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Sound 0.3

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Javidx9 6 years ago committed by GitHub
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  1. 713
      olcPGEX_Sound.h

713
olcPGEX_Sound.h
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@ -3,7 +3,7 @@
+-------------------------------------------------------------+
| OneLoneCoder Pixel Game Engine Extension |
| Sound - v0.2 |
| Sound - v0.3 |
+-------------------------------------------------------------+
What is this?
@ -11,10 +11,18 @@
This is an extension to the olcPixelGameEngine, which provides
sound generation and wave playing routines.
Special Thanks:
~~~~~~~~~~~~~~~
Slavka - For entire non-windows system back end!
Gorbit99 - Testing, Bug Fixes
Cyberdroid - Testing, Bug Fixes
Dragoneye - Testing
Puol - Testing
License (OLC-3)
~~~~~~~~~~~~~~~
Copyright 2018 OneLoneCoder.com
Copyright 2018 - 2019 OneLoneCoder.com
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
@ -56,28 +64,59 @@
Author
~~~~~~
David Barr, aka javidx9, ©OneLoneCoder 2018
David Barr, aka javidx9, ©OneLoneCoder 2019
*/
#ifndef OLC_PGEX_SOUND
#define OLC_PGEX_SOUND
#ifndef OLC_PGEX_SOUND_H
#define OLC_PGEX_SOUND_H
#include <istream>
#include <cstring>
#include <climits>
#include <algorithm>
#undef min
#undef max
// Choose a default sound backend
#if !defined(USE_ALSA) && !defined(USE_OPENAL) && !defined(USE_WINDOWS)
#ifdef __linux__
#define USE_ALSA
#endif
#ifdef __EMSCRIPTEN__
#define USE_OPENAL
#endif
#ifdef _WIN32
#define USE_WINDOWS
#endif
#endif
#ifdef USE_ALSA
#define ALSA_PCM_NEW_HW_PARAMS_API
#include <alsa/asoundlib.h>
#endif
#ifdef USE_OPENAL
#include <AL/al.h>
#include <AL/alc.h>
#include <queue>
#endif
#pragma pack(push, 1)
typedef struct {
unsigned short wFormatTag;
unsigned short nChannels;
unsigned long nSamplesPerSec;
unsigned long nAvgBytesPerSec;
unsigned short nBlockAlign;
unsigned short wBitsPerSample;
unsigned short cbSize;
uint16_t wFormatTag;
uint16_t nChannels;
uint32_t nSamplesPerSec;
uint32_t nAvgBytesPerSec;
uint16_t nBlockAlign;
uint16_t wBitsPerSample;
uint16_t cbSize;
} OLC_WAVEFORMATEX;
#pragma pack(pop)
namespace olc
{
@ -87,18 +126,18 @@ namespace olc
// A representation of an affine transform, used to rotate, scale, offset & shear space
public:
class AudioSample
{
{
public:
AudioSample();
AudioSample(std::string sWavFile, olc::ResourcePack *pack = nullptr);
olc::rcode LoadFromFile(std::string sWavFile, olc::ResourcePack *pack = nullptr);
public:
OLC_WAVEFORMATEX wavHeader;
float *fSample = nullptr;
long nSamples = 0;
int nChannels = 0;
bool bSampleValid = false;
bool bSampleValid = false;
};
struct sCurrentlyPlayingSample
@ -119,7 +158,7 @@ namespace olc
static void SetUserFilterFunction(std::function<float(int, float, float)> func);
public:
static unsigned int LoadAudioSample(std::string sWavFile, olc::ResourcePack *pack = nullptr);
static int LoadAudioSample(std::string sWavFile, olc::ResourcePack *pack = nullptr);
static void PlaySample(int id, bool bLoop = false);
static void StopSample(int id);
static void StopAll();
@ -127,8 +166,8 @@ namespace olc
private:
#ifdef WIN32 // Windows specific sound management
