mirror of https://github.com/AxioDL/amuse.git
Several CPU usage optimizations
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fbafa397fe
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@ -12,6 +12,7 @@ set(SOURCES
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lib/SoundMacroState.cpp
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lib/SongState.cpp
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lib/Voice.cpp
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lib/VolumeLUT.cpp
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lib/Submix.cpp
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lib/EffectBase.cpp
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lib/EffectReverb.cpp
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@ -778,7 +778,7 @@ struct AppCallback : boo::IApplicationCallback
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/* Build voice engine */
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std::unique_ptr<boo::IAudioVoiceEngine> voxEngine = boo::NewAudioVoiceEngine();
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amuse::BooBackendVoiceAllocator booBackend(*voxEngine);
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m_engine.emplace(booBackend);
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m_engine.emplace(booBackend, amuse::AmplitudeMode::BlockLinearized);
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/* Load group into engine */
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const amuse::AudioGroup* group = m_engine->addAudioGroup(data);
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@ -20,6 +20,12 @@ class AudioGroup;
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class AudioGroupData;
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class IMIDIReader;
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enum class AmplitudeMode
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{
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PerSample, /**< Per-sample amplitude evaluation (dt = 1.0 / sampleRate, rather CPU demanding) */
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BlockLinearized /**< Per-block lerp amplitude evaluation (dt = 160.0 / sampleRate) */
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};
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/** Main audio playback system for a single audio output */
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class Engine
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{
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@ -29,6 +35,7 @@ class Engine
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friend struct Sequencer::ChannelState;
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IBackendVoiceAllocator& m_backend;
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AmplitudeMode m_ampMode;
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std::unique_ptr<IMIDIReader> m_midiReader;
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std::unordered_map<const AudioGroupData*, std::unique_ptr<AudioGroup>> m_audioGroups;
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std::list<std::shared_ptr<Voice>> m_activeVoices;
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@ -55,7 +62,7 @@ class Engine
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void _5MsCallback(double dt);
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public:
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~Engine();
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Engine(IBackendVoiceAllocator& backend);
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Engine(IBackendVoiceAllocator& backend, AmplitudeMode ampMode=AmplitudeMode::PerSample);
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/** Access voice backend of engine */
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IBackendVoiceAllocator& getBackend() {return m_backend;}
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@ -30,7 +30,7 @@ public:
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void reset(const ADSR* adsr);
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void reset(const ADSRDLS* adsr, int8_t note, int8_t vel);
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void keyOff();
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float nextSample(double sampleRate);
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float advance(double dt);
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bool isComplete() const {return m_phase == State::Complete;}
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};
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@ -61,6 +61,9 @@ class Voice : public Entity
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int16_t m_prev2 = 0; /**< DSPADPCM prev-prev sample */
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double m_sampleRate = 32000.0; /**< Current sample rate computed from relative sample key or SETPITCH */
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double m_voiceTime = 0.0; /**< Current seconds of voice playback (per-sample resolution) */
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uint32_t m_voiceSamples = 0; /**< Count of samples processed over voice's lifetime */
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float m_lastLevel = 0.f; /**< Last computed level ([0,1] mapped to [-10,0] clamped decibels) */
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float m_nextLevel = 0.f; /**< Next computed level used for lerp-mode amplitude */
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VoiceState m_voxState = VoiceState::Dead; /**< Current high-level state of voice */
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bool m_sustained = false; /**< Sustain pedal pressed for this voice */
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@ -23,8 +23,8 @@ Engine::~Engine()
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vox->_destroy();
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}
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Engine::Engine(IBackendVoiceAllocator& backend)
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: m_backend(backend)
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Engine::Engine(IBackendVoiceAllocator& backend, AmplitudeMode ampMode)
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: m_backend(backend), m_ampMode(ampMode)
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{
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backend.register5MsCallback(std::bind(&Engine::_5MsCallback, this, std::placeholders::_1));
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m_midiReader = backend.allocateMIDIReader(*this);
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@ -31,9 +31,9 @@ void Envelope::keyOff()
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m_curTime = 0.0;
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}
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float Envelope::nextSample(double sampleRate)
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float Envelope::advance(double dt)
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{
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m_curTime += 1.0 / sampleRate;
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m_curTime += dt;
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switch (m_phase)
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{
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@ -211,10 +211,16 @@ void Sequencer::ChannelState::setCtrlValue(uint8_t ctrl, int8_t val)
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for (const auto& vox : m_keyoffVoxs)
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vox->_notifyCtrlChange(ctrl, val);
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if (ctrl == 7)
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switch (ctrl)
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{
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case 7:
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setVolume(val / 127.f);
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else if (ctrl == 8 || ctrl == 10)
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break;
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case 10:
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setPan(val / 64.f - 1.f);
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break;
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default: break;
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}
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}
