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SPU2: Cleanup SndOut.cpp 

const, casts, constexpr, initializations.
This commit is contained in:
lightningterror 2023-01-15 01:30:54 +01:00
parent 2740785df5
commit 018692edd0
1 changed files with 32 additions and 32 deletions
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64
pcsx2/SPU2/SndOut.cpp
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@ -1,5 +1,5 @@
/* PCSX2 - PS2 Emulator for PCs
* Copyright (C) 2002-2020 PCSX2 Dev Team
* Copyright (C) 2002-2023 PCSX2 Dev Team
*
* PCSX2 is free software: you can redistribute it and/or modify it under the terms
* of the GNU Lesser General Public License as published by the Free Software Found-
@ -32,8 +32,8 @@ StereoOut32::StereoOut32(const StereoOut16& src)
}
StereoOut32::StereoOut32(const StereoOutFloat& src)
: Left((s32)(src.Left * 2147483647.0f))
, Right((s32)(src.Right * 2147483647.0f))
: Left(static_cast<s32>(src.Left * 2147483647.0f))
, Right(static_cast<s32>(src.Right * 2147483647.0f))
{
}
@ -221,8 +221,8 @@ void SndBuffer::_WriteSamples_Safe(StereoOut32* bData, int nSamples)
// WARNING: This code assumes there's only ONE writing process.
if ((m_size - m_wpos) < nSamples)
{
int b1 = m_size - m_wpos;
int b2 = nSamples - b1;
const int b1 = m_size - m_wpos;
const int b2 = nSamples - b1;
_WriteSamples_Internal(bData, b1);
_WriteSamples_Internal(bData + b1, b2);
@ -238,8 +238,8 @@ void SndBuffer::_ReadSamples_Safe(StereoOut32* bData, int nSamples)
// WARNING: This code assumes there's only ONE reading process.
if ((m_size - m_rpos) < nSamples)
{
int b1 = m_size - m_rpos;
int b2 = nSamples - b1;
const int b1 = m_size - m_rpos;
const int b2 = nSamples - b1;
_ReadSamples_Internal(bData, b1);
_ReadSamples_Internal(bData + b1, b2);
@ -270,7 +270,7 @@ void SndBuffer::ReadSamples(T* bData, int nSamples)
// This will cause one brief hiccup that can never exceed the user's
// set buffer length in duration.
int quietSamples;
int quietSamples = 0;
if (CheckUnderrunStatus(nSamples, quietSamples))
{
pxAssume(nSamples <= SndOutPacketSize);
@ -286,7 +286,7 @@ void SndBuffer::ReadSamples(T* bData, int nSamples)
bData[i].ResampleFrom(m_buffer[i + m_rpos]);
// Second part
int b2 = nSamples - b1;
const int b2 = nSamples - b1;
for (int i = 0; i < b2; i++)
bData[i + b1].ResampleFrom(m_buffer[i]);
@ -336,7 +336,7 @@ void SndBuffer::_WriteSamples(StereoOut32* bData, int nSamples)
// The older portion of the buffer is discarded rather than incoming data,
// so that the overall audio synchronization is better.
int free = m_size - _GetApproximateDataInBuffer(); // -1, but the <= handles that
const int free = m_size - _GetApproximateDataInBuffer(); // -1, but the <= handles that
if (free <= nSamples)
{
// Disabled since the lock-free queue can't handle changing the read end from the write thread
@ -345,7 +345,7 @@ void SndBuffer::_WriteSamples(StereoOut32* bData, int nSamples)
// Dump samples from the read portion of the buffer instead of dropping
// the newly written stuff.
s32 comp;
s32 comp = 0;
if( SynchMode == 0 ) // TimeStrech on
{
@ -513,12 +513,12 @@ float SndBuffer::GetStatusPct()
// Get the buffer status of the output driver too, so that we can
// obtain a more accurate overall buffer status.
