERT(isCopyGood2);
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bool isCopyGood2 = destP && outputP && m_readWriteIndex + divisionSize <= m_outputBuffer.Length();
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MOZ_ASSERT(isCopyGood2);
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if (!isCopyGood2)
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return;
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@ -88,20 +86,21 @@ void FFTConvolver::process(FFTFrame* fftKernel, const float* sourceP, float* des
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// Check if it's time to perform the next FFT
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if (m_readWriteIndex == halfSize) {
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// The input buffer is now filled (get frequency-domain version)
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m_frame.doFFT(m_inputBuffer.data());
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m_frame.multiply(*fftKernel);
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m_frame.doInverseFFT(m_outputBuffer.data());
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m_frame.PerformFFT(m_inputBuffer.Elements());
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m_frame.Multiply(*fftKernel);
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m_frame.PerformInverseFFT(m_outputBuffer.Elements());
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// Overlap-add 1st half from previous time
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vadd(m_outputBuffer.data(), 1, m_lastOverlapBuffer.data(), 1, m_outputBuffer.data(), 1, halfSize);
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AudioBufferAddWithScale(m_lastOverlapBuffer.Elements(), 1.0f,
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m_outputBuffer.Elements(), halfSize);
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// Finally, save 2nd half of result
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bool isCopyGood3 = m_outputBuffer.size() == 2 * halfSize && m_lastOverlapBuffer.size() == halfSize;
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||||
ASSERT(isCopyGood3);
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||||
bool isCopyGood3 = m_outputBuffer.Length() == 2 * halfSize && m_lastOverlapBuffer.Length() == halfSize;
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MOZ_ASSERT(isCopyGood3);
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if (!isCopyGood3)
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return;
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memcpy(m_lastOverlapBuffer.data(), m_outputBuffer.data() + halfSize, sizeof(float) * halfSize);
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||||
memcpy(m_lastOverlapBuffer.Elements(), m_outputBuffer.Elements() + halfSize, sizeof(float) * halfSize);
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|
||||
// Reset index back to start for next time
|
||||
m_readWriteIndex = 0;
|
||||
|
|
@ -111,10 +110,8 @@ void FFTConvolver::process(FFTFrame* fftKernel, const float* sourceP, float* des
|
|||
|
||||
void FFTConvolver::reset()
|
||||
{
|
||||
m_lastOverlapBuffer.zero();
|
||||
PodZero(m_lastOverlapBuffer.Elements(), m_lastOverlapBuffer.Length());
|
||||
m_readWriteIndex = 0;
|
||||
}
|
||||
|
||||
} // namespace WebCore
|
||||
|
||||
#endif // ENABLE(WEB_AUDIO)
|
||||
|
|
|
|||
|
|
@ -29,11 +29,14 @@
|
|||
#ifndef FFTConvolver_h
|
||||
#define FFTConvolver_h
|
||||
|
||||
#include "core/platform/audio/AudioArray.h"
|
||||
#include "core/platform/audio/FFTFrame.h"
|
||||
#include "nsTArray.h"
|
||||
#include "mozilla/FFTBlock.h"
|
||||
|
||||
namespace WebCore {
|
||||
|
||||
typedef nsTArray<float> AudioFloatArray;
|
||||
using mozilla::FFTBlock;
|
||||
|
||||
class FFTConvolver {
|
||||
public:
|
||||
// fftSize must be a power of two
|
||||
|
|
@ -46,14 +49,14 @@ public:
|
|||
// The input to output latency is equal to fftSize / 2
|
||||
//
|
||||
// Processing in-place is allowed...
|
||||
void process(FFTFrame* fftKernel, const float* sourceP, float* destP, size_t framesToProcess);
|
||||
void process(FFTBlock* fftKernel, const float* sourceP, float* destP, size_t framesToProcess);
|
||||
|
||||
void reset();
|
||||
|
||||
size_t fftSize() const { return m_frame.fftSize(); }
|
||||
size_t fftSize() const { return m_frame.FFTSize(); }
|
||||
|
||||
private:
|
||||
FFTFrame m_frame;
|
||||
FFTBlock m_frame;
|
||||
|
||||
// Buffer input until we get fftSize / 2 samples then do an FFT
|
||||
size_t m_readWriteIndex;
|
||||
|
|
|
|||
|
|
@ -15,3 +15,4 @@ LIBXUL_LIBRARY := 1
|
|||
FORCE_STATIC_LIB := 1
|
||||
|
||||
include $(topsrcdir)/config/rules.mk
|
||||
include $(topsrcdir)/ipc/chromium/chromium-config.mk
|
||||
|
|
|
|||
|
|
@ -26,54 +26,39 @@
|
|||
* THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
||||
*/
|
||||
|
||||
#include "config.h"
|
||||
|
||||
#if ENABLE(WEB_AUDIO)
|
||||
|
||||
#include "core/platform/audio/Reverb.h"
|
||||
#include "Reverb.h"
|
||||
|
||||
#include <math.h>
|
||||
#include "core/platform/audio/AudioBus.h"
|
||||
#include "core/platform/audio/AudioFileReader.h"
|
||||
#include "core/platform/audio/ReverbConvolver.h"
|
||||
#include "core/platform/audio/VectorMath.h"
|
||||
#include <wtf/MathExtras.h>
|
||||
#include <wtf/OwnPtr.h>
|
||||
#include <wtf/PassOwnPtr.h>
|
||||
#include "ReverbConvolver.h"
|
||||
#include "mozilla/FloatingPoint.h"
|
||||
|
||||
#if OS(DARWIN)
|
||||
using namespace std;
|
||||
#endif
|
||||
using namespace mozilla;
|
||||
|
||||
namespace WebCore {
|
||||
|
||||
using namespace VectorMath;
|
||||
|
||||
// Empirical gain calibration tested across many impulse responses to ensure perceived volume is same as dry (unprocessed) signal
|
||||
const float GainCalibration = -58;
|
||||
const float GainCalibrationSampleRate = 44100;
|
||||
|
||||
// A minimum power value to when normalizing a silent (or very quiet) impulse response
|
||||
const float MinPower = 0.000125f;
|
||||
|
||||
static float calculateNormalizationScale(AudioBus* response)
|
||||
|
||||
static float calculateNormalizationScale(ThreadSharedFloatArrayBufferList* response, size_t aLength, float sampleRate)
|
||||
{
|
||||
// Normalize by RMS power
|
||||
size_t numberOfChannels = response->numberOfChannels();
|
||||
size_t length = response->length();
|
||||
size_t numberOfChannels = response->GetChannels();
|
||||
|
||||
float power = 0;
|
||||
|
||||
for (size_t i = 0; i < numberOfChannels; ++i) {
|
||||
float channelPower = 0;
|
||||
vsvesq(response->channel(i)->data(), 1, &channelPower, length);
|
||||
float channelPower = AudioBufferSumOfSquares(static_cast<const float*>(response->GetData(i)), aLength);
|
||||
power += channelPower;
|
||||
}
|
||||
|
||||
power = sqrt(power / (numberOfChannels * length));
|
||||
power = sqrt(power / (numberOfChannels * aLength));
|
||||
|
||||
// Protect against accidental overload
|
||||
if (std::isinf(power) || std::isnan(power) || power < MinPower)
|
||||
if (!IsFinite(power) || IsNaN(power) || power < MinPower)
|
||||
power = MinPower;
|
||||
|
||||
float scale = 1 / power;
|
||||
|
|
@ -81,163 +66,174 @@ static float calculateNormalizationScale(AudioBus* response)
|
|||
scale *= powf(10, GainCalibration * 0.05f); // calibrate to make perceived volume same as unprocessed
|
||||
|
||||
// Scale depends on sample-rate.