static void CALLBACK waveOutProc(HWAVEOUT hWaveOut, UINT uMsg, DWORD dwParam1, DWORD dwParam2);
#ifdef USE_WINDOWS // Windows specific sound management
static void CALLBACK waveOutProc(HWAVEOUT hWaveOut, UINT uMsg, DWORD dwParam1, DWORD dwParam2);
static unsigned int m_nSampleRate;
static unsigned int m_nChannels;
static unsigned int m_nBlockCount;
@ -136,26 +175,47 @@ namespace olc
static unsigned int m_nBlockCurrent;
static short* m_pBlockMemory;
static WAVEHDR *m_pWaveHeaders;
static HWAVEOUT m_hwDevice;
static HWAVEOUT m_hwDevice;
static std::atomic<unsigned int> m_nBlockFree;
static std::condition_variable m_cvBlockNotZero;
static std::mutex m_muxBlockNotZero;
#endif
#ifdef USE_ALSA
static snd_pcm_t *m_pPCM;
static unsigned int m_nSampleRate;
static unsigned int m_nChannels;
static unsigned int m_nBlockSamples;
static short* m_pBlockMemory;
#endif
#ifdef USE_OPENAL
static std::queue<ALuint> m_qAvailableBuffers;
static ALuint *m_pBuffers;
static ALuint m_nSource;
static ALCdevice *m_pDevice;
static ALCcontext *m_pContext;
static unsigned int m_nSampleRate;
static unsigned int m_nChannels;
static unsigned int m_nBlockCount;
static unsigned int m_nBlockSamples;
static short* m_pBlockMemory;
#endif
static void AudioThread();
static std::thread m_AudioThread;
static std::atomic<bool> m_bAudioThreadActive;
static std::atomic<float> m_fGlobalTime;
static std::function<float(int, float, float)> funcUserSynth;
static std::function<float(int, float, float)> funcUserFilter;
};
}
#ifdef WIN32
#pragma comment(lib, "winmm.lib")
// Implementation, platform-independent
#ifdef OLC_PGEX_SOUND
#undef OLC_PGEX_SOUND
namespace olc
{
@ -187,20 +247,20 @@ namespace olc
// which are not in the wav file
// Just check if wave format is compatible with olcPGE
if (wavHeader.wBitsPerSample != 16 || wavHeader.nSamplesPerSec != 44100)
if (wavHeader.wBitsPerSample != 16 || wavHeader.nSamplesPerSec != 44100)
return olc::FAIL;
// Search for audio data chunk
long nChunksize = 0;
uint32_t nChunksize = 0;
is.read(dump, sizeof(char) * 4); // Read chunk header
is.read((char*)&nChunksize, sizeof(long)); // Read chunk size
is.read((char*)&nChunksize, sizeof(uint32_t)); // Read chunk size
while (strncmp(dump, "data", 4) != 0)
{
// Not audio data, so just skip it
//std::fseek(f, nChunksize, SEEK_CUR);
is.seekg(nChunksize, std::istream::cur);
is.read(dump, sizeof(char) * 4);
is.read((char*)&nChunksize, sizeof(long));
is.read((char*)&nChunksize, sizeof(uint32_t));
}
// Finally got to data, so read it all in and convert to float samples
@ -221,20 +281,21 @@ namespace olc
{
is.read((char*)&s, sizeof(short));
*pSample = (float)s / (float)(MAXSHORT);
*pSample = (float)s / (float)(SHRT_MAX);
pSample++;
}
}
}
// All done, flag sound as valid
// All done, flag sound as valid
bSampleValid = true;
return olc::OK;
};
if (pack != nullptr)
{
std::istream is(&(pack->GetStreamBuffer(sWavFile)));
olc::ResourcePack::sEntry entry = pack->GetStreamBuffer(sWavFile);
std::istream is(&entry);
return ReadWave(is);
}
else
@ -250,6 +311,131 @@ namespace olc
}
}
// This vector holds all loaded sound samples in memory
std::vector<olc::SOUND::AudioSample> vecAudioSamples;
