mirror of https://github.com/AxioDL/amuse.git
Implement EffectDelay
This commit is contained in:
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02d6478ce3
commit
4353113649
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@ -12,8 +12,7 @@ template <typename T>
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class EffectBase
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{
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public:
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virtual void applyEffect(T* audio, size_t frameCount,
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const ChannelMap& chanMap, double sampleRate)=0;
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virtual void applyEffect(T* audio, size_t frameCount, const ChannelMap& chanMap)=0;
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};
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}
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@ -59,7 +59,7 @@ class EffectChorus : public EffectBase<T>
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uint32_t x94_variation; /**< [0, 5] time error (in ms) to set delay within */
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uint32_t x98_period; /**< [500, 10000] time (in ms) of one delay-shift cycle */
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double m_sampsPerMs;
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uint32_t m_sampsPerMs;
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uint32_t m_blockSamples;
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bool m_dirty = true;
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@ -68,8 +68,7 @@ class EffectChorus : public EffectBase<T>
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public:
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~EffectChorus();
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EffectChorus(uint32_t baseDelay, uint32_t variation, uint32_t period, double sampleRate);
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void applyEffect(T* audio, size_t frameCount,
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const ChannelMap& chanMap, double sampleRate);
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void applyEffect(T* audio, size_t frameCount, const ChannelMap& chanMap);
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};
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}
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@ -3,24 +3,33 @@
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#include "EffectBase.hpp"
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#include <stdint.h>
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#include <memory>
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namespace amuse
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{
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/** Mixes the audio back into itself after specified delay */
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class EffectDelay : public EffectBase
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template <typename T>
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class EffectDelay : public EffectBase<T>
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{
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uint32_t m_delay[8]; /**< [10, 5000] time in ms of each channel's delay */
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uint32_t m_feedback[8]; /**< [0, 100] percent to mix delayed signal with input signal */
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uint32_t m_output[8]; /**< [0, 100] total output percent */
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uint32_t x0_currentSize[8];
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uint32_t xc_currentPos[8];
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uint32_t x18_currentFeedback[8];
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uint32_t x24_currentOutput[8];
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std::unique_ptr<T[]> x30_chanLines[8];
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uint32_t x3c_delay[8]; /**< [10, 5000] time in ms of each channel's delay */
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uint32_t x48_feedback[8]; /**< [0, 100] percent to mix delayed signal with input signal */
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uint32_t x54_output[8]; /**< [0, 100] total output percent */
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uint32_t m_sampsPerMs;
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uint32_t m_blockSamples;
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bool m_dirty = true;
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void _update();
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public:
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EffectDelay(uint32_t initDelay, uint32_t initFeedback, uint32_t initOutput);
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void applyEffect(int16_t* audio, size_t frameCount,
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const ChannelMap& chanMap, double sampleRate);
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void applyEffect(int32_t* audio, size_t frameCount,
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const ChannelMap& chanMap, double sampleRate);
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void applyEffect(float* audio, size_t frameCount,
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const ChannelMap& chanMap, double sampleRate);
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EffectDelay(uint32_t initDelay, uint32_t initFeedback, uint32_t initOutput, double sampleRate);
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void applyEffect(T* audio, size_t frameCount, const ChannelMap& chanMap);
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};
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}
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@ -145,7 +145,7 @@ EffectChorus<T>::EffectChorus(uint32_t baseDelay, uint32_t variation,
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x94_variation(clamp(0u, variation, 5u)),
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x98_period(clamp(500u, period, 10000u)),
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m_sampsPerMs(sampleRate / 1000.0),
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m_blockSamples(sampleRate * 160.0 / 32000.0)
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m_blockSamples(m_sampsPerMs * 160 / 32)
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{
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T* buf = new T[m_blockSamples * AMUSE_CHORUS_NUM_BLOCKS * 8];
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memset(buf, 0, m_blockSamples * AMUSE_CHORUS_NUM_BLOCKS * 8 * sizeof(T));
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@ -303,133 +303,140 @@ void EffectChorus<T>::SrcInfo::doSrc2(size_t blockSamples, size_t chanCount)
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}
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template <typename T>
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void EffectChorus<T>::applyEffect(T* audio, size_t frameCount,