|
||||
if (response->sampleRate())
|
||||
scale *= GainCalibrationSampleRate / response->sampleRate();
|
||||
if (sampleRate)
|
||||
scale *= GainCalibrationSampleRate / sampleRate;
|
||||
|
||||
// True-stereo compensation
|
||||
if (response->numberOfChannels() == 4)
|
||||
if (response->GetChannels() == 4)
|
||||
scale *= 0.5f;
|
||||
|
||||
return scale;
|
||||
}
|
||||
|
||||
Reverb::Reverb(AudioBus* impulseResponse, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads, bool normalize)
|
||||
Reverb::Reverb(ThreadSharedFloatArrayBufferList* impulseResponse, size_t impulseResponseBufferLength, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads, bool normalize, float sampleRate)
|
||||
{
|
||||
float scale = 1;
|
||||
|
||||
if (normalize) {
|
||||
scale = calculateNormalizationScale(impulseResponse);
|
||||
scale = calculateNormalizationScale(impulseResponse, impulseResponseBufferLength, sampleRate);
|
||||
|
||||
if (scale)
|
||||
impulseResponse->scale(scale);
|
||||
if (scale) {
|
||||
for (uint32_t i = 0; i < impulseResponse->GetChannels(); ++i) {
|
||||
AudioBufferInPlaceScale(const_cast<float*>(impulseResponse->GetData(i)),
|
||||
1, scale, impulseResponseBufferLength);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
initialize(impulseResponse, renderSliceSize, maxFFTSize, numberOfChannels, useBackgroundThreads);
|
||||
initialize(impulseResponse, impulseResponseBufferLength, renderSliceSize, maxFFTSize, numberOfChannels, useBackgroundThreads);
|
||||
|
||||
// Undo scaling since this shouldn't be a destructive operation on impulseResponse.
|
||||
// FIXME: What about roundoff? Perhaps consider making a temporary scaled copy
|
||||
// instead of scaling and unscaling in place.
|
||||
if (normalize && scale)
|
||||
impulseResponse->scale(1 / scale);
|
||||
if (normalize && scale) {
|
||||
for (uint32_t i = 0; i < impulseResponse->GetChannels(); ++i) {
|
||||
AudioBufferInPlaceScale(const_cast<float*>(impulseResponse->GetData(i)),
|
||||
1, 1 / scale, impulseResponseBufferLength);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
void Reverb::initialize(AudioBus* impulseResponseBuffer, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads)
|
||||
void Reverb::initialize(ThreadSharedFloatArrayBufferList* impulseResponseBuffer, size_t impulseResponseBufferLength, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads)
|
||||
{
|
||||
m_impulseResponseLength = impulseResponseBuffer->length();
|
||||
m_impulseResponseLength = impulseResponseBufferLength;
|
||||
|
||||
// The reverb can handle a mono impulse response and still do stereo processing
|
||||
size_t numResponseChannels = impulseResponseBuffer->numberOfChannels();
|
||||
m_convolvers.reserveCapacity(numberOfChannels);
|
||||
size_t numResponseChannels = impulseResponseBuffer->GetChannels();
|
||||
m_convolvers.SetCapacity(numberOfChannels);
|
||||
|
||||
int convolverRenderPhase = 0;
|
||||
for (size_t i = 0; i < numResponseChannels; ++i) {
|
||||
AudioChannel* channel = impulseResponseBuffer->channel(i);
|
||||
const float* channel = impulseResponseBuffer->GetData(i);
|
||||
size_t length = impulseResponseBufferLength;
|
||||
|
||||
OwnPtr<ReverbConvolver> convolver = adoptPtr(new ReverbConvolver(channel, renderSliceSize, maxFFTSize, convolverRenderPhase, useBackgroundThreads));
|
||||
m_convolvers.append(convolver.release());
|
||||
nsAutoPtr<ReverbConvolver> convolver(new ReverbConvolver(channel, length, renderSliceSize, maxFFTSize, convolverRenderPhase, useBackgroundThreads));
|
||||
m_convolvers.AppendElement(convolver.forget());
|
||||
|
||||
convolverRenderPhase += renderSliceSize;
|
||||
}
|
||||
|
||||
// For "True" stereo processing we allocate a temporary buffer to avoid repeatedly allocating it in the process() method.
|
||||
// It can be bad to allocate memory in a real-time thread.
|
||||
if (numResponseChannels == 4)
|
||||
m_tempBuffer = AudioBus::create(2, MaxFrameSize);
|
||||
if (numResponseChannels == 4) {
|
||||
AllocateAudioBlock(2, &m_tempBuffer);
|
||||
WriteZeroesToAudioBlock(&m_tempBuffer, 0, WEBAUDIO_BLOCK_SIZE);
|
||||
}
|
||||
}
|
||||
|
||||
void Reverb::process(const AudioBus* sourceBus, AudioBus* destinationBus, size_t framesToProcess)
|
||||
void Reverb::process(const AudioChunk* sourceBus, AudioChunk* destinationBus, size_t framesToProcess)
|
||||
{
|
||||
// Do a fairly comprehensive sanity check.
|
||||
// If these conditions are satisfied, all of the source and destination pointers will be valid for the various matrixing cases.
|
||||
bool isSafeToProcess = sourceBus && destinationBus && sourceBus->numberOfChannels() > 0 && destinationBus->numberOfChannels() > 0
|
||||
&& framesToProcess <= MaxFrameSize && framesToProcess <= sourceBus->length() && framesToProcess <= destinationBus->length();
|
||||
|
||||
ASSERT(isSafeToProcess);
|
||||
bool isSafeToProcess = sourceBus && destinationBus && sourceBus->mChannelData.Length() > 0 && destinationBus->mChannelData.Length() > 0
|
||||
&& framesToProcess <= MaxFrameSize && framesToProcess <= size_t(sourceBus->mDuration) && framesToProcess <= size_t(destinationBus->mDuration);
|
||||
|
||||
MOZ_ASSERT(isSafeToProcess);
|
||||
if (!isSafeToProcess)
|
||||
return;
|
||||
|
||||
// For now only handle mono or stereo output
|
||||
if (destinationBus->numberOfChannels() > 2) {
|
||||
destinationBus->zero();
|
||||
if (destinationBus->mChannelData.Length() > 2) {
|
||||
destinationBus->SetNull(destinationBus->mDuration);
|
||||
return;
|
||||
}
|
||||
|
||||
AudioChannel* destinationChannelL = destinationBus->channel(0);
|
||||
const AudioChannel* sourceChannelL = sourceBus->channel(0);
|
||||
float* destinationChannelL = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[0]));
|
||||
const float* sourceBusL = static_cast<const float*>(sourceBus->mChannelData[0]);
|
||||
|
||||
// Handle input -> output matrixing...