// This structure represents a sound that is currently playing. It only
// holds the sound ID and where this instance of it is up to for its
// current playback
void SOUND::SetUserSynthFunction(std::function<float(int, float, float)> func)
{
funcUserSynth = func;
}
void SOUND::SetUserFilterFunction(std::function<float(int, float, float)> func)
{
funcUserFilter = func;
}
// Load a 16-bit WAVE file @ 44100Hz ONLY into memory. A sample ID
// number is returned if successful, otherwise -1
int SOUND::LoadAudioSample(std::string sWavFile, olc::ResourcePack *pack)
{
olc::SOUND::AudioSample a(sWavFile, pack);
if (a.bSampleValid)
{
vecAudioSamples.push_back(a);
return (unsigned int)vecAudioSamples.size();
}
else
return -1;
}
// Add sample 'id' to the mixers sounds to play list
void SOUND::PlaySample(int id, bool bLoop)
{
olc::SOUND::sCurrentlyPlayingSample a;
a.nAudioSampleID = id;
a.nSamplePosition = 0;
a.bFinished = false;
a.bFlagForStop = false;
a.bLoop = bLoop;
SOUND::listActiveSamples.push_back(a);
}
void SOUND::StopSample(int id)
{
// Find first occurence of sample id
auto s = std::find_if(listActiveSamples.begin(), listActiveSamples.end(), [&](const olc::SOUND::sCurrentlyPlayingSample &s) { return s.nAudioSampleID == id; });
if (s != listActiveSamples.end())
s->bFlagForStop = true;
}
void SOUND::StopAll()
{
for (auto &s : listActiveSamples)
{
s.bFlagForStop = true;
}
}
float SOUND::GetMixerOutput(int nChannel, float fGlobalTime, float fTimeStep)
{
// Accumulate sample for this channel
float fMixerSample = 0.0f;
for (auto &s : listActiveSamples)
{
if (m_bAudioThreadActive)
{
if (s.bFlagForStop)
{
s.bLoop = false;
s.bFinished = true;
}
else
{
// Calculate sample position
s.nSamplePosition += roundf((float)vecAudioSamples[s.nAudioSampleID - 1].wavHeader.nSamplesPerSec * fTimeStep);
// If sample position is valid add to the mix
if (s.nSamplePosition < vecAudioSamples[s.nAudioSampleID - 1].nSamples)
fMixerSample += vecAudioSamples[s.nAudioSampleID - 1].fSample[(s.nSamplePosition * vecAudioSamples[s.nAudioSampleID - 1].nChannels) + nChannel];
else
{
if (s.bLoop)
{
s.nSamplePosition = 0;
}
else
s.bFinished = true; // Else sound has completed
}
}
}
else
return 0.0f;
}
// If sounds have completed then remove them
listActiveSamples.remove_if([](const sCurrentlyPlayingSample &s) {return s.bFinished; });
// The users application might be generating sound, so grab that if it exists
if (funcUserSynth != nullptr)
fMixerSample += funcUserSynth(nChannel, fGlobalTime, fTimeStep);
// Return the sample via an optional user override to filter the sound
if (funcUserFilter != nullptr)
return funcUserFilter(nChannel, fGlobalTime, fMixerSample);
else
return fMixerSample;
}
std::thread SOUND::m_AudioThread;
std::atomic<bool> SOUND::m_bAudioThreadActive{ false };
std::atomic<float> SOUND::m_fGlobalTime{ 0.0f };
std::list<SOUND::sCurrentlyPlayingSample> SOUND::listActiveSamples;
std::function<float(int, float, float)> SOUND::funcUserSynth = nullptr;
std::function<float(int, float, float)> SOUND::funcUserFilter = nullptr;
}
// Implementation, Windows-specific
#ifdef USE_WINDOWS
#pragma comment(lib, "winmm.lib")
namespace olc
{