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bool Sequencer::ChannelState::programChange(int8_t prog)
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@ -11,6 +11,7 @@
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namespace amuse
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{
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extern "C" const float VolumeLUT[];
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void Voice::_destroy()
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{
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@ -201,14 +202,41 @@ std::list<std::shared_ptr<Voice>>::iterator Voice::_destroyVoice(Voice* voice)
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static void ApplyVolume(float vol, int16_t& samp)
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{
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/* -10dB to 0dB mapped to full volume range */
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float factor = std::sqrt(std::pow(10.f, (vol - 1.f))) - 0.317f;
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if (factor < 0.f) factor = 0.f;
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samp *= factor;
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samp *= VolumeLUT[int(vol * 65536)];
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}
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bool Voice::_advanceSample(int16_t& samp, int32_t& newPitch)
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{
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double dt = 1.0 / m_sampleRate;
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double dt;
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/* Block linearized will use a larger `dt` for amplitude sampling;
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* significantly reducing the processing expense */
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switch (m_engine.m_ampMode)
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{
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case AmplitudeMode::PerSample:
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m_voiceSamples += 1;
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dt = 1.0 / m_sampleRate;
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break;
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case AmplitudeMode::BlockLinearized:
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{
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uint32_t rem = m_voiceSamples % 160;
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m_voiceSamples += 1;
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if (rem != 0)
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{
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/* Lerp within 160-sample block */
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float t = rem / 160.f;
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float l = m_lastLevel * (1.f - t) + m_nextLevel * t;
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/* Apply total volume to sample using decibel scale */
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ApplyVolume(l, samp);
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return false;
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}
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dt = 160.0 / m_sampleRate;
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break;
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}
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}
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m_voiceTime += dt;
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bool refresh = false;
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@ -229,8 +257,9 @@ bool Voice::_advanceSample(int16_t& samp, int32_t& newPitch)
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}
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/* Factor in ADSR envelope state */
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float adsr = m_volAdsr.nextSample(m_sampleRate);
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float totalVol = m_userVol * m_curVol * adsr * (m_state.m_curVel / 127.f);
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float adsr = m_volAdsr.advance(dt);
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m_lastLevel = m_nextLevel;
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m_nextLevel = m_userVol * m_curVol * adsr * (m_state.m_curVel / 127.f);
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/* Apply tremolo */
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if (m_state.m_tremoloSel && (m_tremoloScale || m_tremoloModScale))
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@ -241,23 +270,25 @@ bool Voice::_advanceSample(int16_t& samp, int32_t& newPitch)
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float fac = (1.0f - t) + (m_tremoloScale * t);
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float modT = getModWheel() / 127.f;
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float modFac = (1.0f - modT) + (m_tremoloModScale * modT);
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totalVol *= fac * modFac;
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m_nextLevel *= fac * modFac;
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}
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else if (m_tremoloScale)
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{
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float fac = (1.0f - t) + (m_tremoloScale * t);
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totalVol *= fac;
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m_nextLevel *= fac;
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}
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else if (m_tremoloModScale)
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{
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float modT = getModWheel() / 127.f;
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float modFac = (1.0f - modT) + (m_tremoloModScale * modT);
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totalVol *= modFac;
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m_nextLevel *= modFac;
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}
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}
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/* Apple total volume to sample using decibel scale */
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ApplyVolume(ClampFull<float>(totalVol), samp);
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m_nextLevel = ClampFull<float>(m_nextLevel);
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/* Apply total volume to sample using decibel scale */
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ApplyVolume(m_nextLevel, samp);
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/* Process active pan-sweep */
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if (m_panningTime >= 0.f)
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@ -295,7 +326,7 @@ bool Voice::_advanceSample(int16_t& samp, int32_t& newPitch)
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m_pitchDirty = false;
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if (m_pitchEnv)
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{
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newPitch = m_curPitch * m_pitchAdsr.nextSample(m_sampleRate);
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newPitch = m_curPitch * m_pitchAdsr.advance(dt);
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refresh = true;
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}
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@ -452,7 +483,9 @@ size_t Voice::supplyAudio(size_t samples, int16_t* data)
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m_messageTrap.macroId == 0xffff &&
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(!m_curSample || (m_curSample && m_volAdsr.isComplete())))
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{
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m_curSample = nullptr;
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m_voxState = VoiceState::Dead;
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m_backendVoice->stop();
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}
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return samples;
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}
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