|
||||
size_t numInputChannels = sourceBus->numberOfChannels();
|
||||
size_t numOutputChannels = destinationBus->numberOfChannels();
|
||||
size_t numReverbChannels = m_convolvers.size();
|
||||
size_t numInputChannels = sourceBus->mChannelData.Length();
|
||||
size_t numOutputChannels = destinationBus->mChannelData.Length();
|
||||
size_t numReverbChannels = m_convolvers.Length();
|
||||
|
||||
if (numInputChannels == 2 && numReverbChannels == 2 && numOutputChannels == 2) {
|
||||
// 2 -> 2 -> 2
|
||||
const AudioChannel* sourceChannelR = sourceBus->channel(1);
|
||||
AudioChannel* destinationChannelR = destinationBus->channel(1);
|
||||
m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
|
||||
m_convolvers[1]->process(sourceChannelR, destinationChannelR, framesToProcess);
|
||||
const float* sourceBusR = static_cast<const float*>(sourceBus->mChannelData[1]);
|
||||
float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
|
||||
m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
|
||||
m_convolvers[1]->process(sourceBusR, sourceBus->mDuration, destinationChannelR, destinationBus->mDuration, framesToProcess);
|
||||
} else if (numInputChannels == 1 && numOutputChannels == 2 && numReverbChannels == 2) {
|
||||
// 1 -> 2 -> 2
|
||||
for (int i = 0; i < 2; ++i) {
|
||||
AudioChannel* destinationChannel = destinationBus->channel(i);
|
||||
m_convolvers[i]->process(sourceChannelL, destinationChannel, framesToProcess);
|
||||
float* destinationChannel = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[i]));
|
||||
m_convolvers[i]->process(sourceBusL, sourceBus->mDuration, destinationChannel, destinationBus->mDuration, framesToProcess);
|
||||
}
|
||||
} else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 2) {
|
||||
// 1 -> 1 -> 2
|
||||
m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
|
||||
m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
|
||||
|
||||
// simply copy L -> R
|
||||
AudioChannel* destinationChannelR = destinationBus->channel(1);
|
||||
bool isCopySafe = destinationChannelL->data() && destinationChannelR->data() && destinationChannelL->length() >= framesToProcess && destinationChannelR->length() >= framesToProcess;
|
||||
ASSERT(isCopySafe);
|
||||
float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
|
||||
bool isCopySafe = destinationChannelL && destinationChannelR && size_t(destinationBus->mDuration) >= framesToProcess && size_t(destinationBus->mDuration) >= framesToProcess;
|
||||
MOZ_ASSERT(isCopySafe);
|
||||
if (!isCopySafe)
|
||||
return;
|
||||
memcpy(destinationChannelR->mutableData(), destinationChannelL->data(), sizeof(float) * framesToProcess);
|
||||
PodCopy(destinationChannelR, destinationChannelL, framesToProcess);
|
||||
} else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 1) {
|
||||
// 1 -> 1 -> 1
|
||||
m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
|
||||
m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
|
||||
} else if (numInputChannels == 2 && numReverbChannels == 4 && numOutputChannels == 2) {
|
||||
// 2 -> 4 -> 2 ("True" stereo)
|
||||
const AudioChannel* sourceChannelR = sourceBus->channel(1);
|
||||
AudioChannel* destinationChannelR = destinationBus->channel(1);
|
||||
const float* sourceBusR = static_cast<const float*>(sourceBus->mChannelData[1]);
|
||||
float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
|
||||
|
||||
AudioChannel* tempChannelL = m_tempBuffer->channel(0);
|
||||
AudioChannel* tempChannelR = m_tempBuffer->channel(1);
|
||||
float* tempChannelL = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[0]));
|
||||
float* tempChannelR = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[1]));
|
||||
|
||||
// Process left virtual source
|
||||
m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
|
||||
m_convolvers[1]->process(sourceChannelL, destinationChannelR, framesToProcess);
|
||||
m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
|
||||
m_convolvers[1]->process(sourceBusL, sourceBus->mDuration, destinationChannelR, destinationBus->mDuration, framesToProcess);
|
||||
|
||||
// Process right virtual source
|
||||
m_convolvers[2]->process(sourceChannelR, tempChannelL, framesToProcess);
|
||||
m_convolvers[3]->process(sourceChannelR, tempChannelR, framesToProcess);
|
||||
m_convolvers[2]->process(sourceBusR, sourceBus->mDuration, tempChannelL, m_tempBuffer.mDuration, framesToProcess);
|
||||
m_convolvers[3]->process(sourceBusR, sourceBus->mDuration, tempChannelR, m_tempBuffer.mDuration, framesToProcess);
|
||||
|
||||
destinationBus->sumFrom(*m_tempBuffer);
|
||||
AudioBufferAddWithScale(tempChannelL, 1.0f, destinationChannelL, sourceBus->mDuration);
|
||||
AudioBufferAddWithScale(tempChannelR, 1.0f, destinationChannelR, sourceBus->mDuration);
|
||||
} else if (numInputChannels == 1 && numReverbChannels == 4 && numOutputChannels == 2) {
|
||||
// 1 -> 4 -> 2 (Processing mono with "True" stereo impulse response)
|
||||
// This is an inefficient use of a four-channel impulse response, but we should handle the case.
|
||||
AudioChannel* destinationChannelR = destinationBus->channel(1);
|
||||
float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
|
||||
|
||||
AudioChannel* tempChannelL = m_tempBuffer->channel(0);
|
||||
AudioChannel* tempChannelR = m_tempBuffer->channel(1);
|
||||
float* tempChannelL = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[0]));
|
||||
float* tempChannelR = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[1]));
|
||||
|
||||
// Process left virtual source
|
||||
m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
|
||||
m_convolvers[1]->process(sourceChannelL, destinationChannelR, framesToProcess);
|
||||
m_convolvers[0]->process(sourceBusL, sourceBus->mDuration, destinationChannelL, destinationBus->mDuration, framesToProcess);
|
||||
m_convolvers[1]->process(sourceBusL, sourceBus->mDuration, destinationChannelR, destinationBus->mDuration, framesToProcess);
|
||||
|
||||
// Process right virtual source
|
||||
m_convolvers[2]->process(sourceChannelL, tempChannelL, framesToProcess);
|
||||
m_convolvers[3]->process(sourceChannelL, tempChannelR, framesToProcess);
|
||||
m_convolvers[2]->process(sourceBusL, sourceBus->mDuration, tempChannelL, m_tempBuffer.mDuration, framesToProcess);
|
||||
m_convolvers[3]->process(sourceBusL, sourceBus->mDuration, tempChannelR, m_tempBuffer.mDuration, framesToProcess);
|
||||
|
||||
destinationBus->sumFrom(*m_tempBuffer);
|
||||
AudioBufferAddWithScale(tempChannelL, 1.0f, destinationChannelL, sourceBus->mDuration);
|
||||
AudioBufferAddWithScale(tempChannelR, 1.0f, destinationChannelR, sourceBus->mDuration);
|
||||
} else {
|
||||
// Handle gracefully any unexpected / unsupported matrixing
|
||||
// FIXME: add code for 5.1 support...
|
||||
destinationBus->zero();
|
||||
destinationBus->SetNull(destinationBus->mDuration);
|
||||
}
|
||||
}
|
||||
|
||||
void Reverb::reset()
|
||||
{
|
||||
for (size_t i = 0; i < m_convolvers.size(); ++i)
|
||||
for (size_t i = 0; i < m_convolvers.Length(); ++i)
|
||||
m_convolvers[i]->reset();
|
||||
}
|
||||
|
||||
size_t Reverb::latencyFrames() const
|
||||
{
|
||||
return !m_convolvers.isEmpty() ? m_convolvers.first()->latencyFrames() : 0;
|
||||
return !m_convolvers.IsEmpty() ? m_convolvers[0]->latencyFrames() : 0;
|
||||
}
|
||||
|
||||
} // namespace WebCore
|
||||
|
||||
#endif // ENABLE(WEB_AUDIO)
|
||||
|
|
|
|||
|
|
@ -29,13 +29,16 @@
|
|||
#ifndef Reverb_h
|
||||
#define Reverb_h
|
||||
|
||||
#include "core/platform/audio/ReverbConvolver.h"
|
||||
#include <wtf/Vector.h>
|
||||
#include "ReverbConvolver.h"
|
||||
#include "nsAutoPtr.h"
|
||||
#include "nsTArray.h"
|
||||
|
||||
namespace mozilla {
|
||||
class ThreadSharedFloatArrayBufferList;
|
||||
}
|
||||
|
||||
namespace WebCore {
|
||||
|
||||
class AudioBus;
|
||||
|
||||
// Multi-channel convolution reverb with channel matrixing - one or more ReverbConvolver objects are used internally.
|
||||
|
||||
class Reverb {
|
||||
|
|
@ -43,23 +46,23 @@ public:
|
|||
enum { MaxFrameSize = 256 };
|
||||
|
||||
// renderSliceSize is a rendering hint, so the FFTs can be optimized to not all occur at the same time (very bad when rendering on a real-time thread).
