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-rw-r--r--Alc/alc.cpp22
-rw-r--r--Alc/alconfig.cpp4
-rw-r--r--Alc/alu.cpp18
-rw-r--r--Alc/backends/alsa.cpp16
-rw-r--r--Alc/backends/portaudio.cpp2
-rw-r--r--Alc/backends/pulseaudio.cpp6
-rw-r--r--Alc/bformatdec.cpp2
-rw-r--r--Alc/converter.cpp8
-rw-r--r--Alc/effects/chorus.cpp6
-rw-r--r--Alc/effects/compressor.cpp4
-rw-r--r--Alc/effects/equalizer.cpp2
-rw-r--r--Alc/effects/fshifter.cpp4
-rw-r--r--Alc/effects/modulator.cpp16
-rw-r--r--Alc/effects/pshifter.cpp12
-rw-r--r--Alc/effects/reverb.cpp6
-rw-r--r--Alc/helpers.cpp2
-rw-r--r--Alc/hrtf.cpp10
-rw-r--r--Alc/mastering.cpp6
-rw-r--r--Alc/mixer/mixer_c.cpp4
-rw-r--r--Alc/mixer/mixer_sse.cpp18
-rw-r--r--Alc/mixvoice.cpp12
-rw-r--r--Alc/panning.cpp8
22 files changed, 94 insertions, 94 deletions
diff --git a/Alc/alc.cpp b/Alc/alc.cpp
index e44b42c8..9b56ea8d 100644
--- a/Alc/alc.cpp
+++ b/Alc/alc.cpp
@@ -985,7 +985,7 @@ static void alc_initconfig(void)
if(!str[0] || str[0] == ',')
continue;
- size_t len{next ? (size_t)(next-str) : strlen(str)};
+ size_t len{next ? static_cast<size_t>(next-str) : strlen(str)};
while(len > 0 && isspace(str[len-1]))
len--;
if(len == 3 && strncasecmp(str, "sse", len) == 0)
@@ -1062,7 +1062,7 @@ static void alc_initconfig(void)
}
endlist = 1;
- len = (next ? ((size_t)(next-devs)) : strlen(devs));
+ len = (next ? (static_cast<size_t>(next-devs)) : strlen(devs));
while(len > 0 && isspace(devs[len-1]))
len--;
#ifdef HAVE_WASAPI
@@ -1147,7 +1147,7 @@ static void alc_initconfig(void)
if(!str[0] || next == str)
continue;
- size_t len{next ? (size_t)(next-str) : strlen(str)};
+ size_t len{next ? static_cast<size_t>(next-str) : strlen(str)};
for(size_t n{0u};n < countof(gEffectList);n++)
{
if(len == strlen(gEffectList[n].name) &&
@@ -1858,7 +1858,7 @@ static ALCenum UpdateDeviceParams(ALCdevice *device, const ALCint *attrList)
device->HrtfList = EnumerateHrtf(device->DeviceName.c_str());
if(!device->HrtfList.empty())
{
- if(hrtf_id >= 0 && (size_t)hrtf_id < device->HrtfList.size())
+ if(hrtf_id >= 0 && static_cast<size_t>(hrtf_id) < device->HrtfList.size())
hrtf = GetLoadedHrtf(device->HrtfList[hrtf_id].hrtf);
else
hrtf = GetLoadedHrtf(device->HrtfList.front().hrtf);
@@ -1989,7 +1989,7 @@ static ALCenum UpdateDeviceParams(ALCdevice *device, const ALCint *attrList)
if(depth > 0)
{
depth = clampi(depth, 2, 24);
- device->DitherDepth = std::pow(2.0f, (ALfloat)(depth-1));
+ device->DitherDepth = std::pow(2.0f, static_cast<ALfloat>(depth-1));
}
}
if(!(device->DitherDepth > 0.0f))
@@ -2608,7 +2608,7 @@ void AllocateVoices(ALCcontext *context, ALsizei num_voices, ALsizei old_sends)
const size_t size{sizeof(ALvoice*) + sizeof_voice};
auto voices = static_cast<ALvoice**>(al_calloc(16, RoundUp(size*num_voices, 16)));
- auto voice = reinterpret_cast<ALvoice*>((char*)voices + RoundUp(num_voices*sizeof(ALvoice*), 16));
+ auto voice = reinterpret_cast<ALvoice*>(reinterpret_cast<char*>(voices) + RoundUp(num_voices*sizeof(ALvoice*), 16));
auto viter = voices;
if(context->Voices)
@@ -2670,7 +2670,7 @@ void AllocateVoices(ALCcontext *context, ALsizei num_voices, ALsizei old_sends)
/* Set this voice's reference. */
ALvoice *ret = voice;
/* Increment pointer to the next storage space. */
- voice = reinterpret_cast<ALvoice*>((char*)voice + sizeof_voice);
+ voice = reinterpret_cast<ALvoice*>(reinterpret_cast<char*>(voice) + sizeof_voice);
return ret;
};
viter = std::transform(context->Voices, context->Voices+v_count, viter, copy_voice);
@@ -2683,7 +2683,7 @@ void AllocateVoices(ALCcontext *context, ALsizei num_voices, ALsizei old_sends)
auto init_voice = [&voice,sizeof_voice]() -> ALvoice*
{
ALvoice *ret = new (voice) ALvoice{};
- voice = reinterpret_cast<ALvoice*>((char*)voice + sizeof_voice);
+ voice = reinterpret_cast<ALvoice*>(reinterpret_cast<char*>(voice) + sizeof_voice);
return ret;
};
std::generate(viter, voices+num_voices, init_voice);
@@ -3156,7 +3156,7 @@ static ALCsizei GetIntegerv(ALCdevice *device, ALCenum param, ALCsizei size, ALC
{ std::lock_guard<std::mutex> _{device->StateLock};
device->HrtfList.clear();
device->HrtfList = EnumerateHrtf(device->DeviceName.c_str());
- values[0] = (ALCint)device->HrtfList.size();
+ values[0] = static_cast<ALCint>(device->HrtfList.size());
}
return 1;
@@ -4022,7 +4022,7 @@ ALC_API void ALC_APIENTRY alcCaptureSamples(ALCdevice *device, ALCvoid *buffer,
ALCenum err{ALC_INVALID_VALUE};
{ std::lock_guard<std::mutex> _{dev->StateLock};
BackendBase *backend{dev->Backend.get()};