int drvempty = s_output_module->GetEmptySampleCount(); // / 2;
const int drvempty = s_output_module->GetEmptySampleCount(); // / 2;
//ConLog( "Data %d >>> driver: %d predict: %d\n", m_data, drvempty, m_predictData );
int data = _GetApproximateDataInBuffer();
float result = (float)(data + m_predictData - drvempty) - (m_size / 16);
const int data = _GetApproximateDataInBuffer();
float result = static_cast<float>(data + m_predictData - drvempty) - (m_size / 16);
result /= (m_size / 16);
return result;
}
@ -584,8 +584,8 @@ static float addToAvg(float val)
avg_fullness[nextAvgPos] = val;
nextAvgPos = (nextAvgPos + 1U) % AVERAGING_BUFFER_SIZE;
unsigned int actualWindow = std::min(available, AVERAGING_WINDOW);
unsigned int first = (nextAvgPos - actualWindow + AVERAGING_BUFFER_SIZE) % AVERAGING_BUFFER_SIZE;
const unsigned int actualWindow = std::min(available, AVERAGING_WINDOW);
const unsigned int first = (nextAvgPos - actualWindow + AVERAGING_BUFFER_SIZE) % AVERAGING_BUFFER_SIZE;
// Possible optimization: if we know that actualWindow hasn't changed since
// last invocation, we could calculate the running average in O(1) instead of O(N)
@ -613,9 +613,9 @@ static bool IsInRange(const T& val, const T& min, const T& max)
void SndBuffer::UpdateTempoChangeSoundTouch2()
{
long targetSamplesReservoir = 48 * EmuConfig.SPU2.Latency; //48000*SndOutLatencyMS/1000
const long targetSamplesReservoir = 48 * EmuConfig.SPU2.Latency; //48000*SndOutLatencyMS/1000
//base aim at buffer filled %
float baseTargetFullness = (double)targetSamplesReservoir; ///(double)m_size;//0.05;
float baseTargetFullness = static_cast<double>(targetSamplesReservoir); ///(double)m_size;//0.05;
//state vars
static bool inside_hysteresis; //=false;
@ -630,8 +630,8 @@ void SndBuffer::UpdateTempoChangeSoundTouch2()
dynamicTargetFullness = baseTargetFullness;
}
int data = _GetApproximateDataInBuffer();
float bufferFullness = (float)data; ///(float)m_size;
const int data = _GetApproximateDataInBuffer();
const float bufferFullness = static_cast<float>(data); ///(float)m_size;
#ifdef NEWSTRETCHER_USE_DYNAMIC_TUNING
{ //test current iterations/sec every 0.5s, and change algo params accordingly if different than previous IPS more than 30%
@ -657,13 +657,13 @@ void SndBuffer::UpdateTempoChangeSoundTouch2()
#endif
//Algorithm params: (threshold params (hysteresis), etc)
const float hys_ok_factor = 1.04f;
const float hys_bad_factor = 1.2f;
int hys_min_ok_count = std::clamp((int)(50.0 * (float)targetIPS / 750.0), 2, 100); //consecutive iterations within hys_ok before going to 1:1 mode
int compensationDivider = std::clamp((int)(100.0 * (float)targetIPS / 750), 15, 150);
constexpr float hys_ok_factor = 1.04f;
constexpr float hys_bad_factor = 1.2f;
const int hys_min_ok_count = std::clamp((int)(50.0 * (float)targetIPS / 750.0), 2, 100); //consecutive iterations within hys_ok before going to 1:1 mode
const int compensationDivider = std::clamp((int)(100.0 * (float)targetIPS / 750), 15, 150);
float tempoAdjust = bufferFullness / dynamicTargetFullness;
float avgerage = addToAvg(tempoAdjust);
const float avgerage = addToAvg(tempoAdjust);
tempoAdjust = avgerage;
// Dampen the adjustment to avoid overshoots (this means the average will compensate to the other side).
@ -733,8 +733,8 @@ void SndBuffer::UpdateTempoChangeSoundTouch2()
void SndBuffer::UpdateTempoChangeSoundTouch()
{
float statusPct = GetStatusPct();
float pctChange = statusPct - lastPct;
const float statusPct = GetStatusPct();
const float pctChange = statusPct - lastPct;
float tempoChange;
float emergencyAdj = 0;
@ -783,8 +783,8 @@ void SndBuffer::UpdateTempoChangeSoundTouch()
// Prediction based on the buffer fill status:
const float statusWeight = 2.99f;
const float statusRange = 0.068f;
constexpr float statusWeight = 2.99f;
constexpr float statusRange = 0.068f;
// "non-emergency" deadzone: In this area stretching will be strongly discouraged.
// Note: due tot he nature of timestretch latency, it's always a wee bit harder to
@ -830,7 +830,7 @@ void SndBuffer::UpdateTempoChangeSoundTouch()
// is usually better/safer)
if (newTempo < 0.970f || newTempo > 1.045f)
{
cTempo = (float)newcee;
cTempo = static_cast<float>(newcee);
if (newTempo < 0.10f)
newTempo = 0.10f;
@ -842,7 +842,7 @@ void SndBuffer::UpdateTempoChangeSoundTouch()
else if (cTempo > 7.5f)
cTempo = 7.5f;
pSoundTouch->setTempo(eTempo = (float)newTempo);
pSoundTouch->setTempo(eTempo = static_cast<float>(newTempo));
/*ConLog("* SPU2: [Nominal %d%%] [Emergency: %d%%] (baseTempo: %d%% ) (newTempo: %d%%) (buffer: %d%%)\n",
//(relation < 0.0) ? "Normalize" : "",
@ -875,7 +875,7 @@ void SndBuffer::UpdateTempoChangeSoundTouch()
extern uint TickInterval;
void SndBuffer::UpdateTempoChangeAsyncMixing()
{
float statusPct = GetStatusPct();
const float statusPct = GetStatusPct();
lastPct = statusPct;
if (statusPct < -0.1f)