bool SOUND::InitialiseAudio(unsigned int nSampleRate, unsigned int nChannels, unsigned int nBlocks, unsigned int nBlockSamples)
{
// Initialise Sound Engine
@ -386,292 +572,321 @@ namespace olc
}
}
// This vector holds all loaded sound samples in memory
std::vector<olc::SOUND::AudioSample> vecAudioSamples;
unsigned int SOUND::m_nSampleRate = 0;
unsigned int SOUND::m_nChannels = 0;
unsigned int SOUND::m_nBlockCount = 0;
unsigned int SOUND::m_nBlockSamples = 0;
unsigned int SOUND::m_nBlockCurrent = 0;
short* SOUND::m_pBlockMemory = nullptr;
WAVEHDR *SOUND::m_pWaveHeaders = nullptr;
HWAVEOUT SOUND::m_hwDevice;
std::atomic<unsigned int> SOUND::m_nBlockFree = 0;
std::condition_variable SOUND::m_cvBlockNotZero;
std::mutex SOUND::m_muxBlockNotZero;
}
// This structure represents a sound that is currently playing. It only
// holds the sound ID and where this instance of it is up to for its
// current playback
void SOUND::SetUserSynthFunction(std::function<float(int, float, float)> func)
{
funcUserSynth = func;
}
#elif defined(USE_ALSA)
void SOUND::SetUserFilterFunction(std::function<float(int, float, float)> func)
namespace olc
{
bool SOUND::InitialiseAudio(unsigned int nSampleRate, unsigned int nChannels, unsigned int nBlocks, unsigned int nBlockSamples)
{
funcUserFilter = func;
}
// Initialise Sound Engine
m_bAudioThreadActive = false;
m_nSampleRate = nSampleRate;
m_nChannels = nChannels;
m_nBlockSamples = nBlockSamples;
m_pBlockMemory = nullptr;
// Load a 16-bit WAVE file @ 44100Hz ONLY into memory. A sample ID
// number is returned if successful, otherwise -1
unsigned int SOUND::LoadAudioSample(std::string sWavFile, olc::ResourcePack *pack)
{
// Open PCM stream
int rc = snd_pcm_open(&m_pPCM, "default", SND_PCM_STREAM_PLAYBACK, 0);
if (rc < 0)
return DestroyAudio();
olc::SOUND::AudioSample a(sWavFile, pack);
if (a.bSampleValid)
{
vecAudioSamples.push_back(a);
return vecAudioSamples.size();
}
else
return -1;
}
// Add sample 'id' to the mixers sounds to play list
void SOUND::PlaySample(int id, bool bLoop)
{
olc::SOUND::sCurrentlyPlayingSample a;
a.nAudioSampleID = id;
a.nSamplePosition = 0;
a.bFinished = false;
a.bFlagForStop = false;
a.bLoop = bLoop;
SOUND::listActiveSamples.push_back(a);
}
// Prepare the parameter structure and set default parameters
snd_pcm_hw_params_t *params;
snd_pcm_hw_params_alloca(&params);
snd_pcm_hw_params_any(m_pPCM, params);
void SOUND::StopSample(int id)
{
// Find first occurence of sample id
auto s = std::find_if(listActiveSamples.begin(), listActiveSamples.end(), [&](const olc::SOUND::sCurrentlyPlayingSample &s) { return s.nAudioSampleID == id; });
if(s != listActiveSamples.end())
s->bFlagForStop = true;
// Set other parameters
snd_pcm_hw_params_set_format(m_pPCM, params, SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params_set_rate(m_pPCM, params, m_nSampleRate, 0);
snd_pcm_hw_params_set_channels(m_pPCM, params, m_nChannels);
snd_pcm_hw_params_set_period_size(m_pPCM, params, m_nBlockSamples, 0);
snd_pcm_hw_params_set_periods(m_pPCM, params, nBlocks, 0);
// Save these parameters
rc = snd_pcm_hw_params(m_pPCM, params);
if (rc < 0)
return DestroyAudio();
listActiveSamples.clear();
// Allocate Wave|Block Memory
m_pBlockMemory = new short[m_nBlockSamples];