- if(samples >= 0 && backend->availableSamples() >= (ALCuint)samples)
+ if(samples >= 0 && backend->availableSamples() >= static_cast<ALCuint>(samples))
err = backend->captureSamples(buffer, samples);
}
if(err != ALC_NO_ERROR)
@@ -4210,7 +4210,7 @@ ALC_API const ALCchar* ALC_APIENTRY alcGetStringiSOFT(ALCdevice *device, ALCenum
else switch(paramName)
{
case ALC_HRTF_SPECIFIER_SOFT:
- if(index >= 0 && (size_t)index < dev->HrtfList.size())
+ if(index >= 0 && static_cast<size_t>(index) < dev->HrtfList.size())
return dev->HrtfList[index].name.c_str();
alcSetError(dev.get(), ALC_INVALID_VALUE);
break;
diff --git a/Alc/alconfig.cpp b/Alc/alconfig.cpp
index eecaf6fc..c4fde638 100644
--- a/Alc/alconfig.cpp
+++ b/Alc/alconfig.cpp
@@ -93,7 +93,7 @@ std:: string expdup(const char *str)
{
const char *next = std::strchr(str, '$');
addstr = str;
- addstrlen = next ? (size_t)(next-str) : std::strlen(str);
+ addstrlen = next ? static_cast<size_t>(next-str) : std::strlen(str);
str += addstrlen;
}
@@ -104,7 +104,7 @@ std:: string expdup(const char *str)
{
const char *next = std::strchr(str+1, '$');
addstr = str;
- addstrlen = next ? (size_t)(next-str) : std::strlen(str);
+ addstrlen = next ? static_cast<size_t>(next-str) : std::strlen(str);
str += addstrlen;
}
diff --git a/Alc/alu.cpp b/Alc/alu.cpp
index 77505a0c..42e31b88 100644
--- a/Alc/alu.cpp
+++ b/Alc/alu.cpp
@@ -227,7 +227,7 @@ void BsincPrepare(const ALuint increment, BsincState *state, const BSincTable *t
if(increment > FRACTIONONE)
{
- sf = (ALfloat)FRACTIONONE / increment;
+ sf = static_cast<ALfloat>FRACTIONONE / increment;
sf = maxf(0.0f, (BSINC_SCALE_COUNT-1) * (sf-table->scaleBase) * table->scaleRange);
si = float2int(sf);
/* The interpolation factor is fit to this diagonally-symmetric curve
@@ -620,7 +620,7 @@ void CalcPanningAndFilters(ALvoice *voice, const ALfloat Azi, const ALfloat Elev
const ALfloat mdist{maxf(Distance*Listener.Params.MetersPerUnit,
Device->AvgSpeakerDist/4.0f)};
const ALfloat w0{SPEEDOFSOUNDMETRESPERSEC /
- (mdist * (ALfloat)Device->Frequency)};
+ (mdist * static_cast<ALfloat>(Device->Frequency))};
/* Only need to adjust the first channel of a B-Format source. */
voice->Direct.Params[0].NFCtrlFilter.adjust(w0);
@@ -849,7 +849,7 @@ void CalcPanningAndFilters(ALvoice *voice, const ALfloat Azi, const ALfloat Elev
const ALfloat mdist{maxf(Distance*Listener.Params.MetersPerUnit,
Device->AvgSpeakerDist/4.0f)};
const ALfloat w0{SPEEDOFSOUNDMETRESPERSEC /
- (mdist * (ALfloat)Device->Frequency)};
+ (mdist * static_cast<ALfloat>(Device->Frequency))};
/* Adjust NFC filters. */
for(ALsizei c{0};c < num_channels;c++)
@@ -908,7 +908,7 @@ void CalcPanningAndFilters(ALvoice *voice, const ALfloat Azi, const ALfloat Elev
* source moves away from the listener.
*/
const ALfloat w0{SPEEDOFSOUNDMETRESPERSEC /
- (Device->AvgSpeakerDist * (ALfloat)Device->Frequency)};
+ (Device->AvgSpeakerDist * static_cast<ALfloat>(Device->Frequency))};
for(ALsizei c{0};c < num_channels;c++)
voice->Direct.Params[c].NFCtrlFilter.adjust(w0);
@@ -1026,7 +1026,7 @@ void CalcNonAttnSourceParams(ALvoice *voice, const ALvoicePropsBase *props, cons
/* Calculate the stepping value */
const auto Pitch = static_cast<ALfloat>(ALBuffer->Frequency) /
static_cast<ALfloat>(Device->Frequency) * props->Pitch;
- if(Pitch > (ALfloat)MAX_PITCH)
+ if(Pitch > static_cast<ALfloat>(MAX_PITCH))
voice->Step = MAX_PITCH<<FRACTIONBITS;
else
voice->Step = maxi(fastf2i(Pitch * FRACTIONONE), 1);
@@ -1363,8 +1363,8 @@ void CalcAttnSourceParams(ALvoice *voice, const ALvoicePropsBase *props, const A
/* Adjust pitch based on the buffer and output frequencies, and calculate
* fixed-point stepping value.
*/
- Pitch *= (ALfloat)ALBuffer->Frequency/(ALfloat)Device->Frequency;
- if(Pitch > (ALfloat)MAX_PITCH)
+ Pitch *= static_cast<ALfloat>(ALBuffer->Frequency)/static_cast<ALfloat>(Device->Frequency);
+ if(Pitch > static_cast<ALfloat>(MAX_PITCH))
voice->Step = MAX_PITCH<<FRACTIONBITS;
else
voice->Step = maxi(fastf2i(Pitch * FRACTIONONE), 1);
@@ -1648,7 +1648,7 @@ void ApplyDither(ALfloat (*Samples)[BUFFERSIZE], ALuint *dither_seed, const ALfl
ALfloat val = sample * quant_scale;
ALuint rng0 = dither_rng(&seed);
ALuint rng1 = dither_rng(&seed);
- val += (ALfloat)(rng0*(1.0/UINT_MAX) - rng1*(1.0/UINT_MAX));
+ val += static_cast<ALfloat>(rng0*(1.0/UINT_MAX) - rng1*(1.0/UINT_MAX));
return fast_roundf(val) * invscale;
}
);
@@ -1828,7 +1828,7 @@ void aluHandleDisconnect(ALCdevice *device, const char *msg, ...)