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const ChannelMap& chanMap, double sampleRate)
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void EffectChorus<T>::applyEffect(T* audio, size_t frameCount, const ChannelMap& chanMap)
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{
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if (m_dirty)
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_update();
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uint8_t next = x24_currentLast + 1;
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uint8_t buf = next % 3;
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T* leftBuf = x0_lastLeft[buf];
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T* rightBuf = xc_lastRight[buf];
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T* rearLeftBuf = x18_lastRearLeft[buf];
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T* rearRightBuf = x18_lastRearRight[buf];
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T* centerBuf = x18_lastCenter[buf];
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T* lfeBuf = x18_lastLFE[buf];
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T* sideLeftBuf = x18_lastSideLeft[buf];
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T* sideRightBuf = x18_lastSideRight[buf];
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T* inBuf = audio;
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for (size_t f=0 ; f<frameCount && f<m_blockSamples ; ++f)
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size_t remFrames = frameCount;
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for (size_t f=0 ; f<frameCount ;)
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{
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uint8_t next = x24_currentLast + 1;
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uint8_t buf = next % 3;
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T* leftBuf = x0_lastLeft[buf];
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T* rightBuf = xc_lastRight[buf];
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T* rearLeftBuf = x18_lastRearLeft[buf];
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T* rearRightBuf = x18_lastRearRight[buf];
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T* centerBuf = x18_lastCenter[buf];
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T* lfeBuf = x18_lastLFE[buf];
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T* sideLeftBuf = x18_lastSideLeft[buf];
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T* sideRightBuf = x18_lastSideRight[buf];
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T* inBuf = audio;
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for (size_t s=0 ; f<frameCount && s<m_blockSamples ; ++s, ++f)
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{
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for (size_t c=0 ; c<chanMap.m_channelCount ; ++c)
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{
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switch (chanMap.m_channels[c])
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{
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case AudioChannel::FrontLeft:
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*leftBuf++ = *inBuf++;
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break;
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case AudioChannel::FrontRight:
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*rightBuf++ = *inBuf++;
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break;
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case AudioChannel::RearLeft:
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*rearLeftBuf++ = *inBuf++;
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break;
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case AudioChannel::RearRight:
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*rearRightBuf++ = *inBuf++;
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break;
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case AudioChannel::FrontCenter:
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*centerBuf++ = *inBuf++;
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break;
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case AudioChannel::LFE:
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*lfeBuf++ = *inBuf++;
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break;
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case AudioChannel::SideLeft:
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*sideLeftBuf++ = *inBuf++;
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break;
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case AudioChannel::SideRight:
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*sideRightBuf++ = *inBuf++;
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break;
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default:
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inBuf++;
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break;
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}
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}
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}
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x6c_src.x84_pitchHi = (x60_pitchOffset >> 16) + 1;
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x6c_src.x80_pitchLo = (x60_pitchOffset << 16);
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--x64_pitchOffsetPeriodCount;
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if (x64_pitchOffsetPeriodCount == 0)
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{
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x64_pitchOffsetPeriodCount = x68_pitchOffsetPeriod;
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x60_pitchOffset = -x60_pitchOffset;
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}
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T* outBuf = audio;
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size_t bs = std::min(remFrames, size_t(m_blockSamples));
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for (size_t c=0 ; c<chanMap.m_channelCount ; ++c)
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{
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x6c_src.x7c_posHi = x5c_currentPosHi;
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x6c_src.x78_posLo = x58_currentPosLo;
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x6c_src.x6c_dest = outBuf++;
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switch (chanMap.m_channels[c])
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{
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case AudioChannel::FrontLeft:
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*leftBuf++ = *inBuf++;
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x6c_src.x70_smpBase = x0_lastLeft[0];
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x6c_src.x74_old = x28_oldLeft;
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break;
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case AudioChannel::FrontRight:
