mirror of
https://github.com/mozilla/gecko-dev.git
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Backed out changesets aee6b1a6c400 and a633e4d67d31 (bug 875277) for bustage.
This commit is contained in:
Ryan VanderMeulen
parent
08b80c546e
commit
87a4ee15f4
@ -10,7 +10,6 @@
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#include "AudioNodeEngine.h"
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#include "AudioNodeStream.h"
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#include "mozilla/PodOperations.h"
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#include "speex/speex_resampler.h"
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namespace mozilla {
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namespace dom {
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@ -37,171 +36,19 @@ NS_INTERFACE_MAP_END_INHERITING(AudioNode)
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NS_IMPL_ADDREF_INHERITED(WaveShaperNode, AudioNode)
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NS_IMPL_RELEASE_INHERITED(WaveShaperNode, AudioNode)
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static uint32_t ValueOf(OverSampleType aType)
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{
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switch (aType) {
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case OverSampleType::None: return 1;
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case OverSampleType::_2x: return 2;
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case OverSampleType::_4x: return 4;
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default:
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NS_NOTREACHED("We should never reach here");
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return 1;
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}
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}
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class Resampler
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{
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public:
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Resampler()
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: mType(OverSampleType::None)
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, mUpSampler(nullptr)
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, mDownSampler(nullptr)
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, mChannels(0)
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, mSampleRate(0)
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{
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}
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~Resampler()
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{
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Destroy();
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}
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void Reset(uint32_t aChannels, TrackRate aSampleRate, OverSampleType aType)
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{
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if (aChannels == mChannels &&
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aSampleRate == mSampleRate &&
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aType == mType) {
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return;
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}
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mChannels = aChannels;
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mSampleRate = aSampleRate;
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mType = aType;
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Destroy();
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if (aType == OverSampleType::None) {
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mBuffer.Clear();
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return;
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}
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mUpSampler = speex_resampler_init(aChannels,
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aSampleRate,
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aSampleRate * ValueOf(aType),
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SPEEX_RESAMPLER_QUALITY_DEFAULT,
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nullptr);
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mDownSampler = speex_resampler_init(aChannels,
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aSampleRate * ValueOf(aType),
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aSampleRate,
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SPEEX_RESAMPLER_QUALITY_DEFAULT,
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nullptr);
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mBuffer.SetLength(WEBAUDIO_BLOCK_SIZE*ValueOf(aType));
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}
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float* UpSample(uint32_t aChannel, const float* aInputData, uint32_t aBlocks)
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{
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uint32_t inSamples = WEBAUDIO_BLOCK_SIZE;
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uint32_t outSamples = WEBAUDIO_BLOCK_SIZE*aBlocks;
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float* outputData = mBuffer.Elements();
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MOZ_ASSERT(mBuffer.Length() == outSamples);
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speex_resampler_process_float(mUpSampler, aChannel,
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aInputData, &inSamples,
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outputData, &outSamples);
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MOZ_ASSERT(inSamples == WEBAUDIO_BLOCK_SIZE && outSamples == WEBAUDIO_BLOCK_SIZE*aBlocks);
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return outputData;
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}
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void DownSample(uint32_t aChannel, float* aOutputData, uint32_t aBlocks)
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{
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uint32_t inSamples = WEBAUDIO_BLOCK_SIZE*aBlocks;
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uint32_t outSamples = WEBAUDIO_BLOCK_SIZE;
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const float* inputData = mBuffer.Elements();
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MOZ_ASSERT(mBuffer.Length() == inSamples);
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speex_resampler_process_float(mDownSampler, aChannel,
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inputData, &inSamples,
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aOutputData, &outSamples);
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MOZ_ASSERT(inSamples == WEBAUDIO_BLOCK_SIZE*aBlocks && outSamples == WEBAUDIO_BLOCK_SIZE);
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}
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private:
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void Destroy()
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{
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if (mUpSampler) {
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speex_resampler_destroy(mUpSampler);
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mUpSampler = nullptr;
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}
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if (mDownSampler) {
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speex_resampler_destroy(mDownSampler);
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mDownSampler = nullptr;
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}
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}
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private:
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OverSampleType mType;
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SpeexResamplerState* mUpSampler;
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SpeexResamplerState* mDownSampler;
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uint32_t mChannels;
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TrackRate mSampleRate;
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nsTArray<float> mBuffer;
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};
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class WaveShaperNodeEngine : public AudioNodeEngine
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{
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public:
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explicit WaveShaperNodeEngine(AudioNode* aNode)
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: AudioNodeEngine(aNode)
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, mType(OverSampleType::None)
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{
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}
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enum Parameteres {
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TYPE
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};
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virtual void SetRawArrayData(nsTArray<float>& aCurve) MOZ_OVERRIDE
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{
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mCurve.SwapElements(aCurve);
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}
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virtual void SetInt32Parameter(uint32_t aIndex, int32_t aValue) MOZ_OVERRIDE
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{
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switch (aIndex) {
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case TYPE:
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mType = static_cast<OverSampleType>(aValue);
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break;
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default:
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NS_ERROR("Bad WaveShaperNode Int32Parameter");
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}
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}
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template <uint32_t blocks>
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void ProcessCurve(const float* aInputBuffer, float* aOutputBuffer)
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{
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for (uint32_t j = 0; j < WEBAUDIO_BLOCK_SIZE*blocks; ++j) {
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// Index into the curve array based on the amplitude of the
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// incoming signal by clamping the amplitude to [-1, 1] and
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// performing a linear interpolation of the neighbor values.