if (m_pBlockMemory == nullptr)
return DestroyAudio();
std::fill(m_pBlockMemory, m_pBlockMemory + m_nBlockSamples, 0);
// Unsure if really needed, helped prevent underrun on my setup
snd_pcm_start(m_pPCM);
for (unsigned int i = 0; i < nBlocks; i++)
rc = snd_pcm_writei(m_pPCM, m_pBlockMemory, 512);
snd_pcm_start(m_pPCM);
m_bAudioThreadActive = true;
m_AudioThread = std::thread(&SOUND::AudioThread);
return true;
}
void SOUND::StopAll()
// Stop and clean up audio system
bool SOUND::DestroyAudio()
{
for (auto &s : listActiveSamples)
{
s.bFlagForStop = true;
}
m_bAudioThreadActive = false;
m_AudioThread.join();
snd_pcm_drain(m_pPCM);
snd_pcm_close(m_pPCM);
return false;
}
float SOUND::GetMixerOutput(int nChannel, float fGlobalTime, float fTimeStep)
// Audio thread. This loop responds to requests from the soundcard to fill 'blocks'
// with audio data. If no requests are available it goes dormant until the sound
// card is ready for more data. The block is fille by the "user" in some manner
// and then issued to the soundcard.
void SOUND::AudioThread()
{
// Accumulate sample for this channel
float fMixerSample = 0.0f;
m_fGlobalTime = 0.0f;
static float fTimeStep = 1.0f / (float)m_nSampleRate;
for (auto &s : listActiveSamples)
// Goofy hack to get maximum integer for a type at run-time
short nMaxSample = (short)pow(2, (sizeof(short) * 8) - 1) - 1;
float fMaxSample = (float)nMaxSample;
short nPreviousSample = 0;
while (m_bAudioThreadActive)
{
if (m_bAudioThreadActive)
short nNewSample = 0;
auto clip = [](float fSample, float fMax)
{
if (s.bFlagForStop)
{
s.bLoop = false;
s.bFinished = true;
}
if (fSample >= 0.0)
return fmin(fSample, fMax);
else
return fmax(fSample, -fMax);
};
for (unsigned int n = 0; n < m_nBlockSamples; n += m_nChannels)
{
// User Process
for (unsigned int c = 0; c < m_nChannels; c++)
{
// Calculate sample position
s.nSamplePosition += (long)((float)vecAudioSamples[s.nAudioSampleID - 1].wavHeader.nSamplesPerSec * fTimeStep);
nNewSample = (short)(clip(GetMixerOutput(c, m_fGlobalTime, fTimeStep), 1.0) * fMaxSample);
m_pBlockMemory[n + c] = nNewSample;
nPreviousSample = nNewSample;
}
// If sample position is valid add to the mix
if (s.nSamplePosition < vecAudioSamples[s.nAudioSampleID - 1].nSamples)
fMixerSample += vecAudioSamples[s.nAudioSampleID - 1].fSample[(s.nSamplePosition * vecAudioSamples[s.nAudioSampleID - 1].nChannels) + nChannel];
else
{
if (s.bLoop)
{
s.nSamplePosition = 0;
}
else
s.bFinished = true; // Else sound has completed
}
m_fGlobalTime = m_fGlobalTime + fTimeStep;
}
// Send block to sound device
snd_pcm_uframes_t nLeft = m_nBlockSamples;
short *pBlockPos = m_pBlockMemory;
while (nLeft > 0)
{
int rc = snd_pcm_writei(m_pPCM, pBlockPos, nLeft);
if (rc > 0)
{
pBlockPos += rc * m_nChannels;
nLeft -= rc;
}
if (rc == -EAGAIN) continue;
if (rc == -EPIPE) // an underrun occured, prepare the device for more data
snd_pcm_prepare(m_pPCM);
}
else
return 0.0f;
}
// If sounds have completed then remove them
listActiveSamples.remove_if([](const sCurrentlyPlayingSample &s) {return s.bFinished; });
// The users application might be generating sound, so grab that if it exists
if(funcUserSynth != nullptr)
fMixerSample += funcUserSynth(nChannel, fGlobalTime, fTimeStep);