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*rightBuf++ = *inBuf++;
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x6c_src.x70_smpBase = xc_lastRight[0];
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x6c_src.x74_old = x38_oldRight;
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break;
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case AudioChannel::RearLeft:
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*rearLeftBuf++ = *inBuf++;
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x6c_src.x70_smpBase = x18_lastRearLeft[0];
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x6c_src.x74_old = x48_oldRearLeft;
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break;
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case AudioChannel::RearRight:
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*rearRightBuf++ = *inBuf++;
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x6c_src.x70_smpBase = x18_lastRearRight[0];
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x6c_src.x74_old = x48_oldRearRight;
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break;
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case AudioChannel::FrontCenter:
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*centerBuf++ = *inBuf++;
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x6c_src.x70_smpBase = x18_lastCenter[0];
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x6c_src.x74_old = x48_oldCenter;
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break;
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case AudioChannel::LFE:
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*lfeBuf++ = *inBuf++;
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x6c_src.x70_smpBase = x18_lastLFE[0];
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x6c_src.x74_old = x48_oldLFE;
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break;
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case AudioChannel::SideLeft:
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*sideLeftBuf++ = *inBuf++;
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x6c_src.x70_smpBase = x18_lastSideLeft[0];
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x6c_src.x74_old = x48_oldSideLeft;
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break;
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case AudioChannel::SideRight:
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*sideRightBuf++ = *inBuf++;
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x6c_src.x70_smpBase = x18_lastSideRight[0];
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x6c_src.x74_old = x48_oldSideRight;
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break;
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default:
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inBuf++;
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default: break;
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}
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switch (x6c_src.x84_pitchHi)
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{
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case 0:
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x6c_src.doSrc1(bs, chanMap.m_channelCount);
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break;
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case 1:
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x6c_src.doSrc2(bs, chanMap.m_channelCount);
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break;
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default: break;
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}
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}
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audio = outBuf;
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remFrames -= bs;
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size_t chanPitch = m_blockSamples * AMUSE_CHORUS_NUM_BLOCKS;
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size_t fifteenSamps = 15 * m_sampsPerMs;
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x5c_currentPosHi = x6c_src.x7c_posHi % (chanPitch / fifteenSamps * fifteenSamps);
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x58_currentPosLo = x6c_src.x78_posLo;
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x24_currentLast = buf;
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}
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x6c_src.x84_pitchHi = (x60_pitchOffset >> 16) + 1;
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x6c_src.x80_pitchLo = (x60_pitchOffset << 16);
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--x64_pitchOffsetPeriodCount;
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if (x64_pitchOffsetPeriodCount == 0)
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{
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x64_pitchOffsetPeriodCount = x68_pitchOffsetPeriod;
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x60_pitchOffset = -x60_pitchOffset;
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}
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T* outBuf = audio;
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for (size_t c=0 ; c<chanMap.m_channelCount ; ++c)
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{
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x6c_src.x7c_posHi = x5c_currentPosHi;
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x6c_src.x78_posLo = x58_currentPosLo;
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x6c_src.x6c_dest = outBuf++;
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switch (chanMap.m_channels[c])
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{
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case AudioChannel::FrontLeft:
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x6c_src.x70_smpBase = x0_lastLeft[0];
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x6c_src.x74_old = x28_oldLeft;
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break;
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case AudioChannel::FrontRight:
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x6c_src.x70_smpBase = xc_lastRight[0];
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x6c_src.x74_old = x38_oldRight;
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break;
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case AudioChannel::RearLeft:
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x6c_src.x70_smpBase = x18_lastRearLeft[0];
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x6c_src.x74_old = x48_oldRearLeft;
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break;
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case AudioChannel::RearRight:
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x6c_src.x70_smpBase = x18_lastRearRight[0];
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x6c_src.x74_old = x48_oldRearRight;
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break;
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case AudioChannel::FrontCenter:
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x6c_src.x70_smpBase = x18_lastCenter[0];
