Added Sound Extension

pull/113/head
Javidx9 6 years ago committed by GitHub
parent 2ebb88a942
commit 786b33b950
  1. 269
      OneLoneCoder_PGE_SoundTest.cpp
  2. BIN
      SampleA.wav
  3. BIN
      SampleB.wav
  4. BIN
      SampleC.wav
  5. 513
      olcPGEX_Sound.h

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/*
Simple example code for olcPGEX_Sound.h - Mind your speakers!
You will need SampleA.wav, SampleB.wav and SampleC.wav for this demo.
License (OLC-3)
~~~~~~~~~~~~~~~
Copyright 2018 OneLoneCoder.com
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
1. Redistributions or derivations of source code must retain the above
copyright notice, this list of conditions and the following disclaimer.
2. Redistributions or derivative works in binary form must reproduce
the above copyright notice. This list of conditions and the following
disclaimer must be reproduced in the documentation and/or other
materials provided with the distribution.
3. Neither the name of the copyright holder nor the names of its
contributors may be used to endorse or promote products derived
from this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
HOLDER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
Links
~~~~~
YouTube: https://www.youtube.com/javidx9
Discord: https://discord.gg/WhwHUMV
Twitter: https://www.twitter.com/javidx9
Twitch: https://www.twitch.tv/javidx9
GitHub: https://www.github.com/onelonecoder
Homepage: https://www.onelonecoder.com
Patreon: https://www.patreon.com/javidx9
Author
~~~~~~
David Barr, aka javidx9, ©OneLoneCoder 2018
*/
#define OLC_PGE_APPLICATION
#include "olcPixelGameEngine.h"
#include "olcPGEX_Sound.h"
#include <list>
class SoundTest : public olc::PixelGameEngine
{
public:
SoundTest()
{
sAppName = "Sound Test";
}
private:
int sndSampleA;
int sndSampleB;
int sndSampleC;
bool bToggle = false;
static bool bSynthPlaying;
static float fSynthFrequency;
static float fFilterVolume;
const olc::Key keys[12] = { olc::Key::Z, olc::Key::S, olc::Key::X, olc::Key::D, olc::Key::C,
olc::Key::V, olc::Key::G, olc::Key::B, olc::Key::H, olc::Key::N, olc::Key::J, olc::Key::M};
static float fPreviousSamples[128];
static int nSamplePos;
private:
// This is an optional function that allows the user to generate or synthesize sounds
// in a custom way, it is fed into the output mixer bu the extension
static float MyCustomSynthFunction(int nChannel, float fGlobalTime, float fTimeStep)
{
// Just generate a sine wave of the appropriate frequency
if (bSynthPlaying)
return sin(fSynthFrequency * 2.0f * 3.14159f * fGlobalTime);
else
return 0.0f;
}
// This is an optional function that allows the user to filter the output from
// the internal mixer of the extension. Here you could add effects or just
// control volume. I also like to use it to extract information about
// the currently playing output waveform
static float MyCustomFilterFunction(int nChannel, float fGlobalTime, float fSample)
{
// Fundamentally just control volume
float fOutput = fSample * fFilterVolume;
// But also add sample to list of previous samples for visualisation
fPreviousSamples[nSamplePos] = fOutput;
nSamplePos++;
