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stella
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Minor formatting cleanups for recent commits.

This commit is contained in:
Stephen Anthony 2022-09-03 20:10:22 -02:30
parent 21f3b319c2
commit 2296b334dc
5 changed files with 54 additions and 53 deletions
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95
src/common/SoundSDL2.cxx
View File

@ -28,16 +28,16 @@
#include "System.hxx"
#include "OSystem.hxx"
#include "Console.hxx"
#include "SoundSDL2.hxx"
#include "AudioQueue.hxx"
#include "EmulationTiming.hxx"
#include "AudioSettings.hxx"
#include "audio/SimpleResampler.hxx"
#include "audio/LanczosResampler.hxx"
#include "StaggeredLogger.hxx"
#include "ThreadDebugging.hxx"
#include "SoundSDL2.hxx"
// - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
SoundSDL2::SoundSDL2(OSystem& osystem, AudioSettings& audioSettings)
: Sound{osystem},
@ -47,7 +47,8 @@ SoundSDL2::SoundSDL2(OSystem& osystem, AudioSettings& audioSettings)
Logger::debug("SoundSDL2::SoundSDL2 started ...");
if (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) {
if(SDL_InitSubSystem(SDL_INIT_AUDIO) < 0)
{
ostringstream buf;
buf << "WARNING: Failed to initialize SDL audio system! " << endl
@ -72,12 +73,14 @@ SoundSDL2::~SoundSDL2()
{
ASSERT_MAIN_THREAD;
stopWav(); // NOLINT
if(!myIsInitializedFlag)
return;
if(myWavDevice)
{
SDL_CloseAudioDevice(myWavDevice);
if (!myIsInitializedFlag) return;
SDL_FreeWAV(myWavBuffer);
}
SDL_CloseAudioDevice(myDevice);
SDL_QuitSubSystem(SDL_INIT_AUDIO);
}
@ -95,7 +98,8 @@ void SoundSDL2::queryHardware(VariantList& devices)
Logger::debug(s.str());
VarList::push_back(devices, "Default", 0);
for(int i = 0; i < numDevices; ++i) {
for(int i = 0; i < numDevices; ++i)
{
ostringstream ss;
ss << " " << i + 1 << ": " << SDL_GetAudioDeviceName(i, 0);
@ -144,7 +148,8 @@ bool SoundSDL2::openDevice()
void SoundSDL2::setEnabled(bool enable)
{
myAudioSettings.setEnabled(enable);
if (myAudioQueue) myAudioQueue->ignoreOverflows(!enable);
if(myAudioQueue)
myAudioQueue->ignoreOverflows(!enable);
Logger::debug(enable ? "SoundSDL2::setEnabled(true)" :
"SoundSDL2::setEnabled(false)");
@ -199,11 +204,13 @@ void SoundSDL2::open(shared_ptr<AudioQueue> audioQueue,
void SoundSDL2::close()
{
stopWav();
if(!myIsInitializedFlag) return;
if(!myIsInitializedFlag)
return;
mute(true);
if (myAudioQueue) myAudioQueue->closeSink(myCurrentFragment);
if(myAudioQueue)
myAudioQueue->closeSink(myCurrentFragment);
myAudioQueue.reset();
myCurrentFragment = nullptr;
}
@ -233,21 +240,6 @@ bool SoundSDL2::toggleMute()
myOSystem.frameBuffer().showTextMessage(message);
//ostringstream strval;
//uInt32 volume;
//// Now show an onscreen message
//if(enabled)
//{
// volume = myVolume;
// strval << volume << "%";
//}
//else
//{
// volume = 0;
// strval << "Muted";
//}
//myOSystem.frameBuffer().showMessage("Volume", strval.str(), volume);
return enabled;
}
@ -268,9 +260,7 @@ void SoundSDL2::setVolume(uInt32 percent)
// - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
void SoundSDL2::adjustVolume(int direction)
{
ostringstream strval;
Int32 percent = myVolume;
percent = BSPF::clamp(percent + direction * 2, 0, 100);
setVolume(percent);
@ -285,10 +275,8 @@ void SoundSDL2::adjustVolume(int direction)
}
// Now show an onscreen message
if(percent)
strval << percent << "%";
else
strval << "Off";
ostringstream strval;
(percent) ? strval << percent << "%" : strval << "Off";
myOSystem.frameBuffer().showGaugeMessage("Volume", strval.str(), percent);
}
@ -302,7 +290,8 @@ string SoundSDL2::about() const
<< " Channels: " << static_cast<uInt32>(myHardwareSpec.channels)
<< (myAudioQueue->isStereo() ? " (Stereo)" : " (Mono)") << endl
<< " Preset: ";
switch (myAudioSettings.preset()) {
switch(myAudioSettings.preset())
{
case AudioSettings::Preset::custom:
buf << "Custom" << endl;
break;
@ -319,8 +308,10 @@ string SoundSDL2::about() const
buf << "Ultra quality, minimal lag" << endl;
break;
}
buf << " Fragment size: " << static_cast<uInt32>(myHardwareSpec.samples) << " bytes" << endl