int msglen{vsnprintf(evt.u.user.msg, sizeof(evt.u.user.msg), msg, args)};
va_end(args);
- if(msglen < 0 || (size_t)msglen >= sizeof(evt.u.user.msg))
+ if(msglen < 0 || static_cast<size_t>(msglen) >= sizeof(evt.u.user.msg))
evt.u.user.msg[sizeof(evt.u.user.msg)-1] = 0;
ALCcontext *ctx{device->ContextList.load()};
diff --git a/Alc/backends/alsa.cpp b/Alc/backends/alsa.cpp
index 33b3ae49..5fe8ef96 100644
--- a/Alc/backends/alsa.cpp
+++ b/Alc/backends/alsa.cpp
@@ -480,7 +480,7 @@ int AlsaPlayback::mixerProc()
continue;
}
- if((snd_pcm_uframes_t)avail > update_size*(num_updates+1))
+ if(static_cast<snd_pcm_uframes_t>(avail) > update_size*(num_updates+1))
{
WARN("available samples exceeds the buffer size\n");
snd_pcm_reset(mPcmHandle);
@@ -488,7 +488,7 @@ int AlsaPlayback::mixerProc()
}
// make sure there's frames to process
- if((snd_pcm_uframes_t)avail < update_size)
+ if(static_cast<snd_pcm_uframes_t>(avail) < update_size)
{
if(state != SND_PCM_STATE_RUNNING)
{
@@ -520,7 +520,7 @@ int AlsaPlayback::mixerProc()
break;
}
- char *WritePtr{(char*)areas->addr + (offset * areas->step / 8)};
+ char *WritePtr{static_cast<char*>(areas->addr) + (offset * areas->step / 8)};
aluMixData(mDevice, WritePtr, frames);
snd_pcm_sframes_t commitres{snd_pcm_mmap_commit(mPcmHandle, offset, frames)};
@@ -563,14 +563,14 @@ int AlsaPlayback::mixerNoMMapProc()
continue;
}
- if((snd_pcm_uframes_t)avail > update_size*num_updates)
+ if(static_cast<snd_pcm_uframes_t>(avail) > update_size*num_updates)
{
WARN("available samples exceeds the buffer size\n");
snd_pcm_reset(mPcmHandle);
continue;
}
- if((snd_pcm_uframes_t)avail < update_size)
+ if(static_cast<snd_pcm_uframes_t>(avail) < update_size)
{
if(state != SND_PCM_STATE_RUNNING)
{
@@ -1112,7 +1112,7 @@ ALCenum AlsaCapture::captureSamples(ALCvoid *buffer, ALCuint samples)
{
/* First get any data stored from the last stop */
amt = snd_pcm_bytes_to_frames(mPcmHandle, mBuffer.size());
- if((snd_pcm_uframes_t)amt > samples) amt = samples;
+ if(static_cast<snd_pcm_uframes_t>(amt) > samples) amt = samples;
amt = snd_pcm_frames_to_bytes(mPcmHandle, amt);
memcpy(buffer, mBuffer.data(), amt);
@@ -1142,12 +1142,12 @@ ALCenum AlsaCapture::captureSamples(ALCvoid *buffer, ALCuint samples)
}
/* If the amount available is less than what's asked, we lost it
* during recovery. So just give silence instead. */
- if((snd_pcm_uframes_t)amt < samples)
+ if(static_cast<snd_pcm_uframes_t>(amt) < samples)
break;
continue;
}
- buffer = (ALbyte*)buffer + amt;
+ buffer = static_cast<ALbyte*>(buffer) + amt;
samples -= amt;
}
if(samples > 0)
diff --git a/Alc/backends/portaudio.cpp b/Alc/backends/portaudio.cpp
index 258f981e..074508b2 100644
--- a/Alc/backends/portaudio.cpp
+++ b/Alc/backends/portaudio.cpp
@@ -194,7 +194,7 @@ ALCenum PortPlayback::open(const ALCchar *name)
if(!ConfigValueInt(nullptr, "port", "device", &mParams.device) || mParams.device < 0)
mParams.device = Pa_GetDefaultOutputDevice();
mParams.suggestedLatency = (mDevice->UpdateSize*mDevice->NumUpdates) /
- (float)mDevice->Frequency;
+ static_cast<float>(mDevice->Frequency);
mParams.hostApiSpecificStreamInfo = nullptr;
mParams.channelCount = ((mDevice->FmtChans == DevFmtMono) ? 1 : 2);
diff --git a/Alc/backends/pulseaudio.cpp b/Alc/backends/pulseaudio.cpp
index 34c5fbfe..b717d67a 100644
--- a/Alc/backends/pulseaudio.cpp
+++ b/Alc/backends/pulseaudio.cpp
@@ -1131,7 +1131,7 @@ ALCboolean PulsePlayback::reset()
/* Server updated our playback rate, so modify the buffer attribs
* accordingly. */
mDevice->NumUpdates = static_cast<ALuint>(clampd(
- (ALdouble)mSpec.rate/mDevice->Frequency*mDevice->NumUpdates + 0.5, 2.0, 16.0));
+ static_cast<ALdouble>(mSpec.rate)/mDevice->Frequency*mDevice->NumUpdates + 0.5, 2.0, 16.0));
period_size = mDevice->UpdateSize * mFrameSize;
mAttr.maxlength = -1;
@@ -1511,10 +1511,10 @@ ALCenum PulseCapture::captureSamples(ALCvoid *buffer, ALCuint samples)
memcpy(buffer, mCapStore, rem);
- buffer = (ALbyte*)buffer + rem;
+ buffer = static_cast<ALbyte*>(buffer) + rem;
todo -= rem;
- mCapStore = (ALbyte*)mCapStore + rem;
+ mCapStore = reinterpret_cast<const ALbyte*>(mCapStore) + rem;
mCapRemain -= rem;
if(mCapRemain == 0)
{
diff --git a/Alc/bformatdec.cpp b/Alc/bformatdec.cpp
index b5dcfd89..a80ded30 100644
--- a/Alc/bformatdec.cpp