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x6c_src.x74_old = x48_oldCenter;
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break;
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case AudioChannel::LFE:
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x6c_src.x70_smpBase = x18_lastLFE[0];
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x6c_src.x74_old = x48_oldLFE;
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break;
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case AudioChannel::SideLeft:
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x6c_src.x70_smpBase = x18_lastSideLeft[0];
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x6c_src.x74_old = x48_oldSideLeft;
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break;
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case AudioChannel::SideRight:
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x6c_src.x70_smpBase = x18_lastSideRight[0];
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x6c_src.x74_old = x48_oldSideRight;
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break;
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default: break;
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}
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switch (x6c_src.x84_pitchHi)
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{
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case 0:
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x6c_src.doSrc1(m_blockSamples, chanMap.m_channelCount);
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break;
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case 1:
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x6c_src.doSrc2(m_blockSamples, chanMap.m_channelCount);
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break;
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default: break;
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}
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}
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size_t chanPitch = m_blockSamples * AMUSE_CHORUS_NUM_BLOCKS;
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size_t fifteenSamps = 15 * m_sampsPerMs;
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x5c_currentPosHi = x6c_src.x7c_posHi % (chanPitch / fifteenSamps * fifteenSamps);
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x58_currentPosLo = x6c_src.x78_posLo;
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x24_currentLast = buf;
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}
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template class EffectChorus<int16_t>;
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@ -0,0 +1,64 @@
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#include "amuse/EffectDelay.hpp"
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#include "amuse/Common.hpp"
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#include "amuse/IBackendVoice.hpp"
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#include <string.h>
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namespace amuse
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{
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template <typename T>
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EffectDelay<T>::EffectDelay(uint32_t initDelay, uint32_t initFeedback,
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uint32_t initOutput, double sampleRate)
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: m_sampsPerMs(sampleRate / 1000.0),
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m_blockSamples(m_sampsPerMs * 160 / 32)
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{
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initDelay = clamp(10u, initDelay, 5000u);
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initFeedback = clamp(0u, initFeedback, 100u);
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initOutput = clamp(0u, initOutput, 100u);
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for (int i=0 ; i<8 ; ++i)
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{
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x3c_delay[i] = initDelay;
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x48_feedback[i] = initFeedback;
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x54_output[i] = initOutput;
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}
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_update();
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}
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template <typename T>
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void EffectDelay<T>::_update()
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{
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for (int i=0 ; i<8 ; ++i)
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{
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x0_currentSize[i] = ((x3c_delay[i] - 5) * m_sampsPerMs + 159) / 160;
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xc_currentPos[i] = 0;
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x18_currentFeedback[i] = x48_feedback[i] * 128 / 100;
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x24_currentOutput[i] = x54_output[i] * 128 / 100;
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x30_chanLines[i].reset(new T[m_blockSamples * x0_currentSize[i]]);
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memset(x30_chanLines[i], 0, m_blockSamples * x0_currentSize[i] * sizeof(T));
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}
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m_dirty = false;
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}
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template <typename T>
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void EffectDelay<T>::applyEffect(T* audio, size_t frameCount, const ChannelMap& chanMap)
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{
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for (size_t f=0 ; f<frameCount ;)
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{
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for (int c=0 ; c<chanMap.m_channelCount ; ++c)
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{
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for (int i=0 ; i<m_blockSamples && f<frameCount ; ++i, ++f)
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{
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T& samp = x30_chanLines[c][xc_currentPos[c] * m_blockSamples + i];
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samp = samp * x18_currentFeedback[c] / 128 + *audio;
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*audio++ = samp * x24_currentOutput[c] / 128;
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}
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xc_currentPos = (xc_currentPos[c] + 1) % x0_currentSize[c];
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}
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}
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}
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}
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