// Return the sample via an optional user override to filter the sound
if (funcUserFilter != nullptr)
return funcUserFilter(nChannel, fGlobalTime, fMixerSample);
else
return fMixerSample;
}
snd_pcm_t* SOUND::m_pPCM = nullptr;
unsigned int SOUND::m_nSampleRate = 0;
unsigned int SOUND::m_nChannels = 0;
unsigned int SOUND::m_nBlockCount = 0;
unsigned int SOUND::m_nBlockSamples = 0;
unsigned int SOUND::m_nBlockCurrent = 0;
short* SOUND::m_pBlockMemory = nullptr;
WAVEHDR *SOUND::m_pWaveHeaders = nullptr;
HWAVEOUT SOUND::m_hwDevice;
std::thread SOUND::m_AudioThread;
std::atomic<bool> SOUND::m_bAudioThreadActive = false;
std::atomic<unsigned int> SOUND::m_nBlockFree = 0;
std::condition_variable SOUND::m_cvBlockNotZero;
std::mutex SOUND::m_muxBlockNotZero;
std::atomic<float> SOUND::m_fGlobalTime = 0.0f;
std::list<SOUND::sCurrentlyPlayingSample> SOUND::listActiveSamples;
std::function<float(int, float, float)> SOUND::funcUserSynth = nullptr;
std::function<float(int, float, float)> SOUND::funcUserFilter = nullptr;
}
#else // Non Windows
#elif defined(USE_OPENAL)
namespace olc
{
SOUND::AudioSample::AudioSample()
{}
SOUND::AudioSample::AudioSample(std::string sWavFile, olc::ResourcePack *pack)
bool SOUND::InitialiseAudio(unsigned int nSampleRate, unsigned int nChannels, unsigned int nBlocks, unsigned int nBlockSamples)
{
LoadFromFile(sWavFile, pack);
}
// Initialise Sound Engine
m_bAudioThreadActive = false;
m_nSampleRate = nSampleRate;
m_nChannels = nChannels;
m_nBlockCount = nBlocks;
m_nBlockSamples = nBlockSamples;
m_pBlockMemory = nullptr;
olc::rcode SOUND::AudioSample::LoadFromFile(std::string sWavFile, olc::ResourcePack *pack)
{
return olc::OK;
}
// Open the device and create the context
m_pDevice = alcOpenDevice(NULL);
if (m_pDevice)
{
m_pContext = alcCreateContext(m_pDevice, NULL);
alcMakeContextCurrent(m_pContext);
}
else
return DestroyAudio();
bool SOUND::InitialiseAudio(unsigned int nSampleRate, unsigned int nChannels, unsigned int nBlocks, unsigned int nBlockSamples)
{
// Allocate memory for sound data
alGetError();
m_pBuffers = new ALuint[m_nBlockCount];
alGenBuffers(m_nBlockCount, m_pBuffers);
alGenSources(1, &m_nSource);
for (unsigned int i = 0; i < m_nBlockCount; i++)
m_qAvailableBuffers.push(m_pBuffers[i]);
listActiveSamples.clear();
// Allocate Wave|Block Memory
m_pBlockMemory = new short[m_nBlockSamples];
if (m_pBlockMemory == nullptr)
return DestroyAudio();
std::fill(m_pBlockMemory, m_pBlockMemory + m_nBlockSamples, 0);
m_bAudioThreadActive = true;
m_AudioThread = std::thread(&SOUND::AudioThread);
return true;
}
// Stop and clean up audio system
bool SOUND::DestroyAudio()
{
m_bAudioThreadActive = false;
m_AudioThread.join();
alDeleteBuffers(m_nBlockCount, m_pBuffers);
delete[] m_pBuffers;
alDeleteSources(1, &m_nSource);
alcMakeContextCurrent(NULL);
alcDestroyContext(m_pContext);
alcCloseDevice(m_pDevice);
return false;
}
// Audio thread. This loop responds to requests from the soundcard to fill 'blocks'
// with audio data. If no requests are available it goes dormant until the sound
// card is ready for more data. The block is fille by the "user" in some manner
// and then issued to the soundcard.
