Debuggers/pge/src/olcPGEX_Sound.cpp

#include "olcPGEX_Sound.h"

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
			uint32_t nChunksize = 0;
			is.read(dump, sizeof(char) * 4); // Read chunk header
			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(uint32_t));
			}

			// 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)(SHRT_MAX);
						pSample++;
					}
				}
			}

			// All done, flag sound as valid
			bSampleValid = true;
			return olc::OK;
		};

		if (pack != nullptr)
		{
			olc::ResourcePack::sEntry entry = pack->GetStreamBuffer(sWavFile);
			std::istream is(&entry);
			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;
		}
	}

	// 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
#ifndef __MINGW32__
#pragma comment(lib, "winmm.lib")
#endif

namespace olc
{
	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_PTR dwInstance, 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;

		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;
				}

				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;
		}
	}

	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;
}

#elif defined(USE_ALSA)

namespace olc
{
	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_nBlockSamples = nBlockSamples;
		m_pBlockMemory = nullptr;

		// Open PCM stream
		int rc = snd_pcm_open(&m_pPCM, "default", SND_PCM_STREAM_PLAYBACK, 0);
		if (rc < 0)
			return DestroyAudio();


		// 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);

		// 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;
	}

	// Stop and clean up audio system
	bool SOUND::DestroyAudio()
	{
		m_bAudioThreadActive = false;
		m_AudioThread.join();
		snd_pcm_drain(m_pPCM);
		snd_pcm_close(m_pPCM);
		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()
	{
		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;

		while (m_bAudioThreadActive)
		{
			short nNewSample = 0;

			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[n + c] = nNewSample;
				}

				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);
			}
		}
	}

	snd_pcm_t* SOUND::m_pPCM = nullptr;
	unsigned int SOUND::m_nSampleRate = 0;
	unsigned int SOUND::m_nChannels = 0;
	unsigned int SOUND::m_nBlockSamples = 0;
	short* SOUND::m_pBlockMemory = nullptr;
}

#elif defined(USE_OPENAL)

namespace olc
{
	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_pBlockMemory = nullptr;

		// 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();

		// 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()
	{
		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;

		std::vector<ALuint> vProcessed;

		while (m_bAudioThreadActive)
		{
			ALint nState, nProcessed;
			alGetSourcei(m_nSource, AL_SOURCE_STATE, &nState);
			alGetSourcei(m_nSource, AL_BUFFERS_PROCESSED, &nProcessed);

			// Add processed buffers to our queue
			vProcessed.resize(nProcessed);
			alSourceUnqueueBuffers(m_nSource, nProcessed, vProcessed.data());
			for (ALint nBuf : vProcessed) m_qAvailableBuffers.push(nBuf);

			// Wait until there is a free buffer (ewww)
			if (m_qAvailableBuffers.empty()) continue;

			short nNewSample = 0;

			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[n + c] = nNewSample;
					nPreviousSample = nNewSample;
				}

				m_fGlobalTime = m_fGlobalTime + fTimeStep;
			}

			// 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);
		}
	}

	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;
}

#else // Some other platform

namespace olc
{
	bool SOUND::InitialiseAudio(unsigned int nSampleRate, unsigned int nChannels, unsigned int nBlocks, unsigned int nBlockSamples)
	{
		return true;
	}

	// Stop and clean up audio system
	bool SOUND::DestroyAudio()
	{
		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()
	{	}
}

#endif