+++ b/Alc/bformatdec.cpp
@@ -75,7 +75,7 @@ void BFormatDec::reset(const AmbDecConf *conf, bool allow_2band, ALsizei inchans
{ return mask | (1 << chan); }
);
- const ALfloat xover_norm{conf->XOverFreq / (float)srate};
+ const ALfloat xover_norm{conf->XOverFreq / static_cast<float>(srate)};
const ALsizei out_order{
(conf->ChanMask > AMBI_3ORDER_MASK) ? 4 :
diff --git a/Alc/converter.cpp b/Alc/converter.cpp
index 22a01552..49c5cb3b 100644
--- a/Alc/converter.cpp
+++ b/Alc/converter.cpp
@@ -155,7 +155,7 @@ SampleConverterPtr CreateSampleConverter(DevFmtType srcType, DevFmtType dstType,
/* Have to set the mixer FPU mode since that's what the resampler code expects. */
FPUCtl mixer_mode{};
auto step = static_cast<ALsizei>(
- mind((ALdouble)srcRate/dstRate*FRACTIONONE + 0.5, MAX_PITCH*FRACTIONONE));
+ mind(static_cast<ALdouble>(srcRate)/dstRate*FRACTIONONE + 0.5, MAX_PITCH*FRACTIONONE));
converter->mIncrement = maxi(step, 1);
if(converter->mIncrement == FRACTIONONE)
converter->mResample = Resample_copy_C;
@@ -203,7 +203,7 @@ ALsizei SampleConverter::availableOut(ALsizei srcframes) const
DataSize64 -= mFracOffset;
/* If we have a full prep, we can generate at least one sample. */
- return (ALsizei)clampu64((DataSize64 + mIncrement-1)/mIncrement, 1, BUFFERSIZE);
+ return static_cast<ALsizei>(clampu64((DataSize64 + mIncrement-1)/mIncrement, 1, BUFFERSIZE));
}
ALsizei SampleConverter::convert(const ALvoid **src, ALsizei *srcframes, ALvoid *dst, ALsizei dstframes)
@@ -267,7 +267,7 @@ ALsizei SampleConverter::convert(const ALvoid **src, ALsizei *srcframes, ALvoid
for(ALsizei chan{0};chan < mNumChannels;chan++)
{
const ALbyte *SrcSamples = SamplesIn + mSrcTypeSize*chan;
- ALbyte *DstSamples = (ALbyte*)dst + mDstTypeSize*chan;
+ ALbyte *DstSamples = static_cast<ALbyte*>(dst) + mDstTypeSize*chan;
/* Load the previous samples into the source data first, then the
* new samples from the input buffer.
@@ -309,7 +309,7 @@ ALsizei SampleConverter::convert(const ALvoid **src, ALsizei *srcframes, ALvoid
SamplesIn += SrcFrameSize*(DataPosFrac>>FRACTIONBITS);
NumSrcSamples -= mini(NumSrcSamples, (DataPosFrac>>FRACTIONBITS));
- dst = (ALbyte*)dst + DstFrameSize*DstSize;
+ dst = static_cast<ALbyte*>(dst) + DstFrameSize*DstSize;
pos += DstSize;
}
diff --git a/Alc/effects/chorus.cpp b/Alc/effects/chorus.cpp
index 1132a33a..990b3cc4 100644
--- a/Alc/effects/chorus.cpp
+++ b/Alc/effects/chorus.cpp
@@ -141,7 +141,7 @@ void ChorusState::update(const ALCcontext *Context, const ALeffectslot *Slot, co
const ALCdevice *device{Context->Device};
auto frequency = static_cast<ALfloat>(device->Frequency);
mDelay = maxi(float2int(props->Chorus.Delay*frequency*FRACTIONONE + 0.5f), mindelay);
- mDepth = minf(props->Chorus.Depth * mDelay, (ALfloat)(mDelay - mindelay));
+ mDepth = minf(props->Chorus.Depth * mDelay, static_cast<ALfloat>(mDelay - mindelay));
mFeedback = props->Chorus.Feedback;
@@ -168,9 +168,9 @@ void ChorusState::update(const ALCcontext *Context, const ALeffectslot *Slot, co
/* Calculate LFO coefficient (number of samples per cycle). Limit the
* max range to avoid overflow when calculating the displacement.
*/
- ALsizei lfo_range = float2int(minf(frequency/rate + 0.5f, (ALfloat)(INT_MAX/360 - 180)));
+ ALsizei lfo_range = float2int(minf(frequency/rate + 0.5f, static_cast<ALfloat>(INT_MAX/360 - 180)));
- mLfoOffset = float2int((ALfloat)mLfoOffset/mLfoRange*lfo_range + 0.5f) % lfo_range;
+ mLfoOffset = float2int(static_cast<ALfloat>(mLfoOffset)/mLfoRange*lfo_range + 0.5f) % lfo_range;
mLfoRange = lfo_range;
switch(mWaveform)
{
diff --git a/Alc/effects/compressor.cpp b/Alc/effects/compressor.cpp
index ddf104f4..1b840c44 100644
--- a/Alc/effects/compressor.cpp
+++ b/Alc/effects/compressor.cpp
@@ -60,8 +60,8 @@ ALboolean ALcompressorState::deviceUpdate(const ALCdevice *device)
/* Number of samples to do a full attack and release (non-integer sample
* counts are okay).
*/
- const ALfloat attackCount = (ALfloat)device->Frequency * ATTACK_TIME;
- const ALfloat releaseCount = (ALfloat)device->Frequency * RELEASE_TIME;
+ const ALfloat attackCount = static_cast<ALfloat>(device->Frequency) * ATTACK_TIME;
+ const ALfloat releaseCount = static_cast<ALfloat>(device->Frequency) * RELEASE_TIME;
/* Calculate per-sample multipliers to attack and release at the desired
* rates.