|
||||
Reverb(AudioBus* impulseResponseBuffer, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads, bool normalize);
|
||||
Reverb(mozilla::ThreadSharedFloatArrayBufferList* impulseResponseBuffer, size_t impulseResponseBufferLength, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads, bool normalize, float sampleRate);
|
||||
|
||||
void process(const AudioBus* sourceBus, AudioBus* destinationBus, size_t framesToProcess);
|
||||
void process(const mozilla::AudioChunk* sourceBus, mozilla::AudioChunk* destinationBus, size_t framesToProcess);
|
||||
void reset();
|
||||
|
||||
size_t impulseResponseLength() const { return m_impulseResponseLength; }
|
||||
size_t latencyFrames() const;
|
||||
|
||||
private:
|
||||
void initialize(AudioBus* impulseResponseBuffer, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads);
|
||||
void initialize(mozilla::ThreadSharedFloatArrayBufferList* impulseResponseBuffer, size_t impulseResponseBufferLength, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads);
|
||||
|
||||
size_t m_impulseResponseLength;
|
||||
|
||||
Vector<OwnPtr<ReverbConvolver> > m_convolvers;
|
||||
nsTArray<nsAutoPtr<ReverbConvolver> > m_convolvers;
|
||||
|
||||
// For "True" stereo processing
|
||||
RefPtr<AudioBus> m_tempBuffer;
|
||||
mozilla::AudioChunk m_tempBuffer;
|
||||
};
|
||||
|
||||
} // namespace WebCore
|
||||
|
|
|
|||
|
|
@ -26,31 +26,29 @@
|
|||
* THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
||||
*/
|
||||
|
||||
#include "config.h"
|
||||
#include "ReverbAccumulationBuffer.h"
|
||||
#include "AudioNodeEngine.h"
|
||||
#include "mozilla/PodOperations.h"
|
||||
#include <algorithm>
|
||||
|
||||
#if ENABLE(WEB_AUDIO)
|
||||
|
||||
#include "core/platform/audio/ReverbAccumulationBuffer.h"
|
||||
|
||||
#include "core/platform/audio/VectorMath.h"
|
||||
using namespace mozilla;
|
||||
|
||||
namespace WebCore {
|
||||
|
||||
using namespace VectorMath;
|
||||
|
||||
ReverbAccumulationBuffer::ReverbAccumulationBuffer(size_t length)
|
||||
: m_buffer(length)
|
||||
, m_readIndex(0)
|
||||
: m_readIndex(0)
|
||||
, m_readTimeFrame(0)
|
||||
{
|
||||
m_buffer.SetLength(length);
|
||||
PodZero(m_buffer.Elements(), length);
|
||||
}
|
||||
|
||||
void ReverbAccumulationBuffer::readAndClear(float* destination, size_t numberOfFrames)
|
||||
{
|
||||
size_t bufferLength = m_buffer.size();
|
||||
size_t bufferLength = m_buffer.Length();
|
||||
bool isCopySafe = m_readIndex <= bufferLength && numberOfFrames <= bufferLength;
|
||||
|
||||
ASSERT(isCopySafe);
|
||||
|
||||
MOZ_ASSERT(isCopySafe);
|
||||
if (!isCopySafe)
|
||||
return;
|
||||
|
||||
|
|
@ -58,7 +56,7 @@ void ReverbAccumulationBuffer::readAndClear(float* destination, size_t numberOfF
|
|||
size_t numberOfFrames1 = std::min(numberOfFrames, framesAvailable);
|
||||
size_t numberOfFrames2 = numberOfFrames - numberOfFrames1;
|
||||
|
||||
float* source = m_buffer.data();
|
||||
float* source = m_buffer.Elements();
|
||||
memcpy(destination, source + m_readIndex, sizeof(float) * numberOfFrames1);
|
||||
memset(source + m_readIndex, 0, sizeof(float) * numberOfFrames1);
|
||||
|
||||
|
|
@ -75,13 +73,13 @@ void ReverbAccumulationBuffer::readAndClear(float* destination, size_t numberOfF
|
|||
void ReverbAccumulationBuffer::updateReadIndex(int* readIndex, size_t numberOfFrames) const
|
||||
{
|
||||
// Update caller's readIndex
|
||||
*readIndex = (*readIndex + numberOfFrames) % m_buffer.size();
|
||||
*readIndex = (*readIndex + numberOfFrames) % m_buffer.Length();
|
||||
}
|
||||
|
||||
int ReverbAccumulationBuffer::accumulate(float* source, size_t numberOfFrames, int* readIndex, size_t delayFrames)
|
||||
{
|
||||
size_t bufferLength = m_buffer.size();
|
||||
|
||||
size_t bufferLength = m_buffer.Length();
|
||||
|
||||
size_t writeIndex = (*readIndex + delayFrames) % bufferLength;
|
||||
|
||||
// Update caller's readIndex
|
||||
|
|
@ -91,29 +89,28 @@ int ReverbAccumulationBuffer::accumulate(float* source, size_t numberOfFrames, i
|
|||
size_t numberOfFrames1 = std::min(numberOfFrames, framesAvailable);
|
||||
size_t numberOfFrames2 = numberOfFrames - numberOfFrames1;
|
||||
|
||||
float* destination = m_buffer.data();
|
||||
float* destination = m_buffer.Elements();
|
||||
|
||||
bool isSafe = writeIndex <= bufferLength && numberOfFrames1 + writeIndex <= bufferLength && numberOfFrames2 <= bufferLength;
|
||||
ASSERT(isSafe);
|
||||
MOZ_ASSERT(isSafe);
|
||||
if (!isSafe)
|
||||
return 0;
|
||||
|
||||
vadd(source, 1, destination + writeIndex, 1, destination + writeIndex, 1, numberOfFrames1);
|
||||
AudioBufferAddWithScale(source, 1.0f, destination + writeIndex, numberOfFrames1);
|
||||
|
||||
// Handle wrap-around if necessary
|
||||
if (numberOfFrames2 > 0)
|
||||
vadd(source + numberOfFrames1, 1, destination, 1, destination, 1, numberOfFrames2);
|
||||
if (numberOfFrames2 > 0) {
|
||||
AudioBufferAddWithScale(source + numberOfFrames1, 1.0f, destination, numberOfFrames2);
|
||||
}
|
||||
|
||||
return writeIndex;
|
||||
}
|
||||
|
||||
void ReverbAccumulationBuffer::reset()
|
||||
{
|
||||
m_buffer.zero();
|
||||
PodZero(m_buffer.Elements(), m_buffer.Length());
|
||||
m_readIndex = 0;
|
||||
m_readTimeFrame = 0;
|
||||
}
|
||||
|
||||
} // namespace WebCore
|
||||
|
||||
#endif // ENABLE(WEB_AUDIO)
|
||||
|
|
|
|||
|
|
@ -29,10 +29,12 @@
|
|||
#ifndef ReverbAccumulationBuffer_h
|
||||
#define ReverbAccumulationBuffer_h
|
||||
|
||||
#include "core/platform/audio/AudioArray.h"
|
||||
#include "nsTArray.h"
|
||||
|
||||
namespace WebCore {
|
||||
|
||||
typedef nsTArray<float> AudioFloatArray;
|
||||
|
||||
// ReverbAccumulationBuffer is a circular delay buffer with one client reading from it and multiple clients
|
||||
// writing/accumulating to it at different delay offsets from the read position. The read operation will zero the memory
|
||||
// just read from the buffer, so it will be ready for accumulation the next time around.
|
||||
|
|
|
|||
|
|
@ -26,19 +26,19 @@
|
|||
* THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
||||
*/
|
||||
|
||||
#include "config.h"
|
||||
#include "ReverbConvolver.h"
|
||||
|
||||
#if ENABLE(WEB_AUDIO)
|
||||
using namespace mozilla;
|
||||
|
||||
#include "core/platform/audio/ReverbConvolver.h"
|
||||
|
||||
#include "core/platform/audio/AudioBus.h"
|
||||
#include "core/platform/audio/VectorMath.h"
|
||||
template<>
|
||||
struct RunnableMethodTraits<WebCore::ReverbConvolver>
|
||||
{
|
||||
static void RetainCallee(WebCore::ReverbConvolver* obj) {}
|
||||
static void ReleaseCallee(WebCore::ReverbConvolver* obj) {}
|
||||
};
|
||||
|
||||
namespace WebCore {
|
||||
|
||||
using namespace VectorMath;
|
||||
|
||||
const int InputBufferSize = 8 * 16384;
|
||||
|
||||
// We only process the leading portion of the impulse response in the real-time thread. We don't exceed this length.
