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Several CPU usage optimizations

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This commit is contained in:
Jack Andersen
2016-05-21 22:35:55 -10:00
parent fbafa397fe
commit 654eccf82d
10 changed files with 65623 additions and 21 deletions
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57
lib/Voice.cpp
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@@ -11,6 +11,7 @@
namespace amuse
{
extern "C" const float VolumeLUT[];
void Voice::_destroy()
{
@@ -201,14 +202,41 @@ std::list<std::shared_ptr<Voice>>::iterator Voice::_destroyVoice(Voice* voice)
static void ApplyVolume(float vol, int16_t& samp)
{
/* -10dB to 0dB mapped to full volume range */
float factor = std::sqrt(std::pow(10.f, (vol - 1.f))) - 0.317f;
if (factor < 0.f) factor = 0.f;
samp *= factor;
samp *= VolumeLUT[int(vol * 65536)];
}
bool Voice::_advanceSample(int16_t& samp, int32_t& newPitch)
{
double dt = 1.0 / m_sampleRate;
double dt;
/* Block linearized will use a larger `dt` for amplitude sampling;
* significantly reducing the processing expense */
switch (m_engine.m_ampMode)
{
case AmplitudeMode::PerSample:
m_voiceSamples += 1;
dt = 1.0 / m_sampleRate;
break;
case AmplitudeMode::BlockLinearized:
{
uint32_t rem = m_voiceSamples % 160;
m_voiceSamples += 1;
if (rem != 0)
{
/* Lerp within 160-sample block */
float t = rem / 160.f;
float l = m_lastLevel * (1.f - t) + m_nextLevel * t;
/* Apply total volume to sample using decibel scale */
ApplyVolume(l, samp);
return false;
}
dt = 160.0 / m_sampleRate;
break;
}
}
m_voiceTime += dt;
bool refresh = false;
@@ -229,8 +257,9 @@ bool Voice::_advanceSample(int16_t& samp, int32_t& newPitch)
}
/* Factor in ADSR envelope state */
float adsr = m_volAdsr.nextSample(m_sampleRate);
float totalVol = m_userVol * m_curVol * adsr * (m_state.m_curVel / 127.f);
float adsr = m_volAdsr.advance(dt);
m_lastLevel = m_nextLevel;
m_nextLevel = m_userVol * m_curVol * adsr * (m_state.m_curVel / 127.f);
/* Apply tremolo */
if (m_state.m_tremoloSel && (m_tremoloScale || m_tremoloModScale))
@@ -241,23 +270,25 @@ bool Voice::_advanceSample(int16_t& samp, int32_t& newPitch)
float fac = (1.0f - t) + (m_tremoloScale * t);
float modT = getModWheel() / 127.f;
float modFac = (1.0f - modT) + (m_tremoloModScale * modT);
totalVol *= fac * modFac;
m_nextLevel *= fac * modFac;
}
else if (m_tremoloScale)
{
float fac = (1.0f - t) + (m_tremoloScale * t);
totalVol *= fac;
m_nextLevel *= fac;
}
else if (m_tremoloModScale)
{
float modT = getModWheel() / 127.f;
float modFac = (1.0f - modT) + (m_tremoloModScale * modT);
totalVol *= modFac;
m_nextLevel *= modFac;
}
}
/* Apple total volume to sample using decibel scale */
ApplyVolume(ClampFull<float>(totalVol), samp);
m_nextLevel = ClampFull<float>(m_nextLevel);
/* Apply total volume to sample using decibel scale */
ApplyVolume(m_nextLevel, samp);
/* Process active pan-sweep */
if (m_panningTime >= 0.f)
@@ -295,7 +326,7 @@ bool Voice::_advanceSample(int16_t& samp, int32_t& newPitch)
m_pitchDirty = false;
if (m_pitchEnv)
{
newPitch = m_curPitch * m_pitchAdsr.nextSample(m_sampleRate);
newPitch = m_curPitch * m_pitchAdsr.advance(dt);
refresh = true;
}
@@ -452,7 +483,9 @@ size_t Voice::supplyAudio(size_t samples, int16_t* data)
m_messageTrap.macroId == 0xffff &&
(!m_curSample || (m_curSample && m_volAdsr.isComplete())))
{
m_curSample = nullptr;
m_voxState = VoiceState::Dead;
m_backendVoice->stop();
}
return samples;
}