diff --git a/Alc/effects/equalizer.cpp b/Alc/effects/equalizer.cpp
index 94c760ea..defe1485 100644
--- a/Alc/effects/equalizer.cpp
+++ b/Alc/effects/equalizer.cpp
@@ -113,7 +113,7 @@ ALboolean ALequalizerState::deviceUpdate(const ALCdevice *UNUSED(device))
void ALequalizerState::update(const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props, const EffectTarget target)
{
const ALCdevice *device = context->Device;
- ALfloat frequency = (ALfloat)device->Frequency;
+ ALfloat frequency = static_cast<ALfloat>(device->Frequency);
ALfloat gain, f0norm;
ALuint i;
diff --git a/Alc/effects/fshifter.cpp b/Alc/effects/fshifter.cpp
index c444872c..994dd90c 100644
--- a/Alc/effects/fshifter.cpp
+++ b/Alc/effects/fshifter.cpp
@@ -111,7 +111,7 @@ void ALfshifterState::update(const ALCcontext *context, const ALeffectslot *slot
{
const ALCdevice *device{context->Device};
- ALfloat step{props->Fshifter.Frequency / (ALfloat)device->Frequency};
+ ALfloat step{props->Fshifter.Frequency / static_cast<ALfloat>(device->Frequency)};
mPhaseStep = fastf2i(minf(step, 0.5f) * FRACTIONONE);
switch(props->Fshifter.LeftDirection)
@@ -190,7 +190,7 @@ void ALfshifterState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT Samp
for(k = 0;k < SamplesToDo;k++)
{
double phase = mPhase * ((1.0/FRACTIONONE) * al::MathDefs<double>::Tau());
- BufferOut[k] = (float)(mOutdata[k].real()*std::cos(phase) +
+ BufferOut[k] = static_cast<float>(mOutdata[k].real()*std::cos(phase) +
mOutdata[k].imag()*std::sin(phase)*mLdSign);
mPhase += mPhaseStep;
diff --git a/Alc/effects/modulator.cpp b/Alc/effects/modulator.cpp
index 3544188b..9549740e 100644
--- a/Alc/effects/modulator.cpp
+++ b/Alc/effects/modulator.cpp
@@ -43,17 +43,17 @@
static inline ALfloat Sin(ALsizei index)
{
- return std::sin((ALfloat)index * (al::MathDefs<float>::Tau() / (ALfloat)WAVEFORM_FRACONE));
+ return std::sin(static_cast<ALfloat>(index) * (al::MathDefs<float>::Tau() / static_cast<ALfloat>WAVEFORM_FRACONE));
}
static inline ALfloat Saw(ALsizei index)
{
- return (ALfloat)index*(2.0f/WAVEFORM_FRACONE) - 1.0f;
+ return static_cast<ALfloat>(index)*(2.0f/WAVEFORM_FRACONE) - 1.0f;
}
static inline ALfloat Square(ALsizei index)
{
- return (ALfloat)(((index>>(WAVEFORM_FRACBITS-2))&2) - 1);
+ return static_cast<ALfloat>(((index>>(WAVEFORM_FRACBITS-2))&2) - 1);
}
static inline ALfloat One(ALsizei UNUSED(index))
@@ -111,7 +111,7 @@ void ALmodulatorState::update(const ALCcontext *context, const ALeffectslot *slo
ALfloat f0norm;
ALsizei i;
- mStep = fastf2i(props->Modulator.Frequency / (ALfloat)device->Frequency * WAVEFORM_FRACONE);
+ mStep = fastf2i(props->Modulator.Frequency / static_cast<ALfloat>(device->Frequency) * WAVEFORM_FRACONE);
mStep = clampi(mStep, 0, WAVEFORM_FRACONE-1);
if(mStep == 0)
@@ -123,7 +123,7 @@ void ALmodulatorState::update(const ALCcontext *context, const ALeffectslot *slo
else /*if(Slot->Params.EffectProps.Modulator.Waveform == AL_RING_MODULATOR_SQUARE)*/
mGetSamples = Modulate<Square>;
- f0norm = props->Modulator.HighPassCutoff / (ALfloat)device->Frequency;
+ f0norm = props->Modulator.HighPassCutoff / static_cast<ALfloat>(device->Frequency);
f0norm = clampf(f0norm, 1.0f/512.0f, 0.49f);
/* Bandwidth value is constant in octaves. */
mChans[0].Filter.setParams(BiquadType::HighPass, 1.0f, f0norm,
@@ -214,7 +214,7 @@ void ALmodulator_setParami(ALeffect *effect, ALCcontext *context, ALenum param,
{
case AL_RING_MODULATOR_FREQUENCY:
case AL_RING_MODULATOR_HIGHPASS_CUTOFF:
- ALmodulator_setParamf(effect, context, param, (ALfloat)val);
+ ALmodulator_setParamf(effect, context, param, static_cast<ALfloat>(val));
break;
case AL_RING_MODULATOR_WAVEFORM:
@@ -236,10 +236,10 @@ void ALmodulator_getParami(const ALeffect *effect, ALCcontext *context, ALenum p
switch(param)
{
case AL_RING_MODULATOR_FREQUENCY:
- *val = (ALint)props->Modulator.Frequency;
+ *val = static_cast<ALint>(props->Modulator.Frequency);
break;
case AL_RING_MODULATOR_HIGHPASS_CUTOFF:
- *val = (ALint)props->Modulator.HighPassCutoff;
+ *val = static_cast<ALint>(props->Modulator.HighPassCutoff);
break;
case AL_RING_MODULATOR_WAVEFORM:
*val = props->Modulator.Waveform;
diff --git a/Alc/effects/pshifter.cpp b/Alc/effects/pshifter.cpp
index 7c6fb51e..f0b9de1c 100644
--- a/Alc/effects/pshifter.cpp
+++ b/Alc/effects/pshifter.cpp
@@ -72,7 +72,7 @@ inline int double2int(double d)
#else
- return (ALint)d;
+ return static_cast<ALint>(d);
#endif
}
@@ -156,7 +156,7 @@ ALboolean ALpshifterState::deviceUpdate(const ALCdevice *device)
mCount = FIFO_LATENCY;
mPitchShiftI = FRACTIONONE;
mPitchShift = 1.0f;
- mFreqPerBin = device->Frequency / (ALfloat)STFT_SIZE;
+ mFreqPerBin = device->Frequency / static_cast<ALfloat>(STFT_SIZE);
std::fill(std::begin(mInFIFO), std::end(mInFIFO), 0.0f);
std::fill(std::begin(mOutFIFO), std::end(mOutFIFO), 0.0f);
@@ -176,7 +176,7 @@ ALboolean ALpshifterState::deviceUpdate(const ALCdevice *device)
void ALpshifterState::update(const ALCcontext* UNUSED(context), const ALeffectslot *slot, const ALeffectProps *props, const EffectTarget target)
{
const float pitch{std::pow(2.0f,