void SOUND::AudioThread()
{
}
// This vector holds all loaded sound samples in memory
std::vector<olc::SOUND::AudioSample> vecAudioSamples;
// This structure represents a sound that is currently playing. It only
// holds the sound ID and where this instance of it is up to for its
// current playback
m_fGlobalTime = 0.0f;
static float fTimeStep = 1.0f / (float)m_nSampleRate;
void SOUND::SetUserSynthFunction(std::function<float(int, float, float)> func)
{
funcUserSynth = func;
}
// Goofy hack to get maximum integer for a type at run-time
short nMaxSample = (short)pow(2, (sizeof(short) * 8) - 1) - 1;
float fMaxSample = (float)nMaxSample;
short nPreviousSample = 0;
void SOUND::SetUserFilterFunction(std::function<float(int, float, float)> func)
{
funcUserFilter = func;
}
std::vector<ALuint> vProcessed;
// Load a 16-bit WAVE file @ 44100Hz ONLY into memory. A sample ID
// number is returned if successful, otherwise -1
unsigned int SOUND::LoadAudioSample(std::string sWavFile, olc::ResourcePack *pack)
{
olc::SOUND::AudioSample a(sWavFile, pack);
if (a.bSampleValid)
while (m_bAudioThreadActive)
{
vecAudioSamples.push_back(a);
return vecAudioSamples.size();
}
else
return -1;
}
ALint nState, nProcessed;
alGetSourcei(m_nSource, AL_SOURCE_STATE, &nState);
alGetSourcei(m_nSource, AL_BUFFERS_PROCESSED, &nProcessed);
// Add sample 'id' to the mixers sounds to play list
void SOUND::PlaySample(int id, bool bLoop)
{
olc::SOUND::sCurrentlyPlayingSample a;
a.nAudioSampleID = id;
a.nSamplePosition = 0;
a.bFinished = false;
a.bFlagForStop = false;
a.bLoop = bLoop;
SOUND::listActiveSamples.push_back(a);
}
// Add processed buffers to our queue
vProcessed.resize(nProcessed);
alSourceUnqueueBuffers(m_nSource, nProcessed, vProcessed.data());
for (ALint nBuf : vProcessed) m_qAvailableBuffers.push(nBuf);
void SOUND::StopSample(int id)
{
// Find first occurence of sample id
auto s = std::find_if(listActiveSamples.begin(), listActiveSamples.end(), [&](const olc::SOUND::sCurrentlyPlayingSample &s) { return s.nAudioSampleID == id; });
if (s != listActiveSamples.end())
s->bFlagForStop = true;
}
// Wait until there is a free buffer (ewww)
if (m_qAvailableBuffers.empty()) continue;
void SOUND::StopAll()
{
for (auto &s : listActiveSamples)
{
s.bFlagForStop = true;
}
}
short nNewSample = 0;
float SOUND::GetMixerOutput(int nChannel, float fGlobalTime, float fTimeStep)
{
// Accumulate sample for this channel
float fMixerSample = 0.0f;
auto clip = [](float fSample, float fMax)
{
if (fSample >= 0.0)
return fmin(fSample, fMax);
else
return fmax(fSample, -fMax);
};
for (auto &s : listActiveSamples)
{
if (m_bAudioThreadActive)
for (unsigned int n = 0; n < m_nBlockSamples; n += m_nChannels)
{
if (s.bFlagForStop)
// User Process
for (unsigned int c = 0; c < m_nChannels; c++)
{
s.bLoop = false;
s.bFinished = true;
nNewSample = (short)(clip(GetMixerOutput(c, m_fGlobalTime, fTimeStep), 1.0) * fMaxSample);
m_pBlockMemory[n + c] = nNewSample;
nPreviousSample = nNewSample;