|
||||
|
|
@ -53,20 +53,15 @@ const size_t RealtimeFrameLimit = 8192 + 4096; // ~278msec @ 44.1KHz
|
|||
const size_t MinFFTSize = 128;
|
||||
const size_t MaxRealtimeFFTSize = 2048;
|
||||
|
||||
static void backgroundThreadEntry(void* threadData)
|
||||
{
|
||||
ReverbConvolver* reverbConvolver = static_cast<ReverbConvolver*>(threadData);
|
||||
reverbConvolver->backgroundThreadEntry();
|
||||
}
|
||||
|
||||
ReverbConvolver::ReverbConvolver(AudioChannel* impulseResponse, size_t renderSliceSize, size_t maxFFTSize, size_t convolverRenderPhase, bool useBackgroundThreads)
|
||||
: m_impulseResponseLength(impulseResponse->length())
|
||||
, m_accumulationBuffer(impulseResponse->length() + renderSliceSize)
|
||||
ReverbConvolver::ReverbConvolver(const float* impulseResponseData, size_t impulseResponseLength, size_t renderSliceSize, size_t maxFFTSize, size_t convolverRenderPhase, bool useBackgroundThreads)
|
||||
: m_impulseResponseLength(impulseResponseLength)
|
||||
, m_accumulationBuffer(impulseResponseLength + renderSliceSize)
|
||||
, m_inputBuffer(InputBufferSize)
|
||||
, m_minFFTSize(MinFFTSize) // First stage will have this size - successive stages will double in size each time
|
||||
, m_maxFFTSize(maxFFTSize) // until we hit m_maxFFTSize
|
||||
, m_backgroundThread("ConvolverWorker")
|
||||
, m_backgroundThreadMonitor("ConvolverMonitor")
|
||||
, m_useBackgroundThreads(useBackgroundThreads)
|
||||
, m_backgroundThread(0)
|
||||
, m_wantsToExit(false)
|
||||
, m_moreInputBuffered(false)
|
||||
{
|
||||
|
|
@ -80,8 +75,8 @@ ReverbConvolver::ReverbConvolver(AudioChannel* impulseResponse, size_t renderSli
|
|||
// Otherwise, assume we're being run from a command-line tool.
|
||||
bool hasRealtimeConstraint = useBackgroundThreads;
|
||||
|
||||
const float* response = impulseResponse->data();
|
||||
size_t totalResponseLength = impulseResponse->length();
|
||||
const float* response = impulseResponseData;
|
||||
size_t totalResponseLength = impulseResponseLength;
|
||||
|
||||
// The total latency is zero because the direct-convolution is used in the leading portion.
|
||||
size_t reverbTotalLatency = 0;
|
||||
|
|
@ -102,15 +97,15 @@ ReverbConvolver::ReverbConvolver(AudioChannel* impulseResponse, size_t renderSli
|
|||
|
||||
bool useDirectConvolver = !stageOffset;
|
||||
|
||||
OwnPtr<ReverbConvolverStage> stage = adoptPtr(new ReverbConvolverStage(response, totalResponseLength, reverbTotalLatency, stageOffset, stageSize, fftSize, renderPhase, renderSliceSize, &m_accumulationBuffer, useDirectConvolver));
|
||||
nsAutoPtr<ReverbConvolverStage> stage(new ReverbConvolverStage(response, totalResponseLength, reverbTotalLatency, stageOffset, stageSize, fftSize, renderPhase, renderSliceSize, &m_accumulationBuffer, useDirectConvolver));
|
||||
|
||||
bool isBackgroundStage = false;
|
||||
|
||||
if (this->useBackgroundThreads() && stageOffset > RealtimeFrameLimit) {
|
||||
m_backgroundStages.append(stage.release());
|
||||
m_backgroundStages.AppendElement(stage.forget());
|
||||
isBackgroundStage = true;
|
||||
} else
|
||||
m_stages.append(stage.release());
|
||||
m_stages.AppendElement(stage.forget());
|
||||
|
||||
stageOffset += stageSize;
|
||||
++i;
|
||||
|
|
@ -128,24 +123,27 @@ ReverbConvolver::ReverbConvolver(AudioChannel* impulseResponse, size_t renderSli
|
|||
|
||||
// Start up background thread
|
||||
// FIXME: would be better to up the thread priority here. It doesn't need to be real-time, but higher than the default...
|
||||
if (this->useBackgroundThreads() && m_backgroundStages.size() > 0)
|
||||
m_backgroundThread = createThread(WebCore::backgroundThreadEntry, this, "convolution background thread");
|
||||
if (this->useBackgroundThreads() && m_backgroundStages.Length() > 0) {
|
||||
m_backgroundThread.Start();
|
||||
CancelableTask* task = NewRunnableMethod(this, &ReverbConvolver::backgroundThreadEntry);
|
||||
m_backgroundThread.message_loop()->PostTask(FROM_HERE, task);
|
||||
}
|
||||
}
|
||||
|
||||
ReverbConvolver::~ReverbConvolver()
|
||||
{
|
||||
// Wait for background thread to stop
|
||||
if (useBackgroundThreads() && m_backgroundThread) {
|
||||
if (useBackgroundThreads() && m_backgroundThread.IsRunning()) {
|
||||
m_wantsToExit = true;
|
||||
|
||||
// Wake up thread so it can return
|
||||
{
|
||||
MutexLocker locker(m_backgroundThreadLock);
|
||||
MonitorAutoLock locker(m_backgroundThreadMonitor);
|
||||
m_moreInputBuffered = true;
|
||||
m_backgroundThreadCondition.signal();
|
||||
locker.Notify();
|
||||
}
|
||||
|
||||
waitForThreadCompletion(m_backgroundThread);
|
||||
m_backgroundThread.Stop();
|
||||
}
|
||||
}
|
||||
|
||||
|
|
@ -153,11 +151,11 @@ void ReverbConvolver::backgroundThreadEntry()
|
|||
{
|
||||
while (!m_wantsToExit) {
|
||||
// Wait for realtime thread to give us more input
|
||||
m_moreInputBuffered = false;
|
||||
m_moreInputBuffered = false;
|
||||
{
|
||||
MutexLocker locker(m_backgroundThreadLock);
|
||||
MonitorAutoLock locker(m_backgroundThreadMonitor);
|
||||
while (!m_moreInputBuffered && !m_wantsToExit)
|
||||
m_backgroundThreadCondition.wait(m_backgroundThreadLock);
|
||||
locker.Wait();
|
||||
}
|
||||
|
||||
// Process all of the stages until their read indices reach the input buffer's write index
|
||||
|
|
@ -172,23 +170,25 @@ void ReverbConvolver::backgroundThreadEntry()
|
|||
const int SliceSize = MinFFTSize / 2;
|
||||
|
||||
// Accumulate contributions from each stage
|
||||
for (size_t i = 0; i < m_backgroundStages.size(); ++i)
|
||||
for (size_t i = 0; i < m_backgroundStages.Length(); ++i)
|
||||
m_backgroundStages[i]->processInBackground(this, SliceSize);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
void ReverbConvolver::process(const AudioChannel* sourceChannel, AudioChannel* destinationChannel, size_t framesToProcess)
|
||||
void ReverbConvolver::process(const float* sourceChannelData, size_t sourceChannelLength,
|
||||
float* destinationChannelData, size_t destinationChannelLength,
|
||||
size_t framesToProcess)
|
||||
{
|
||||
bool isSafe = sourceChannel && destinationChannel && sourceChannel->length() >= framesToProcess && destinationChannel->length() >= framesToProcess;
|
||||
ASSERT(isSafe);
|
||||
bool isSafe = sourceChannelData && destinationChannelData && sourceChannelLength >= framesToProcess && destinationChannelLength >= framesToProcess;
|
||||
MOZ_ASSERT(isSafe);
|
||||
if (!isSafe)
|
||||
return;
|
||||
|
||||
const float* source = sourceChannel->data();
|
||||
float* destination = destinationChannel->mutableData();
|
||||
|
||||
const float* source = sourceChannelData;
|
||||
float* destination = destinationChannelData;
|
||||
bool isDataSafe = source && destination;
|
||||
ASSERT(isDataSafe);
|
||||
MOZ_ASSERT(isDataSafe);
|
||||
if (!isDataSafe)
|
||||
return;
|
||||
|
||||
|
|
@ -196,32 +196,30 @@ void ReverbConvolver::process(const AudioChannel* sourceChannel, AudioChannel* d
|
|||
m_inputBuffer.write(source, framesToProcess);
|
||||
|
||||
// Accumulate contributions from each stage
|
||||
for (size_t i = 0; i < m_stages.size(); ++i)
|
||||
for (size_t i = 0; i < m_stages.Length(); ++i)
|
||||
m_stages[i]->process(source, framesToProcess);
|
||||
|
||||
// Finally read from accumulation buffer
|
||||
m_accumulationBuffer.readAndClear(destination, framesToProcess);
|
||||
|
||||
|
||||
// Now that we've buffered more input, wake up our background thread.