- (ALfloat)(props->Pshifter.CoarseTune*100 + props->Pshifter.FineTune) / 1200.0f
+ static_cast<ALfloat>(props->Pshifter.CoarseTune*100 + props->Pshifter.FineTune) / 1200.0f
)};
mPitchShiftI = fastf2i(pitch*FRACTIONONE);
mPitchShift = mPitchShiftI * (1.0f/FRACTIONONE);
@@ -304,7 +304,7 @@ void ALpshifterState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT Samp
/* Shift accumulator, input & output FIFO */
ALsizei j, k;
- for(k = 0;k < STFT_STEP;k++) mOutFIFO[k] = (ALfloat)mOutputAccum[k];
+ for(k = 0;k < STFT_STEP;k++) mOutFIFO[k] = static_cast<ALfloat>(mOutputAccum[k]);
for(j = 0;k < STFT_SIZE;k++,j++) mOutputAccum[j] = mOutputAccum[k];
for(;j < STFT_SIZE;j++) mOutputAccum[j] = 0.0;
for(k = 0;k < FIFO_LATENCY;k++)
@@ -375,10 +375,10 @@ void ALpshifter_getParami(const ALeffect *effect, ALCcontext *context, ALenum pa
switch(param)
{
case AL_PITCH_SHIFTER_COARSE_TUNE:
- *val = (ALint)props->Pshifter.CoarseTune;
+ *val = props->Pshifter.CoarseTune;
break;
case AL_PITCH_SHIFTER_FINE_TUNE:
- *val = (ALint)props->Pshifter.FineTune;
+ *val = props->Pshifter.FineTune;
break;
default:
diff --git a/Alc/effects/reverb.cpp b/Alc/effects/reverb.cpp
index 6f1b1bb1..a63cc4c3 100644
--- a/Alc/effects/reverb.cpp
+++ b/Alc/effects/reverb.cpp
@@ -359,7 +359,7 @@ inline ALvoid RealizeLineOffset(ALfloat *sampleBuffer, DelayLineI *Delay)
ALfloat *f;
ALfloat (*f4)[NUM_LINES];
} u;
- u.f = &sampleBuffer[(ptrdiff_t)Delay->Line * NUM_LINES];
+ u.f = &sampleBuffer[reinterpret_cast<ptrdiff_t>(Delay->Line) * NUM_LINES];
Delay->Line = u.f4;
}
@@ -377,7 +377,7 @@ ALuint CalcLineLength(const ALfloat length, const ptrdiff_t offset, const ALuint
/* All lines share a single sample buffer. */
Delay->Mask = samples - 1;
- Delay->Line = (ALfloat(*)[NUM_LINES])offset;
+ Delay->Line = reinterpret_cast<ALfloat(*)[NUM_LINES]>(offset);
/* Return the sample count for accumulation. */
return samples;
@@ -658,7 +658,7 @@ ALvoid UpdateLateLines(const ALfloat density, const ALfloat diffusion, const ALf
/* Scaling factor to convert the normalized reference frequencies from
* representing 0...freq to 0...max_reference.
*/
- const ALfloat norm_weight_factor = (ALfloat)frequency / AL_EAXREVERB_MAX_HFREFERENCE;
+ const ALfloat norm_weight_factor = static_cast<ALfloat>(frequency) / AL_EAXREVERB_MAX_HFREFERENCE;
/* To compensate for changes in modal density and decay time of the late
* reverb signal, the input is attenuated based on the maximal energy of
diff --git a/Alc/helpers.cpp b/Alc/helpers.cpp
index 4a80c7e5..88c222bf 100644
--- a/Alc/helpers.cpp
+++ b/Alc/helpers.cpp
@@ -531,7 +531,7 @@ const PathNamePair &GetProcBinary()
len = readlink(selfname, pathname.data(), pathname.size());
}
- while(len > 0 && (size_t)len == pathname.size())
+ while(len > 0 && static_cast<size_t>(len) == pathname.size())
{
pathname.resize(pathname.size() << 1);
len = readlink(selfname, pathname.data(), pathname.size());
diff --git a/Alc/hrtf.cpp b/Alc/hrtf.cpp
index 070de55f..4d1c27ee 100644
--- a/Alc/hrtf.cpp
+++ b/Alc/hrtf.cpp
@@ -329,7 +329,7 @@ void BuildBFormatHrtf(const HrtfEntry *Hrtf, DirectHrtfState *state, const ALsiz
{
for(ALsizei i{0};i < NumChannels;++i)
{
- const ALdouble mult{(ALdouble)AmbiOrderHFGain[OrderFromChan[i]] * AmbiMatrix[c][i]};
+ const ALdouble mult{static_cast<ALdouble>(AmbiOrderHFGain[OrderFromChan[i]]) * AmbiMatrix[c][i]};
const ALsizei numirs{mini(Hrtf->irSize, HRIR_LENGTH-maxi(ldelay, rdelay))};
ALsizei lidx{ldelay}, ridx{rdelay};
for(ALsizei j{0};j < numirs;++j)
@@ -384,8 +384,8 @@ void BuildBFormatHrtf(const HrtfEntry *Hrtf, DirectHrtfState *state, const ALsiz
{
for(ALsizei idx{0};idx < HRIR_LENGTH;idx++)
{
- state->Chan[i].Coeffs[idx][0] = (ALfloat)tmpres[i][idx][0];
- state->Chan[i].Coeffs[idx][1] = (ALfloat)tmpres[i][idx][1];
+ state->Chan[i].Coeffs[idx][0] = static_cast<ALfloat>(tmpres[i][idx][0]);
+ state->Chan[i].Coeffs[idx][1] = static_cast<ALfloat>(tmpres[i][idx][1]);
}
}
tmpres.clear();
@@ -435,7 +435,7 @@ HrtfEntry *CreateHrtfStore(ALuint rate, ALsizei irSize, ALfloat distance, ALsize
else
{
uintptr_t offset = sizeof(HrtfEntry);
- char *base = (char*)Hrtf;
+ char *base = reinterpret_cast<char*>(Hrtf);
ALushort *_evOffset;
ALubyte *_azCount;
ALubyte (*_delays)[2];
@@ -941,7 +941,7 @@ HrtfEntry *LoadHrtf02(std::istream &data, const char *filename)
}
return CreateHrtfStore(rate, irSize,
- (ALfloat)distance / 1000.0f, evCount, irCount, azCount.data(), evOffset.data(),
+ static_cast<ALfloat>(distance) / 1000.0f, evCount, irCount, azCount.data(), evOffset.data(),
&reinterpret_cast<ALfloat(&)[2]>(coeffs[0]),
&reinterpret_cast<ALubyte(&)[2]>(delays[0]), filename
);
diff --git a/Alc/mastering.cpp b/Alc/mastering.cpp
index dcc5cf40..c71b3cc9 100644
--- a/Alc/mastering.cpp
+++ b/Alc/mastering.cpp
@@ -398,15 +398,15 @@ std::unique_ptr<Compressor> CompressorInit(const ALsizei NumChans, const ALuint
{
if(hold > 1)
{
- Comp->mHold = new ((void*)(Comp.get() + 1)) SlidingHold{};
+ Comp->mHold = new (reinterpret_cast<void*>(Comp.get() + 1)) SlidingHold{};
Comp->mHold->mValues[0] = -std::numeric_limits<float>::infinity();
