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+/*
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+ olcPGEX_Sound.h
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+
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+ +-------------------------------------------------------------+
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+ | OneLoneCoder Pixel Game Engine Extension |
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+ | Sound - v0.3 |
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+ +-------------------------------------------------------------+
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+
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+ What is this?
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+ ~~~~~~~~~~~~~
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+ This is an extension to the olcPixelGameEngine, which provides
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+ sound generation and wave playing routines.
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+
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+ Special Thanks:
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+ ~~~~~~~~~~~~~~~
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+ Slavka - For entire non-windows system back end!
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+ Gorbit99 - Testing, Bug Fixes
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+ Cyberdroid - Testing, Bug Fixes
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+ Dragoneye - Testing
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+ Puol - Testing
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+
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+ License (OLC-3)
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+ ~~~~~~~~~~~~~~~
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+
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+ Copyright 2018 - 2019 OneLoneCoder.com
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+
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+ Redistribution and use in source and binary forms, with or without
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+ modification, are permitted provided that the following conditions
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+ are met:
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+
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+ 1. Redistributions or derivations of source code must retain the above
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+ copyright notice, this list of conditions and the following disclaimer.
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+
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+ 2. Redistributions or derivative works in binary form must reproduce
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+ the above copyright notice. This list of conditions and the following
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+ disclaimer must be reproduced in the documentation and/or other
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+ materials provided with the distribution.
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+
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+ 3. Neither the name of the copyright holder nor the names of its
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+ contributors may be used to endorse or promote products derived
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+ from this software without specific prior written permission.
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+
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+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
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+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
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+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
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+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
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+ HOLDER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
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+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
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+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
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+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
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+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
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+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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+
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+ Links
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+ ~~~~~
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+ YouTube: https://www.youtube.com/javidx9
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+ Discord: https://discord.gg/WhwHUMV
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+ Twitter: https://www.twitter.com/javidx9
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+ Twitch: https://www.twitch.tv/javidx9
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+ GitHub: https://www.github.com/onelonecoder
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+ Homepage: https://www.onelonecoder.com
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+ Patreon: https://www.patreon.com/javidx9
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+
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+ Author
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+ ~~~~~~
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+ David Barr, aka javidx9, ©OneLoneCoder 2019
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+*/
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+
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+
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+#ifndef OLC_PGEX_SOUND_H
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+#define OLC_PGEX_SOUND_H
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+
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+#include <istream>
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+#include <cstring>
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+#include <climits>
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+
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+#include <algorithm>
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+#undef min
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+#undef max
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+
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+// Choose a default sound backend
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+#if !defined(USE_ALSA) && !defined(USE_OPENAL) && !defined(USE_WINDOWS)
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+#ifdef __linux__
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+#define USE_ALSA
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+#endif
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+
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+#ifdef __EMSCRIPTEN__
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+#define USE_OPENAL
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+#endif
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+
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+#ifdef _WIN32
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+#define USE_WINDOWS
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+#endif
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+
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+#endif
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+
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+#ifdef USE_ALSA
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+#define ALSA_PCM_NEW_HW_PARAMS_API
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+#include <alsa/asoundlib.h>
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+#endif
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+
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+#ifdef USE_OPENAL
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+#include <AL/al.h>
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+#include <AL/alc.h>
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+#include <queue>
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+#endif
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+
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+#pragma pack(push, 1)
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+typedef struct {
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+ uint16_t wFormatTag;
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+ uint16_t nChannels;
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+ uint32_t nSamplesPerSec;
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+ uint32_t nAvgBytesPerSec;
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+ uint16_t nBlockAlign;
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+ uint16_t wBitsPerSample;
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+ uint16_t cbSize;
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+} OLC_WAVEFORMATEX;
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+#pragma pack(pop)
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+
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+namespace olc
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+{