<< " Sample rate: " << static_cast<uInt32>(myHardwareSpec.freq) << " Hz" << endl;
buf << " Fragment size: " << static_cast<uInt32>(myHardwareSpec.samples)
<< " bytes" << endl
<< " Sample rate: " << static_cast<uInt32>(myHardwareSpec.freq)
<< " Hz" << endl;
buf << " Resampling: ";
switch(myAudioSettings.resamplingQuality())
{
@ -346,7 +337,7 @@ void SoundSDL2::processFragment(float* stream, uInt32 length)
{
myResampler->fillFragment(stream, length);
for (uInt32 i = 0; i < length; ++i)
for(uInt32 i = 0; i < length; ++i)
stream[i] *= myVolumeFactor;
}
@ -356,34 +347,41 @@ void SoundSDL2::initResampler()
Resampler::NextFragmentCallback nextFragmentCallback = [this] () -> Int16* {
Int16* nextFragment = nullptr;
if (myUnderrun)
if(myUnderrun)
nextFragment = myAudioQueue->size() >= myEmulationTiming->prebufferFragmentCount() ?
myAudioQueue->dequeue(myCurrentFragment) : nullptr;
else
nextFragment = myAudioQueue->dequeue(myCurrentFragment);
myUnderrun = nextFragment == nullptr;
if (nextFragment) myCurrentFragment = nextFragment;
if(nextFragment)
myCurrentFragment = nextFragment;
return nextFragment;
};
Resampler::Format formatFrom =
Resampler::Format(myEmulationTiming->audioSampleRate(), myAudioQueue->fragmentSize(), myAudioQueue->isStereo());
Resampler::Format(myEmulationTiming->audioSampleRate(),
myAudioQueue->fragmentSize(), myAudioQueue->isStereo());
Resampler::Format formatTo =
Resampler::Format(myHardwareSpec.freq, myHardwareSpec.samples, myHardwareSpec.channels > 1);
Resampler::Format(myHardwareSpec.freq, myHardwareSpec.samples,
myHardwareSpec.channels > 1);
switch (myAudioSettings.resamplingQuality()) {
switch(myAudioSettings.resamplingQuality())
{
case AudioSettings::ResamplingQuality::nearestNeightbour:
myResampler = make_unique<SimpleResampler>(formatFrom, formatTo, nextFragmentCallback);
myResampler = make_unique<SimpleResampler>(formatFrom, formatTo,
nextFragmentCallback);
break;
case AudioSettings::ResamplingQuality::lanczos_2:
myResampler = make_unique<LanczosResampler>(formatFrom, formatTo, nextFragmentCallback, 2);
myResampler = make_unique<LanczosResampler>(formatFrom, formatTo,
nextFragmentCallback, 2);
break;
case AudioSettings::ResamplingQuality::lanczos_3:
myResampler = make_unique<LanczosResampler>(formatFrom, formatTo, nextFragmentCallback, 3);
myResampler = make_unique<LanczosResampler>(formatFrom, formatTo,
nextFragmentCallback, 3);
break;
default:
@ -396,14 +394,14 @@ void SoundSDL2::callback(void* udata, uInt8* stream, int len)
{
auto* self = static_cast<SoundSDL2*>(udata);
if (self->myAudioQueue)
if(self->myAudioQueue)
self->processFragment(reinterpret_cast<float*>(stream), len >> 2);
else
SDL_memset(stream, 0, len);
}
// - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
bool SoundSDL2::playWav(const char* fileName, uInt32 position, uInt32 length)
bool SoundSDL2::playWav(const string& fileName, uInt32 position, uInt32 length)
{
uInt32 wavLength{0};
@ -411,7 +409,7 @@ bool SoundSDL2::playWav(const char* fileName, uInt32 position, uInt32 length)
stopWav();
// Load WAV file
SDL_AudioSpec* result = SDL_LoadWAV(fileName, &myWavSpec, &myWavBuffer, &wavLength);
auto* result = SDL_LoadWAV(fileName.c_str(), &myWavSpec, &myWavBuffer, &wavLength);
if(result == nullptr || position > wavLength)
return false;
@ -452,6 +450,7 @@ void SoundSDL2::stopWav()
}
}
// - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
uInt32 SoundSDL2::wavSize() const
{
return myWavBuffer ? myWavLen /*SDL_GetQueuedAudioSize(myWavDevice)*/ : 0;
@ -465,6 +464,7 @@ void SoundSDL2::wavCallback(void* udata, uInt8* stream, int len)
{
if(static_cast<uInt32>(len) > myWavLen)
len = myWavLen;
// Mix volume adjusted WAV into silent buffer
SDL_MixAudioFormat(stream, myWavPos, myWavSpec.format, len,
SDL_MIX_MAXVOLUME * myVolumeFactor);
@ -473,6 +473,7 @@ void SoundSDL2::wavCallback(void* udata, uInt8* stream, int len)
}
}
// - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
float SoundSDL2::myVolumeFactor = 0xffff;
SDL_AudioSpec SoundSDL2::myWavSpec; // audio output format
uInt8* SoundSDL2::myWavPos = nullptr; // pointer to the audio buffer to be played