Comp->mHold->mExpiries[0] = hold;
Comp->mHold->mLength = hold;
- Comp->mDelay = (ALfloat(*)[BUFFERSIZE])(Comp->mHold + 1);
+ Comp->mDelay = reinterpret_cast<ALfloat(*)[BUFFERSIZE]>(Comp->mHold + 1);
}
else
{
- Comp->mDelay = (ALfloat(*)[BUFFERSIZE])(Comp.get() + 1);
+ Comp->mDelay = reinterpret_cast<ALfloat(*)[BUFFERSIZE]>(Comp.get() + 1);
}
}
diff --git a/Alc/mixer/mixer_c.cpp b/Alc/mixer/mixer_c.cpp
index 31a5cee4..1b16b733 100644
--- a/Alc/mixer/mixer_c.cpp
+++ b/Alc/mixer/mixer_c.cpp
@@ -46,7 +46,7 @@ const ALfloat *Resample_copy_C(const InterpState* UNUSED(state),
ASSUME(numsamples > 0);
#if defined(HAVE_SSE) || defined(HAVE_NEON)
/* Avoid copying the source data if it's aligned like the destination. */
- if((((intptr_t)src)&15) == (((intptr_t)dst)&15))
+ if((reinterpret_cast<intptr_t>(src)&15) == (reinterpret_cast<intptr_t>(dst)&15))
return src;
#endif
std::copy_n(src, numsamples, dst);
@@ -137,7 +137,7 @@ void Mix_C(const ALfloat *data, ALsizei OutChans, ALfloat (*RESTRICT OutBuffer)[
ASSUME(OutChans > 0);
ASSUME(BufferSize > 0);
- const ALfloat delta{(Counter > 0) ? 1.0f/(ALfloat)Counter : 0.0f};
+ const ALfloat delta{(Counter > 0) ? 1.0f / static_cast<ALfloat>(Counter) : 0.0f};
for(ALsizei c{0};c < OutChans;c++)
{
ALsizei pos{0};
diff --git a/Alc/mixer/mixer_sse.cpp b/Alc/mixer/mixer_sse.cpp
index df5270e7..9eadfd9e 100644
--- a/Alc/mixer/mixer_sse.cpp
+++ b/Alc/mixer/mixer_sse.cpp
@@ -37,10 +37,10 @@ const ALfloat *Resample_bsinc_SSE(const InterpState *state, const ALfloat *RESTR
#undef FRAC_PHASE_BITDIFF
ALsizei offset{m*pi*4};
- const __m128 *fil{(const __m128*)(filter + offset)}; offset += m;
- const __m128 *scd{(const __m128*)(filter + offset)}; offset += m;
- const __m128 *phd{(const __m128*)(filter + offset)}; offset += m;
- const __m128 *spd{(const __m128*)(filter + offset)};
+ const __m128 *fil{reinterpret_cast<const __m128*>(filter + offset)}; offset += m;
+ const __m128 *scd{reinterpret_cast<const __m128*>(filter + offset)}; offset += m;
+ const __m128 *phd{reinterpret_cast<const __m128*>(filter + offset)}; offset += m;
+ const __m128 *spd{reinterpret_cast<const __m128*>(filter + offset)};
// Apply the scale and phase interpolated filter.
__m128 r4{_mm_setzero_ps()};
@@ -92,10 +92,10 @@ static inline void ApplyCoeffs(ALsizei Offset, ALfloat (&Values)[HRIR_LENGTH][2]
__m128 imp0, imp1;
__m128 coeffs{_mm_load_ps(&Coeffs[0][0])};
- __m128 vals{_mm_loadl_pi(_mm_setzero_ps(), (__m64*)&Values[Offset][0])};
+ __m128 vals{_mm_loadl_pi(_mm_setzero_ps(), reinterpret_cast<__m64*>(&Values[Offset][0]))};
imp0 = _mm_mul_ps(lrlr, coeffs);
vals = _mm_add_ps(imp0, vals);
- _mm_storel_pi((__m64*)&Values[Offset][0], vals);
+ _mm_storel_pi(reinterpret_cast<__m64*>(&Values[Offset][0]), vals);
++Offset;
for(ALsizei i{1};;)
{
@@ -115,10 +115,10 @@ static inline void ApplyCoeffs(ALsizei Offset, ALfloat (&Values)[HRIR_LENGTH][2]
break;
count = IrSize-1;
}
- vals = _mm_loadl_pi(vals, (__m64*)&Values[Offset][0]);
+ vals = _mm_loadl_pi(vals, reinterpret_cast<__m64*>(&Values[Offset][0]));
imp0 = _mm_movehl_ps(imp0, imp0);
vals = _mm_add_ps(imp0, vals);
- _mm_storel_pi((__m64*)&Values[Offset][0], vals);
+ _mm_storel_pi(reinterpret_cast<__m64*>(&Values[Offset][0]), vals);
}
else
{
@@ -156,7 +156,7 @@ void Mix_SSE(const ALfloat *data, ALsizei OutChans, ALfloat (*RESTRICT OutBuffer
ASSUME(OutChans > 0);
ASSUME(BufferSize > 0);
- const ALfloat delta{(Counter > 0) ? 1.0f/(ALfloat)Counter : 0.0f};
+ const ALfloat delta{(Counter > 0) ? 1.0f / static_cast<ALfloat>(Counter) : 0.0f};
for(ALsizei c{0};c < OutChans;c++)
{
ALsizei pos{0};
diff --git a/Alc/mixvoice.cpp b/Alc/mixvoice.cpp
index eb219bad..e52b330d 100644
--- a/Alc/mixvoice.cpp
+++ b/Alc/mixvoice.cpp
@@ -211,7 +211,7 @@ template<> inline ALfloat LoadSample<FmtShort>(FmtTypeTraits<FmtShort>::Type val
template<> inline ALfloat LoadSample<FmtFloat>(FmtTypeTraits<FmtFloat>::Type val)
{ return val; }
template<> inline ALfloat LoadSample<FmtDouble>(FmtTypeTraits<FmtDouble>::Type val)
-{ return (ALfloat)val; }
+{ return static_cast<ALfloat>(val); }
template<> inline ALfloat LoadSample<FmtMulaw>(FmtTypeTraits<FmtMulaw>::Type val)
{ return muLawDecompressionTable[val] * (1.0f/32768.0f); }
template<> inline ALfloat LoadSample<FmtAlaw>(FmtTypeTraits<FmtAlaw>::Type val)
@@ -295,7 +295,7 @@ ALboolean MixSource(ALvoice *voice, const ALuint SourceID, ALCcontext *Context,
/* Get source info */
bool isplaying{true}; /* Will only be called while playing. */
bool isstatic{(voice->Flags&VOICE_IS_STATIC) != 0};
- ALsizei DataPosInt{(ALsizei)voice->position.load(std::memory_order_acquire)};
+ ALsizei DataPosInt{static_cast<ALsizei>(voice->position.load(std::memory_order_acquire))};
ALsizei DataPosFrac{voice->position_fraction.load(std::memory_order_relaxed)};
ALbufferlistitem *BufferListItem{voice->current_buffer.load(std::memory_order_relaxed)};
ALbufferlistitem *BufferLoopItem{voice->loop_buffer.load(std::memory_order_relaxed)};
@@ -603,13 +603,13 @@ ALboolean MixSource(ALvoice *voice, const ALuint SourceID, ALCcontext *Context,
* this mix handles.