|
||||
|
||||
|
||||
// Not using a MutexLocker looks strange, but we use a tryLock() instead because this is run on the real-time
|
||||
// thread where it is a disaster for the lock to be contended (causes audio glitching). It's OK if we fail to
|
||||
// signal from time to time, since we'll get to it the next time we're called. We're called repeatedly
|
||||
// and frequently (around every 3ms). The background thread is processing well into the future and has a considerable amount of
|
||||
// leeway here...
|
||||
if (m_backgroundThreadLock.tryLock()) {
|
||||
m_moreInputBuffered = true;
|
||||
m_backgroundThreadCondition.signal();
|
||||
m_backgroundThreadLock.unlock();
|
||||
}
|
||||
MonitorAutoLock locker(m_backgroundThreadMonitor);
|
||||
m_moreInputBuffered = true;
|
||||
locker.Notify();
|
||||
}
|
||||
|
||||
void ReverbConvolver::reset()
|
||||
{
|
||||
for (size_t i = 0; i < m_stages.size(); ++i)
|
||||
for (size_t i = 0; i < m_stages.Length(); ++i)
|
||||
m_stages[i]->reset();
|
||||
|
||||
for (size_t i = 0; i < m_backgroundStages.size(); ++i)
|
||||
for (size_t i = 0; i < m_backgroundStages.Length(); ++i)
|
||||
m_backgroundStages[i]->reset();
|
||||
|
||||
m_accumulationBuffer.reset();
|
||||
|
|
@ -234,5 +232,3 @@ size_t ReverbConvolver::latencyFrames() const
|
|||
}
|
||||
|
||||
} // namespace WebCore
|
||||
|
||||
#endif // ENABLE(WEB_AUDIO)
|
||||
|
|
|
|||
|
|
@ -29,16 +29,19 @@
|
|||
#ifndef ReverbConvolver_h
|
||||
#define ReverbConvolver_h
|
||||
|
||||
#include "core/platform/audio/AudioArray.h"
|
||||
#include "core/platform/audio/DirectConvolver.h"
|
||||
#include "core/platform/audio/FFTConvolver.h"
|
||||
#include "core/platform/audio/ReverbAccumulationBuffer.h"
|
||||
#include "core/platform/audio/ReverbConvolverStage.h"
|
||||
#include "core/platform/audio/ReverbInputBuffer.h"
|
||||
#include <wtf/OwnPtr.h>
|
||||
#include <wtf/RefCounted.h>
|
||||
#include <wtf/Threading.h>
|
||||
#include <wtf/Vector.h>
|
||||
#include "DirectConvolver.h"
|
||||
#include "FFTConvolver.h"
|
||||
#include "ReverbAccumulationBuffer.h"
|
||||
#include "ReverbConvolverStage.h"
|
||||
#include "ReverbInputBuffer.h"
|
||||
#include "nsAutoPtr.h"
|
||||
#include "nsTArray.h"
|
||||
#include "nsCOMPtr.h"
|
||||
#include "mozilla/Monitor.h"
|
||||
#ifdef LOG
|
||||
#undef LOG
|
||||
#endif
|
||||
#include "base/thread.h"
|
||||
|
||||
namespace WebCore {
|
||||
|
||||
|
|
@ -50,10 +53,12 @@ public:
|
|||
// For certain tweaky de-convolving applications the phase errors add up quickly and lead to non-sensical results with
|
||||
// larger FFT sizes and single-precision floats. In these cases 2048 is a good size.
|
||||
// If not doing multi-threaded convolution, then should not go > 8192.
|
||||
ReverbConvolver(AudioChannel* impulseResponse, size_t renderSliceSize, size_t maxFFTSize, size_t convolverRenderPhase, bool useBackgroundThreads);
|
||||
ReverbConvolver(const float* impulseResponseData, size_t impulseResponseLength, size_t renderSliceSize, size_t maxFFTSize, size_t convolverRenderPhase, bool useBackgroundThreads);
|
||||
~ReverbConvolver();
|
||||
|
||||
void process(const AudioChannel* sourceChannel, AudioChannel* destinationChannel, size_t framesToProcess);
|
||||
void process(const float* sourceChannelData, size_t sourceChannelLength,
|
||||
float* destinationChannelData, size_t destinationChannelLength,
|
||||
size_t framesToProcess);
|
||||
void reset();
|
||||
|
||||
size_t impulseResponseLength() const { return m_impulseResponseLength; }
|
||||
|
|
@ -65,8 +70,8 @@ public:
|
|||
|
||||
size_t latencyFrames() const;
|
||||
private:
|
||||
Vector<OwnPtr<ReverbConvolverStage> > m_stages;
|
||||
Vector<OwnPtr<ReverbConvolverStage> > m_backgroundStages;
|
||||
nsTArray<nsAutoPtr<ReverbConvolverStage> > m_stages;
|
||||
nsTArray<nsAutoPtr<ReverbConvolverStage> > m_backgroundStages;
|
||||
size_t m_impulseResponseLength;
|
||||
|
||||
ReverbAccumulationBuffer m_accumulationBuffer;
|
||||
|
|
@ -82,12 +87,11 @@ private:
|
|||
size_t m_maxRealtimeFFTSize;
|
||||
|
||||
// Background thread and synchronization
|
||||
base::Thread m_backgroundThread;
|
||||
mozilla::Monitor m_backgroundThreadMonitor;
|
||||
bool m_useBackgroundThreads;
|
||||
ThreadIdentifier m_backgroundThread;
|
||||
bool m_wantsToExit;
|
||||
bool m_moreInputBuffered;
|
||||
mutable Mutex m_backgroundThreadLock;
|
||||
mutable ThreadCondition m_backgroundThreadCondition;
|
||||
};
|
||||
|
||||
} // namespace WebCore
|
||||
|
|
|
|||
|
|
@ -26,23 +26,17 @@
|
|||
* THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
||||
*/
|
||||
|
||||
#include "config.h"
|
||||
#include "ReverbConvolverStage.h"
|
||||
|
||||
#if ENABLE(WEB_AUDIO)
|
||||
#include "ReverbAccumulationBuffer.h"
|
||||
#include "ReverbConvolver.h"
|
||||
#include "ReverbInputBuffer.h"
|
||||
#include "mozilla/PodOperations.h"
|
||||
|
||||
#include "core/platform/audio/ReverbConvolverStage.h"
|
||||
|
||||
#include "core/platform/audio/ReverbAccumulationBuffer.h"
|
||||
#include "core/platform/audio/ReverbConvolver.h"
|
||||
#include "core/platform/audio/ReverbInputBuffer.h"
|
||||
#include "core/platform/audio/VectorMath.h"
|
||||
#include <wtf/OwnPtr.h>
|
||||
#include <wtf/PassOwnPtr.h>
|
||||
using namespace mozilla;
|
||||
|
||||
namespace WebCore {
|
||||
|
||||
using namespace VectorMath;
|
||||
|
||||
ReverbConvolverStage::ReverbConvolverStage(const float* impulseResponse, size_t, size_t reverbTotalLatency, size_t stageOffset, size_t stageLength,
|
||||
size_t fftSize, size_t renderPhase, size_t renderSliceSize, ReverbAccumulationBuffer* accumulationBuffer, bool directMode)
|
||||
: m_accumulationBuffer(accumulationBuffer)
|
||||
|
|
@ -50,19 +44,20 @@ ReverbConvolverStage::ReverbConvolverStage(const float* impulseResponse, size_t,
|
|||
, m_inputReadIndex(0)
|
||||
, m_directMode(directMode)
|
||||
{
|
||||
ASSERT(impulseResponse);
|
||||
ASSERT(accumulationBuffer);
|
||||
MOZ_ASSERT(impulseResponse);
|
||||
MOZ_ASSERT(accumulationBuffer);
|
||||
|
||||
if (!m_directMode) {
|
||||
m_fftKernel = adoptPtr(new FFTFrame(fftSize));
|
||||
m_fftKernel->doPaddedFFT(impulseResponse + stageOffset, stageLength);
|
||||
m_fftConvolver = adoptPtr(new FFTConvolver(fftSize));
|
||||
m_fftKernel = new FFTBlock(fftSize);
|
||||
m_fftKernel->PerformPaddedFFT(impulseResponse + stageOffset, stageLength);
|
||||
m_fftConvolver = new FFTConvolver(fftSize);
|
||||
} else {
|
||||
m_directKernel = adoptPtr(new AudioFloatArray(fftSize / 2));
|
||||
m_directKernel->copyToRange(impulseResponse + stageOffset, 0, fftSize / 2);
|
||||
m_directConvolver = adoptPtr(new DirectConvolver(renderSliceSize));
|
||||
m_directKernel.SetLength(fftSize / 2);
|
||||
PodCopy(m_directKernel.Elements(), impulseResponse + stageOffset, fftSize / 2);
|
||||
m_directConvolver = new DirectConvolver(renderSliceSize);
|
||||
}
|
||||
m_temporaryBuffer.allocate(renderSliceSize);
|
||||
m_temporaryBuffer.SetLength(renderSliceSize);
|
||||
PodZero(m_temporaryBuffer.Elements(), m_temporaryBuffer.Length());
|
||||
|
||||
// The convolution stage at offset stageOffset needs to have a corresponding delay to cancel out the offset.