nSamplePos %= 128;
return fOutput;
}
bool OnUserCreate()
{
olc::SOUND::InitialiseAudio();
sndSampleA = olc::SOUND::LoadAudioSample("SampleA.wav");
sndSampleB = olc::SOUND::LoadAudioSample("SampleB.wav");
sndSampleC = olc::SOUND::LoadAudioSample("SampleC.wav");
// Give the sound engine a hook to a custom generation function
olc::SOUND::SetUserSynthFunction(MyCustomSynthFunction);
// Give the sound engine a hook to a custom filtering function
olc::SOUND::SetUserFilterFunction(MyCustomFilterFunction);
return true;
}
bool OnUserUpdate(float fElapsedTime)
{
//olc::SOUND::PlaySample(sndTest);
auto PointInRect = [&](int x, int y, int rx, int ry, int rw, int rh)
{
return x >= rx && x < (rx + rw) && y >= ry && y < (ry + rh);
};
int nMouseX = GetMouseX();
int nMouseY = GetMouseY();
if(GetMouse(0).bPressed && PointInRect(nMouseX, nMouseY, 16, 16, 128, 24))
olc::SOUND::PlaySample(sndSampleA); // Plays the sample once
if (GetMouse(0).bPressed && PointInRect(nMouseX, nMouseY, 16, 48, 128, 24))
olc::SOUND::PlaySample(sndSampleB);
if (GetMouse(0).bPressed && PointInRect(nMouseX, nMouseY, 16, 80, 128, 24))
{
bToggle = !bToggle;
if (bToggle)
{
olc::SOUND::PlaySample(sndSampleC, true); // Plays the sample in looping mode
}
else
{
olc::SOUND::StopSample(sndSampleC);
}
}
if (GetMouse(0).bHeld && PointInRect(nMouseX, nMouseY, 160, 16, 90, 24))
fFilterVolume += 2.0f * fElapsedTime;
if (GetMouse(0).bHeld && PointInRect(nMouseX, nMouseY, 160, 48, 90, 24))
fFilterVolume -= 2.0f * fElapsedTime;
if (fFilterVolume < 0.0f) fFilterVolume = 0.0f;
if (fFilterVolume > 1.0f) fFilterVolume = 1.0f;
// Detect keyboard - very simple synthesizer
if (IsFocused())
{
bool bKeyIsPressed = false;
float fFrequency = 0.0f;
for (int i = 0; i < 12; i++)
{
if (GetKey(keys[i]).bHeld)
{
bKeyIsPressed = true;
float fOctaveBaseFrequency = 220.0f;
float f12thRootOf2 = pow(2.0f, 1.0f / 12.0f);
fFrequency = fOctaveBaseFrequency * powf(f12thRootOf2, (float)i);
}
}
fSynthFrequency = fFrequency;
bSynthPlaying = bKeyIsPressed;
}
// Draw Buttons
Clear(olc::BLUE);
DrawRect(16, 16, 128, 24);
DrawString(20, 20, "Play Sample A");
DrawRect(16, 48, 128, 24);
DrawString(20, 52, "Play Sample B");
DrawRect(16, 80, 128, 24);
DrawString(20, 84, (bToggle ? "Stop Sample C" : "Loop Sample C"));
DrawRect(160, 16, 90, 24);
DrawString(164, 20, "Volume +");
DrawRect(160, 48, 90, 24);
DrawString(164, 52, "Volume -");
DrawString(164, 80, "Volume: " + std::to_string((int)(fFilterVolume * 10.0f)));
// Draw Keyboard
// White Keys
for (int i = 0; i < 7; i++)
{
FillRect(i * 16 + 8, 160, 16, 64);
DrawRect(i * 16 + 8, 160, 16, 64, olc::BLACK);
DrawString(i * 16 + 12, 212, std::string(1, "ZXCVBNM"[i]), olc::BLACK);
}
// Black Keys
for (int i = 0; i < 6; i++)
{
if (i != 2)
{
FillRect(i * 16 + 18, 160, 12, 32, olc::BLACK);
DrawString(i * 16 + 20, 180, std::string(1, "SDFGHJ"[i]), olc::WHITE);
}
}
// Draw visualisation
int nStartPos = (nSamplePos + 127) % 128;
for (int i = 127; i >= 0; i--)
{
float fSample = fPreviousSamples[(nSamplePos + i) % 128];
DrawLine(124 + i, 210, 124 + i, 210 + (int)(fSample * 20.0f), olc::RED);
}
return true;
}
// Note we must shut down the sound system too!!