*/
ALfloat gain{lerp(parms.Hrtf.Old.Gain, parms.Hrtf.Target.Gain,
- minf(1.0f, (ALfloat)fademix/Counter))};
+ minf(1.0f, static_cast<ALfloat>(fademix))/Counter)};
MixHrtfParams hrtfparams;
hrtfparams.Coeffs = &parms.Hrtf.Target.Coeffs;
hrtfparams.Delay[0] = parms.Hrtf.Target.Delay[0];
hrtfparams.Delay[1] = parms.Hrtf.Target.Delay[1];
hrtfparams.Gain = 0.0f;
- hrtfparams.GainStep = gain / (ALfloat)fademix;
+ hrtfparams.GainStep = gain / static_cast<ALfloat>(fademix);
MixHrtfBlendSamples(
voice->Direct.Buffer[OutLIdx], voice->Direct.Buffer[OutRIdx],
@@ -631,14 +631,14 @@ ALboolean MixSource(ALvoice *voice, const ALuint SourceID, ALCcontext *Context,
*/
if(Counter > DstBufferSize)
gain = lerp(parms.Hrtf.Old.Gain, gain,
- (ALfloat)todo/(Counter-fademix));
+ static_cast<ALfloat>(todo)/(Counter-fademix));
MixHrtfParams hrtfparams;
hrtfparams.Coeffs = &parms.Hrtf.Target.Coeffs;
hrtfparams.Delay[0] = parms.Hrtf.Target.Delay[0];
hrtfparams.Delay[1] = parms.Hrtf.Target.Delay[1];
hrtfparams.Gain = parms.Hrtf.Old.Gain;
- hrtfparams.GainStep = (gain - parms.Hrtf.Old.Gain) / (ALfloat)todo;
+ hrtfparams.GainStep = (gain - parms.Hrtf.Old.Gain) / static_cast<ALfloat>(todo);
MixHrtfSamples(
voice->Direct.Buffer[OutLIdx], voice->Direct.Buffer[OutRIdx],
samples+fademix, voice->Offset+fademix, OutPos+fademix, IrSize,
diff --git a/Alc/panning.cpp b/Alc/panning.cpp
index c311e31f..e184a13d 100644
--- a/Alc/panning.cpp
+++ b/Alc/panning.cpp
@@ -273,12 +273,12 @@ void InitDistanceComp(ALCdevice *device, const AmbDecConf *conf, const ALsizei (
const ALfloat delay{
std::floor((maxdist - speaker.Distance)/SPEEDOFSOUNDMETRESPERSEC*srate + 0.5f)
};
- if(delay >= (ALfloat)MAX_DELAY_LENGTH)
+ if(delay >= static_cast<ALfloat>(MAX_DELAY_LENGTH))
ERR("Delay for speaker \"%s\" exceeds buffer length (%f >= %d)\n",
speaker.Name.c_str(), delay, MAX_DELAY_LENGTH);
device->ChannelDelay[chan].Length = static_cast<ALsizei>(clampf(
- delay, 0.0f, (ALfloat)(MAX_DELAY_LENGTH-1)
+ delay, 0.0f, static_cast<ALfloat>(MAX_DELAY_LENGTH-1)
));
device->ChannelDelay[chan].Gain = speaker.Distance / maxdist;
TRACE("Channel %u \"%s\" distance compensation: %d samples, %f gain\n", chan,
@@ -951,7 +951,7 @@ void aluInitRenderer(ALCdevice *device, ALint hrtf_id, HrtfRequestMode hrtf_appr
* front-right channels, with a crossover at 5khz (could be
* higher).
*/
- const ALfloat scale{(ALfloat)(5000.0 / device->Frequency)};
+ const ALfloat scale{static_cast<ALfloat>(5000.0 / device->Frequency)};
stablizer->LFilter.init(scale);
stablizer->RFilter = stablizer->LFilter;
@@ -1016,7 +1016,7 @@ void aluInitRenderer(ALCdevice *device, ALint hrtf_id, HrtfRequestMode hrtf_appr
if(device->HrtfList.empty())
device->HrtfList = EnumerateHrtf(device->DeviceName.c_str());
- if(hrtf_id >= 0 && (size_t)hrtf_id < device->HrtfList.size())
+ if(hrtf_id >= 0 && static_cast<size_t>(hrtf_id) < device->HrtfList.size())
{
const EnumeratedHrtf &entry = device->HrtfList[hrtf_id];
HrtfEntry *hrtf{GetLoadedHrtf(entry.hrtf)};
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-02 01:43:51 +0000