|
||||
size_t totalDelay = stageOffset + reverbTotalLatency;
|
||||
|
|
@ -70,7 +65,7 @@ ReverbConvolverStage::ReverbConvolverStage(const float* impulseResponse, size_t,
|
|||
// But, the FFT convolution itself incurs fftSize / 2 latency, so subtract this out...
|
||||
size_t halfSize = fftSize / 2;
|
||||
if (!m_directMode) {
|
||||
ASSERT(totalDelay >= halfSize);
|
||||
MOZ_ASSERT(totalDelay >= halfSize);
|
||||
if (totalDelay >= halfSize)
|
||||
totalDelay -= halfSize;
|
||||
}
|
||||
|
|
@ -88,7 +83,8 @@ ReverbConvolverStage::ReverbConvolverStage(const float* impulseResponse, size_t,
|
|||
|
||||
size_t delayBufferSize = m_preDelayLength < fftSize ? fftSize : m_preDelayLength;
|
||||
delayBufferSize = delayBufferSize < renderSliceSize ? renderSliceSize : delayBufferSize;
|
||||
m_preDelayBuffer.allocate(delayBufferSize);
|
||||
m_preDelayBuffer.SetLength(delayBufferSize);
|
||||
PodZero(m_preDelayBuffer.Elements(), m_preDelayBuffer.Length());
|
||||
}
|
||||
|
||||
void ReverbConvolverStage::processInBackground(ReverbConvolver* convolver, size_t framesToProcess)
|
||||
|
|
@ -100,7 +96,7 @@ void ReverbConvolverStage::processInBackground(ReverbConvolver* convolver, size_
|
|||
|
||||
void ReverbConvolverStage::process(const float* source, size_t framesToProcess)
|
||||
{
|
||||
ASSERT(source);
|
||||
MOZ_ASSERT(source);
|
||||
if (!source)
|
||||
return;
|
||||
|
||||
|
|
@ -112,26 +108,26 @@ void ReverbConvolverStage::process(const float* source, size_t framesToProcess)
|
|||
bool isTemporaryBufferSafe = false;
|
||||
if (m_preDelayLength > 0) {
|
||||
// Handles both the read case (call to process() ) and the write case (memcpy() )
|
||||
bool isPreDelaySafe = m_preReadWriteIndex + framesToProcess <= m_preDelayBuffer.size();
|
||||
ASSERT(isPreDelaySafe);
|
||||
bool isPreDelaySafe = m_preReadWriteIndex + framesToProcess <= m_preDelayBuffer.Length();
|
||||
MOZ_ASSERT(isPreDelaySafe);
|
||||
if (!isPreDelaySafe)
|
||||
return;
|
||||
|
||||
isTemporaryBufferSafe = framesToProcess <= m_temporaryBuffer.size();
|
||||
isTemporaryBufferSafe = framesToProcess <= m_temporaryBuffer.Length();
|
||||
|
||||
preDelayedDestination = m_preDelayBuffer.data() + m_preReadWriteIndex;
|
||||
preDelayedDestination = m_preDelayBuffer.Elements() + m_preReadWriteIndex;
|
||||
preDelayedSource = preDelayedDestination;
|
||||
temporaryBuffer = m_temporaryBuffer.data();
|
||||
temporaryBuffer = m_temporaryBuffer.Elements();
|
||||
} else {
|
||||
// Zero delay
|
||||
preDelayedDestination = 0;
|
||||
preDelayedSource = source;
|
||||
temporaryBuffer = m_preDelayBuffer.data();
|
||||
temporaryBuffer = m_preDelayBuffer.Elements();
|
||||
|
||||
isTemporaryBufferSafe = framesToProcess <= m_preDelayBuffer.size();
|
||||
isTemporaryBufferSafe = framesToProcess <= m_preDelayBuffer.Length();
|
||||
}
|
||||
|
||||
ASSERT(isTemporaryBufferSafe);
|
||||
MOZ_ASSERT(isTemporaryBufferSafe);
|
||||
if (!isTemporaryBufferSafe)
|
||||
return;
|
||||
|
||||
|
|
@ -144,9 +140,9 @@ void ReverbConvolverStage::process(const float* source, size_t framesToProcess)
|
|||
// An expensive FFT will happen every fftSize / 2 frames.
|
||||
// We process in-place here...
|
||||
if (!m_directMode)
|
||||
m_fftConvolver->process(m_fftKernel.get(), preDelayedSource, temporaryBuffer, framesToProcess);
|
||||
m_fftConvolver->process(m_fftKernel, preDelayedSource, temporaryBuffer, framesToProcess);
|
||||
else
|
||||
m_directConvolver->process(m_directKernel.get(), preDelayedSource, temporaryBuffer, framesToProcess);
|
||||
m_directConvolver->process(&m_directKernel, preDelayedSource, temporaryBuffer, framesToProcess);
|
||||
|
||||
// Now accumulate into reverb's accumulation buffer.
|
||||
m_accumulationBuffer->accumulate(temporaryBuffer, framesToProcess, &m_accumulationReadIndex, m_postDelayLength);
|
||||
|
|
@ -157,7 +153,7 @@ void ReverbConvolverStage::process(const float* source, size_t framesToProcess)
|
|||
memcpy(preDelayedDestination, source, sizeof(float) * framesToProcess);
|
||||
m_preReadWriteIndex += framesToProcess;
|
||||
|
||||
ASSERT(m_preReadWriteIndex <= m_preDelayLength);
|
||||
MOZ_ASSERT(m_preReadWriteIndex <= m_preDelayLength);
|
||||
if (m_preReadWriteIndex >= m_preDelayLength)
|
||||
m_preReadWriteIndex = 0;
|
||||
}
|
||||
|
|
@ -171,12 +167,10 @@ void ReverbConvolverStage::reset()
|
|||
m_fftConvolver->reset();
|
||||
else
|
||||
m_directConvolver->reset();
|
||||
m_preDelayBuffer.zero();
|
||||
PodZero(m_preDelayBuffer.Elements(), m_preDelayBuffer.Length());
|
||||
m_accumulationReadIndex = 0;
|
||||
m_inputReadIndex = 0;
|
||||
m_framesProcessed = 0;
|
||||
}
|
||||
|
||||
} // namespace WebCore
|
||||
|
||||
#endif // ENABLE(WEB_AUDIO)
|
||||
|
|
|
|||
|
|
@ -29,17 +29,18 @@
|
|||
#ifndef ReverbConvolverStage_h
|
||||
#define ReverbConvolverStage_h
|
||||
|
||||
#include "core/platform/audio/AudioArray.h"
|
||||
#include "core/platform/audio/FFTFrame.h"
|
||||
#include <wtf/OwnPtr.h>
|
||||
#include "nsTArray.h"
|
||||
#include "mozilla/FFTBlock.h"
|
||||
|
||||
namespace WebCore {
|
||||
|
||||
using mozilla::FFTBlock;
|
||||
|
||||
class ReverbAccumulationBuffer;
|
||||
class ReverbConvolver;
|
||||
class FFTConvolver;
|
||||
class DirectConvolver;
|
||||
|
||||
|
||||
// A ReverbConvolverStage represents the convolution associated with a sub-section of a large impulse response.