}
else
{
// Calculate sample position
s.nSamplePosition += (long)((float)vecAudioSamples[s.nAudioSampleID - 1].wavHeader.nSamplesPerSec * fTimeStep);
// If sample position is valid add to the mix
if (s.nSamplePosition < vecAudioSamples[s.nAudioSampleID - 1].nSamples)
fMixerSample += vecAudioSamples[s.nAudioSampleID - 1].fSample[(s.nSamplePosition * vecAudioSamples[s.nAudioSampleID - 1].nChannels) + nChannel];
else
{
if (s.bLoop)
{
s.nSamplePosition = 0;
}
else
s.bFinished = true; // Else sound has completed
}
}
m_fGlobalTime = m_fGlobalTime + fTimeStep;
}
else
return 0.0f;
// Fill OpenAL data buffer
alBufferData(
m_qAvailableBuffers.front(),
m_nChannels == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16,
m_pBlockMemory,
2 * m_nBlockSamples,
m_nSampleRate
);
// Add it to the OpenAL queue
alSourceQueueBuffers(m_nSource, 1, &m_qAvailableBuffers.front());
// Remove it from ours
m_qAvailableBuffers.pop();
// If it's not playing for some reason, change that
if (nState != AL_PLAYING)
alSourcePlay(m_nSource);
}
}
// If sounds have completed then remove them
listActiveSamples.remove_if([](const sCurrentlyPlayingSample &s) {return s.bFinished; });
std::queue<ALuint> SOUND::m_qAvailableBuffers;
ALuint *SOUND::m_pBuffers = nullptr;
ALuint SOUND::m_nSource = 0;
ALCdevice *SOUND::m_pDevice = nullptr;
ALCcontext *SOUND::m_pContext = nullptr;
unsigned int SOUND::m_nSampleRate = 0;
unsigned int SOUND::m_nChannels = 0;
unsigned int SOUND::m_nBlockCount = 0;
unsigned int SOUND::m_nBlockSamples = 0;
short* SOUND::m_pBlockMemory = nullptr;
}
// The users application might be generating sound, so grab that if it exists
if (funcUserSynth != nullptr)
fMixerSample += funcUserSynth(nChannel, fGlobalTime, fTimeStep);
#else // Some other platform
// Return the sample via an optional user override to filter the sound
if (funcUserFilter != nullptr)
return funcUserFilter(nChannel, fGlobalTime, fMixerSample);
else
return fMixerSample;
namespace olc
{
bool SOUND::InitialiseAudio(unsigned int nSampleRate, unsigned int nChannels, unsigned int nBlocks, unsigned int nBlockSamples)
{
return true;
}
std::thread SOUND::m_AudioThread;
std::atomic<bool> SOUND::m_bAudioThreadActive{ false };
std::atomic<float> SOUND::m_fGlobalTime{ 0.0f };
std::list<SOUND::sCurrentlyPlayingSample> SOUND::listActiveSamples;
std::function<float(int, float, float)> SOUND::funcUserSynth = nullptr;
std::function<float(int, float, float)> SOUND::funcUserFilter = nullptr;
}
#endif
// Stop and clean up audio system
bool SOUND::DestroyAudio()
{
return false;
}
#endif
// Audio thread. This loop responds to requests from the soundcard to fill 'blocks'
// with audio data. If no requests are available it goes dormant until the sound
// card is ready for more data. The block is fille by the "user" in some manner
// and then issued to the soundcard.
void SOUND::AudioThread()
{ }
}
#endif
#endif
#endif // OLC_PGEX_SOUND
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