bool OnUserDestroy()
{
olc::SOUND::DestroyAudio();
return true;
}
};
bool SoundTest::bSynthPlaying = false;
float SoundTest::fSynthFrequency = 0.0f;
float SoundTest::fFilterVolume = 1.0f;
int SoundTest::nSamplePos = 0;
float SoundTest::fPreviousSamples[128];
int main()
{
SoundTest demo;
if(demo.Construct(256, 240, 4, 4))
demo.Start();
return 0;
}

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/*
olcPGEX_Sound.h
+-------------------------------------------------------------+
| OneLoneCoder Pixel Game Engine Extension |
| Sound - v0.2 |
+-------------------------------------------------------------+
What is this?
~~~~~~~~~~~~~
This is an extension to the olcPixelGameEngine, which provides
sound generation and wave playing routines.
License (OLC-3)
~~~~~~~~~~~~~~~
Copyright 2018 OneLoneCoder.com
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
1. Redistributions or derivations of source code must retain the above
copyright notice, this list of conditions and the following disclaimer.
2. Redistributions or derivative works in binary form must reproduce
the above copyright notice. This list of conditions and the following
disclaimer must be reproduced in the documentation and/or other
materials provided with the distribution.
3. Neither the name of the copyright holder nor the names of its
contributors may be used to endorse or promote products derived
from this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
HOLDER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
Links
~~~~~
YouTube: https://www.youtube.com/javidx9
Discord: https://discord.gg/WhwHUMV
Twitter: https://www.twitter.com/javidx9
Twitch: https://www.twitch.tv/javidx9
GitHub: https://www.github.com/onelonecoder
Homepage: https://www.onelonecoder.com
Patreon: https://www.patreon.com/javidx9
Author
~~~~~~
David Barr, aka javidx9, ©OneLoneCoder 2018
*/
#ifndef OLC_PGEX_SOUND
#define OLC_PGEX_SOUND
#include <istream>
#include <algorithm>
#undef min
#undef max
namespace olc
{
// Container class for Advanced 2D Drawing functions
class SOUND : public olc::PGEX
{
// 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:
WAVEFORMATEX wavHeader;
float *fSample = nullptr;
long nSamples = 0;
int nChannels = 0;
bool bSampleValid = false;
};
struct sCurrentlyPlayingSample
{
int nAudioSampleID = 0;
long nSamplePosition = 0;
bool bFinished = false;
bool bLoop = false;
bool bFlagForStop = false;
};
static std::list<sCurrentlyPlayingSample> listActiveSamples;
public:
static bool InitialiseAudio(unsigned int nSampleRate = 44100, unsigned int nChannels = 1, unsigned int nBlocks = 8, unsigned int nBlockSamples = 512);
static bool DestroyAudio();
static void SetUserSynthFunction(std::function<float(int, float, float)> func);
static void SetUserFilterFunction(std::function<float(int, float, float)> func);
public:
static unsigned 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();
static float GetMixerOutput(int nChannel, float fGlobalTime, float fTimeStep);
#ifdef WIN32
private:
static void CALLBACK waveOutProc(HWAVEOUT hWaveOut, UINT uMsg, DWORD dwParam1, DWORD dwParam2);
static void AudioThread();
static unsigned int m_nSampleRate;
static unsigned int m_nChannels;
static unsigned int m_nBlockCount;
static unsigned int m_nBlockSamples;
static unsigned int m_nBlockCurrent;
static short* m_pBlockMemory;
static WAVEHDR *m_pWaveHeaders;