3
src/common/SoundSDL2.hxx
View File

@ -116,7 +116,8 @@ class SoundSDL2 : public Sound
@return True, if the WAV file can be played
*/
bool playWav(const char* fileName, uInt32 position = 0, uInt32 length = 0) override;
bool playWav(const string& fileName, uInt32 position = 0,
uInt32 length = 0) override;
/**
Stop any currently playing WAV file.

5
src/emucore/KidVid.cxx
View File

@ -191,13 +191,12 @@ void KidVid::setNextSong()
mySongLength = ourSongStart[temp + 1] - ourSongStart[temp];
// Play the WAV file
const string fileName = (temp < 10) ? myBaseDir + "KVSHARED.WAV" : mySampleFile;
mySound.playWav(fileName.c_str(), ourSongStart[temp], mySongLength);
const string& fileName = (temp < 10) ? myBaseDir + "KVSHARED.WAV" : mySampleFile;
mySound.playWav(fileName, ourSongStart[temp], mySongLength);
#ifdef DEBUG_BUILD
cerr << fileName << ": " << (ourSongPositions[mySongPointer] & 0x7f) << endl;
#endif
mySongPlaying = myTapeBusy = true;
++mySongPointer;
}

2
src/emucore/KidVid.hxx
View File

@ -33,7 +33,7 @@ class Sound;
This code was heavily borrowed from z26.
@author Stephen Anthony & z26 team
@author Thomas Jentzsch & z26 team
*/
class KidVid : public Controller
{

2
src/emucore/Sound.hxx
View File

@ -114,7 +114,7 @@ class Sound
@return True, if the WAV file can be played
*/
virtual bool playWav(const char* fileName, uInt32 position = 0,
virtual bool playWav(const string& fileName, uInt32 position = 0,
uInt32 length = 0) { return false; }
/**