|
||||
// It incorporates a delay line to account for the offset of the sub-section within the larger impulse response.
|
||||
class ReverbConvolverStage {
|
||||
|
|
@ -59,10 +60,10 @@ public:
|
|||
int inputReadIndex() const { return m_inputReadIndex; }
|
||||
|
||||
private:
|
||||
OwnPtr<FFTFrame> m_fftKernel;
|
||||
OwnPtr<FFTConvolver> m_fftConvolver;
|
||||
nsAutoPtr<FFTBlock> m_fftKernel;
|
||||
nsAutoPtr<FFTConvolver> m_fftConvolver;
|
||||
|
||||
AudioFloatArray m_preDelayBuffer;
|
||||
nsTArray<float> m_preDelayBuffer;
|
||||
|
||||
ReverbAccumulationBuffer* m_accumulationBuffer;
|
||||
int m_accumulationReadIndex;
|
||||
|
|
@ -73,11 +74,11 @@ private:
|
|||
size_t m_preReadWriteIndex;
|
||||
size_t m_framesProcessed;
|
||||
|
||||
AudioFloatArray m_temporaryBuffer;
|
||||
nsTArray<float> m_temporaryBuffer;
|
||||
|
||||
bool m_directMode;
|
||||
OwnPtr<AudioFloatArray> m_directKernel;
|
||||
OwnPtr<DirectConvolver> m_directConvolver;
|
||||
nsTArray<float> m_directKernel;
|
||||
nsAutoPtr<DirectConvolver> m_directConvolver;
|
||||
};
|
||||
|
||||
} // namespace WebCore
|
||||
|
|
|
|||
|
|
@ -26,32 +26,32 @@
|
|||
* THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
||||
*/
|
||||
|
||||
#include "config.h"
|
||||
#include "ReverbInputBuffer.h"
|
||||
#include "mozilla/PodOperations.h"
|
||||
|
||||
#if ENABLE(WEB_AUDIO)
|
||||
|
||||
#include "core/platform/audio/ReverbInputBuffer.h"
|
||||
using namespace mozilla;
|
||||
|
||||
namespace WebCore {
|
||||
|
||||
ReverbInputBuffer::ReverbInputBuffer(size_t length)
|
||||
: m_buffer(length)
|
||||
, m_writeIndex(0)
|
||||
: m_writeIndex(0)
|
||||
{
|
||||
m_buffer.SetLength(length);
|
||||
PodZero(m_buffer.Elements(), length);
|
||||
}
|
||||
|
||||
void ReverbInputBuffer::write(const float* sourceP, size_t numberOfFrames)
|
||||
{
|
||||
size_t bufferLength = m_buffer.size();
|
||||
size_t bufferLength = m_buffer.Length();
|
||||
bool isCopySafe = m_writeIndex + numberOfFrames <= bufferLength;
|
||||
ASSERT(isCopySafe);
|
||||
MOZ_ASSERT(isCopySafe);
|
||||
if (!isCopySafe)
|
||||
return;
|
||||
|
||||
memcpy(m_buffer.data() + m_writeIndex, sourceP, sizeof(float) * numberOfFrames);
|
||||
|
||||
memcpy(m_buffer.Elements() + m_writeIndex, sourceP, sizeof(float) * numberOfFrames);
|
||||
|
||||
m_writeIndex += numberOfFrames;
|
||||
ASSERT(m_writeIndex <= bufferLength);
|
||||
MOZ_ASSERT(m_writeIndex <= bufferLength);
|
||||
|
||||
if (m_writeIndex >= bufferLength)
|
||||
m_writeIndex = 0;
|
||||
|
|
@ -59,17 +59,17 @@ void ReverbInputBuffer::write(const float* sourceP, size_t numberOfFrames)
|
|||
|
||||
float* ReverbInputBuffer::directReadFrom(int* readIndex, size_t numberOfFrames)
|
||||
{
|
||||
size_t bufferLength = m_buffer.size();
|
||||
size_t bufferLength = m_buffer.Length();
|
||||
bool isPointerGood = readIndex && *readIndex >= 0 && *readIndex + numberOfFrames <= bufferLength;
|
||||
ASSERT(isPointerGood);
|
||||
MOZ_ASSERT(isPointerGood);
|
||||
if (!isPointerGood) {
|
||||
// Should never happen in practice but return pointer to start of buffer (avoid crash)
|
||||
if (readIndex)
|
||||
*readIndex = 0;
|
||||
return m_buffer.data();
|
||||
return m_buffer.Elements();
|
||||
}
|
||||
|
||||
float* sourceP = m_buffer.data();
|
||||
|
||||
float* sourceP = m_buffer.Elements();
|
||||
float* p = sourceP + *readIndex;
|
||||
|
||||
// Update readIndex
|
||||
|
|
@ -80,10 +80,8 @@ float* ReverbInputBuffer::directReadFrom(int* readIndex, size_t numberOfFrames)
|
|||
|
||||
void ReverbInputBuffer::reset()
|
||||
{
|
||||
m_buffer.zero();
|
||||
PodZero(m_buffer.Elements(), m_buffer.Length());
|
||||
m_writeIndex = 0;
|
||||
}
|
||||
|
||||
} // namespace WebCore
|
||||
|
||||
#endif // ENABLE(WEB_AUDIO)
|
||||
|
|
|
|||
|
|
@ -29,7 +29,7 @@
|
|||
#ifndef ReverbInputBuffer_h
|
||||
#define ReverbInputBuffer_h
|
||||
|
||||
#include "core/platform/audio/AudioArray.h"
|
||||
#include "nsTArray.h"
|
||||
|
||||
namespace WebCore {
|
||||
|
||||
|
|
@ -55,7 +55,7 @@ public:
|
|||
void reset();
|
||||
|
||||
private:
|
||||
AudioFloatArray m_buffer;
|
||||
nsTArray<float> m_buffer;
|
||||
size_t m_writeIndex;
|
||||
};
|
||||
|
||||
|
|
|
|||
|
|
@ -8,8 +8,15 @@ MODULE = 'content'
|
|||
|
||||
CPP_SOURCES += [
|
||||
'Biquad.cpp',
|
||||
'DirectConvolver.cpp',
|
||||
'DynamicsCompressor.cpp',
|
||||
'DynamicsCompressorKernel.cpp',
|
||||
'FFTConvolver.cpp',
|
||||
'Reverb.cpp',
|
||||
'ReverbAccumulationBuffer.cpp',
|
||||
'ReverbConvolver.cpp',
|
||||
'ReverbConvolverStage.cpp',
|
||||
'ReverbInputBuffer.cpp',
|
||||
'ZeroPole.cpp',
|
||||
]
|
||||
|
||||
|
|
|
|||
|
|
@ -17,6 +17,10 @@ EXPORTS += [
|
|||
'WebAudioUtils.h',
|
||||
]
|
||||
|
||||
EXPORTS.mozilla += [
|
||||
'FFTBlock.h',
|
||||
]
|
||||
|
||||
EXPORTS.mozilla.dom += [
|
||||
'AnalyserNode.h',
|
||||
'AudioBuffer.h',
|
||||
|
|
|
|||
Загрузка…
Ссылка в новой задаче