static HWAVEOUT m_hwDevice;
static std::thread m_AudioThread;
static std::atomic<bool> m_bAudioThreadActive;
static std::atomic<unsigned int> m_nBlockFree;
static std::condition_variable m_cvBlockNotZero;
static std::mutex m_muxBlockNotZero;
static std::atomic<float> m_fGlobalTime;
static std::function<float(int, float, float)> funcUserSynth;
static std::function<float(int, float, float)> funcUserFilter;
#endif
};
}
#ifdef WIN32
#pragma comment(lib, "winmm.lib")
namespace olc
{
SOUND::AudioSample::AudioSample()
{
}
SOUND::AudioSample::AudioSample(std::string sWavFile, olc::ResourcePack *pack)
{
LoadFromFile(sWavFile, pack);
}
olc::rcode SOUND::AudioSample::LoadFromFile(std::string sWavFile, olc::ResourcePack *pack)
{
auto ReadWave = [&](std::istream &is)
{
char dump[4];
is.read(dump, sizeof(char) * 4); // Read "RIFF"
if (strncmp(dump, "RIFF", 4) != 0) return olc::FAIL;
is.read(dump, sizeof(char) * 4); // Not Interested
is.read(dump, sizeof(char) * 4); // Read "WAVE"
if (strncmp(dump, "WAVE", 4) != 0) return olc::FAIL;
// Read Wave description chunk
is.read(dump, sizeof(char) * 4); // Read "fmt "
unsigned int nHeaderSize = 0;
is.read((char*)&nHeaderSize, sizeof(unsigned int)); // Not Interested
is.read((char*)&wavHeader, nHeaderSize);// sizeof(WAVEFORMATEX)); // Read Wave Format Structure chunk
// Note the -2, because the structure has 2 bytes to indicate its own size
// which are not in the wav file
// Just check if wave format is compatible with olcPGE
if (wavHeader.wBitsPerSample != 16 || wavHeader.nSamplesPerSec != 44100)
return olc::FAIL;
// Search for audio data chunk
long nChunksize = 0;
is.read(dump, sizeof(char) * 4); // Read chunk header
is.read((char*)&nChunksize, sizeof(long)); // 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));
}
// Finally got to data, so read it all in and convert to float samples
nSamples = nChunksize / (wavHeader.nChannels * (wavHeader.wBitsPerSample >> 3));
nChannels = wavHeader.nChannels;
// Create floating point buffer to hold audio sample
fSample = new float[nSamples * nChannels];
float *pSample = fSample;
// Read in audio data and normalise
for (long i = 0; i < nSamples; i++)
{
for (int c = 0; c < nChannels; c++)
{
short s = 0;
if (!is.eof())
{
is.read((char*)&s, sizeof(short));
*pSample = (float)s / (float)(MAXSHORT);
pSample++;
}
}
}
// All done, flag sound as valid
bSampleValid = true;
return olc::OK;
};
if (pack != nullptr)
{
std::istream is(&(pack->GetStreamBuffer(sWavFile)));
return ReadWave(is);
}
else
{
// Read from file
std::ifstream ifs(sWavFile, std::ifstream::binary);
if (ifs.is_open())
{
return ReadWave(ifs);
}
else
return olc::FAIL;
}
}
bool SOUND::InitialiseAudio(unsigned int nSampleRate, unsigned int nChannels, unsigned int nBlocks, unsigned int nBlockSamples)
{
// Initialise Sound Engine
m_bAudioThreadActive = false;
m_nSampleRate = nSampleRate;
m_nChannels = nChannels;
m_nBlockCount = nBlocks;
m_nBlockSamples = nBlockSamples;
m_nBlockFree = m_nBlockCount;
m_nBlockCurrent = 0;
m_pBlockMemory = nullptr;
m_pWaveHeaders = nullptr;
// Device is available
WAVEFORMATEX waveFormat;
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
waveFormat.nSamplesPerSec = m_nSampleRate;
waveFormat.wBitsPerSample = sizeof(short) * 8;
waveFormat.nChannels = m_nChannels;
waveFormat.nBlockAlign = (waveFormat.wBitsPerSample / 8) * waveFormat.nChannels;
waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
waveFormat.cbSize = 0;
listActiveSamples.clear();
// Open Device if valid
if (waveOutOpen(&m_hwDevice, WAVE_MAPPER, &waveFormat, (DWORD_PTR)SOUND::waveOutProc, (DWORD_PTR)0, CALLBACK_FUNCTION) != S_OK)
return DestroyAudio();
// Allocate Wave|Block Memory
m_pBlockMemory = new short[m_nBlockCount * m_nBlockSamples];
if (m_pBlockMemory == nullptr)
return DestroyAudio();
ZeroMemory(m_pBlockMemory, sizeof(short) * m_nBlockCount * m_nBlockSamples);
m_pWaveHeaders = new WAVEHDR[m_nBlockCount];
if (m_pWaveHeaders == nullptr)
return DestroyAudio();
ZeroMemory(m_pWaveHeaders, sizeof(WAVEHDR) * m_nBlockCount);
// Link headers to block memory
for (unsigned int n = 0; n < m_nBlockCount; n++)
{
m_pWaveHeaders[n].dwBufferLength = m_nBlockSamples * sizeof(short);
m_pWaveHeaders[n].lpData = (LPSTR)(m_pBlockMemory + (n * m_nBlockSamples));
}
m_bAudioThreadActive = true;
m_AudioThread = std::thread(&SOUND::AudioThread);
// Start the ball rolling with the sound delivery thread
std::unique_lock<std::mutex> lm(m_muxBlockNotZero);
m_cvBlockNotZero.notify_one();
return true;
}
// Stop and clean up audio system
bool SOUND::DestroyAudio()
{
m_bAudioThreadActive = false;
m_AudioThread.join();
return false;
}
// Handler for soundcard request for more data
void CALLBACK SOUND::waveOutProc(HWAVEOUT hWaveOut, UINT uMsg, DWORD dwParam1, DWORD dwParam2)
{
if (uMsg != WOM_DONE) return;
m_nBlockFree++;
std::unique_lock<std::mutex> lm(m_muxBlockNotZero);
m_cvBlockNotZero.notify_one();
}
// 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()
{
m_fGlobalTime = 0.0f;
static float fTimeStep = 1.0f / (float)m_nSampleRate;
// 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)
{
// Wait for block to become available
if (m_nBlockFree == 0)
{
std::unique_lock<std::mutex> lm(m_muxBlockNotZero);
while (m_nBlockFree == 0) // sometimes, Windows signals incorrectly
m_cvBlockNotZero.wait(lm);
}
// Block is here, so use it
m_nBlockFree--;
// Prepare block for processing
if (m_pWaveHeaders[m_nBlockCurrent].dwFlags & WHDR_PREPARED)
waveOutUnprepareHeader(m_hwDevice, &m_pWaveHeaders[m_nBlockCurrent], sizeof(WAVEHDR));
short nNewSample = 0;
int nCurrentBlock = m_nBlockCurrent * m_nBlockSamples;
auto clip = [](float fSample, float fMax)
{
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++)
{
nNewSample = (short)(clip(GetMixerOutput(c, m_fGlobalTime, fTimeStep), 1.0) * fMaxSample);
m_pBlockMemory[nCurrentBlock + n + c] = nNewSample;
nPreviousSample = nNewSample;
}
m_fGlobalTime = m_fGlobalTime + fTimeStep;
}
// Send block to sound device
waveOutPrepareHeader(m_hwDevice, &m_pWaveHeaders[m_nBlockCurrent], sizeof(WAVEHDR));
waveOutWrite(m_hwDevice, &m_pWaveHeaders[m_nBlockCurrent], sizeof(WAVEHDR));
m_nBlockCurrent++;
m_nBlockCurrent %= m_nBlockCount;
}
}
// 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
unsigned int SOUND::LoadAudioSample(std::string sWavFile, olc::ResourcePack *pack)
{
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);
}
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 += (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
}
}
}
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;
}
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;
}
#endif
// Currently no Linux implementation so just go blank :(
#endif
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