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float index = std::max(0.0f, std::min(float(mCurve.Length() - 1),
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mCurve.Length() * (aInputBuffer[j] + 1) / 2));
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uint32_t indexLower = uint32_t(index);
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uint32_t indexHigher = uint32_t(index + 1.0f);
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if (indexHigher == mCurve.Length()) {
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aOutputBuffer[j] = mCurve[indexLower];
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} else {
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float interpolationFactor = index - indexLower;
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aOutputBuffer[j] = (1.0f - interpolationFactor) * mCurve[indexLower] +
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interpolationFactor * mCurve[indexHigher];
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}
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}
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}
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virtual void ProduceAudioBlock(AudioNodeStream* aStream,
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const AudioChunk& aInput,
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AudioChunk* aOutput,
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@ -219,35 +66,27 @@ public:
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for (uint32_t i = 0; i < channelCount; ++i) {
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const float* inputBuffer = static_cast<const float*>(aInput.mChannelData[i]);
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float* outputBuffer = const_cast<float*> (static_cast<const float*>(aOutput->mChannelData[i]));
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float* sampleBuffer;
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switch (mType) {
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case OverSampleType::None:
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mResampler.Reset(channelCount, aStream->SampleRate(), OverSampleType::None);
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ProcessCurve<1>(inputBuffer, outputBuffer);
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break;
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case OverSampleType::_2x:
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mResampler.Reset(channelCount, aStream->SampleRate(), OverSampleType::_2x);
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sampleBuffer = mResampler.UpSample(i, inputBuffer, 2);
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ProcessCurve<2>(sampleBuffer, sampleBuffer);
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mResampler.DownSample(i, outputBuffer, 2);
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break;
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case OverSampleType::_4x:
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mResampler.Reset(channelCount, aStream->SampleRate(), OverSampleType::_4x);
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sampleBuffer = mResampler.UpSample(i, inputBuffer, 4);
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ProcessCurve<4>(sampleBuffer, sampleBuffer);
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mResampler.DownSample(i, outputBuffer, 4);
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break;
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default:
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NS_NOTREACHED("We should never reach here");
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for (uint32_t j = 0; j < WEBAUDIO_BLOCK_SIZE; ++j) {
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// Index into the curve array based on the amplitude of the
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// incoming signal by clamping the amplitude to [-1, 1] and
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// performing a linear interpolation of the neighbor values.
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float index = std::max(0.0f, std::min(float(mCurve.Length() - 1),
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mCurve.Length() * (inputBuffer[j] + 1) / 2));
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uint32_t indexLower = uint32_t(index);
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uint32_t indexHigher = uint32_t(index + 1.0f);
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if (indexHigher == mCurve.Length()) {
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outputBuffer[j] = mCurve[indexLower];
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} else {
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float interpolationFactor = index - indexLower;
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outputBuffer[j] = (1.0f - interpolationFactor) * mCurve[indexLower] +
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interpolationFactor * mCurve[indexHigher];
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}
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}
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}
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}
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private:
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nsTArray<float> mCurve;
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OverSampleType mType;
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Resampler mResampler;
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};
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WaveShaperNode::WaveShaperNode(AudioContext* aContext)
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@ -256,7 +95,6 @@ WaveShaperNode::WaveShaperNode(AudioContext* aContext)
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ChannelCountMode::Max,
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ChannelInterpretation::Speakers)
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, mCurve(nullptr)
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, mType(OverSampleType::None)
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{
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NS_HOLD_JS_OBJECTS(this, WaveShaperNode);
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@ -300,12 +138,5 @@ WaveShaperNode::SetCurve(const Nullable<Float32Array>& aCurve)
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ns->SetRawArrayData(curve);
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}
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void
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WaveShaperNode::SetOversample(OverSampleType aType)
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{
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mType = aType;
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SendInt32ParameterToStream(WaveShaperNodeEngine::TYPE, static_cast<int32_t>(aType));
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}
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}
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}
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@ -9,7 +9,6 @@
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#include "AudioNode.h"
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#include "AudioParam.h"
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#include "mozilla/dom/WaveShaperNodeBinding.h"
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namespace mozilla {
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namespace dom {
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@ -34,12 +33,6 @@ public:
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}
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void SetCurve(const Nullable<Float32Array>& aData);
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OverSampleType Oversample() const
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{
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return mType;
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}
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void SetOversample(OverSampleType aType);
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private:
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void ClearCurve();
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@ -10,17 +10,10 @@
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* liability, trademark and document use rules apply.
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*/
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enum OverSampleType {
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"none",
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"2x",
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"4x"
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};
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[PrefControlled]
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interface WaveShaperNode : AudioNode {
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attribute Float32Array? curve;
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attribute OverSampleType oversample;
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};
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