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+ // Container class for Advanced 2D Drawing functions
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+ class SOUND : public olc::PGEX
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+ {
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+ // A representation of an affine transform, used to rotate, scale, offset & shear space
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+ public:
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+ class AudioSample
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+ {
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+ public:
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+ AudioSample();
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+ AudioSample(std::string sWavFile, olc::ResourcePack *pack = nullptr);
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+ olc::rcode LoadFromFile(std::string sWavFile, olc::ResourcePack *pack = nullptr);
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+
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+ public:
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+ OLC_WAVEFORMATEX wavHeader;
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+ float *fSample = nullptr;
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+ long nSamples = 0;
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+ int nChannels = 0;
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+ bool bSampleValid = false;
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+ };
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+
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+ struct sCurrentlyPlayingSample
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+ {
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+ int nAudioSampleID = 0;
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+ long nSamplePosition = 0;
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+ bool bFinished = false;
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+ bool bLoop = false;
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+ bool bFlagForStop = false;
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+ };
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+
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+ static std::list<sCurrentlyPlayingSample> listActiveSamples;
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+
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+ public:
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+ static bool InitialiseAudio(unsigned int nSampleRate = 44100, unsigned int nChannels = 1, unsigned int nBlocks = 8, unsigned int nBlockSamples = 512);
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+ static bool DestroyAudio();
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+ static void SetUserSynthFunction(std::function<float(int, float, float)> func);
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+ static void SetUserFilterFunction(std::function<float(int, float, float)> func);
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+
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+ public:
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+ static int LoadAudioSample(std::string sWavFile, olc::ResourcePack *pack = nullptr);
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+ static void PlaySample(int id, bool bLoop = false);
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+ static void StopSample(int id);
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+ static void StopAll();
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+ static float GetMixerOutput(int nChannel, float fGlobalTime, float fTimeStep);
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+
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+
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+ private:
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+#ifdef USE_WINDOWS // Windows specific sound management
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+ static void CALLBACK waveOutProc(HWAVEOUT hWaveOut, UINT uMsg, DWORD dwParam1, DWORD dwParam2);
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+ static unsigned int m_nSampleRate;
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+ static unsigned int m_nChannels;
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+ static unsigned int m_nBlockCount;
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+ static unsigned int m_nBlockSamples;
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+ static unsigned int m_nBlockCurrent;
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+ static short* m_pBlockMemory;
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+ static WAVEHDR *m_pWaveHeaders;
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+ static HWAVEOUT m_hwDevice;
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+ static std::atomic<unsigned int> m_nBlockFree;
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+ static std::condition_variable m_cvBlockNotZero;
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+ static std::mutex m_muxBlockNotZero;
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+#endif
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+
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+#ifdef USE_ALSA
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+ static snd_pcm_t *m_pPCM;
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+ static unsigned int m_nSampleRate;
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+ static unsigned int m_nChannels;
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+ static unsigned int m_nBlockSamples;
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+ static short* m_pBlockMemory;
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+#endif
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+
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+#ifdef USE_OPENAL
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+ static std::queue<ALuint> m_qAvailableBuffers;
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+ static ALuint *m_pBuffers;
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+ static ALuint m_nSource;
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+ static ALCdevice *m_pDevice;
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+ static ALCcontext *m_pContext;
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+ static unsigned int m_nSampleRate;
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+ static unsigned int m_nChannels;
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+ static unsigned int m_nBlockCount;
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+ static unsigned int m_nBlockSamples;
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+ static short* m_pBlockMemory;
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+#endif
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+
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+ static void AudioThread();
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+ static std::thread m_AudioThread;
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+ static std::atomic<bool> m_bAudioThreadActive;
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+ static std::atomic<float> m_fGlobalTime;
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+ static std::function<float(int, float, float)> funcUserSynth;
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+ static std::function<float(int, float, float)> funcUserFilter;
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+ };
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+}
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+
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+
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+// Implementation, platform-independent
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+
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+#ifdef OLC_PGEX_SOUND
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+#undef OLC_PGEX_SOUND
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+
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+namespace olc
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+{
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+ SOUND::AudioSample::AudioSample()
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+ { }
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+
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+ SOUND::AudioSample::AudioSample(std::string sWavFile, olc::ResourcePack *pack)
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+ {
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+ LoadFromFile(sWavFile, pack);
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+ }
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+
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+ olc::rcode SOUND::AudioSample::LoadFromFile(std::string sWavFile, olc::ResourcePack *pack)
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+ {
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+ auto ReadWave = [&](std::istream &is)
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+ {
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+ char dump[4];
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+ is.read(dump, sizeof(char) * 4); // Read "RIFF"
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+ if (strncmp(dump, "RIFF", 4) != 0) return olc::FAIL;
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+ is.read(dump, sizeof(char) * 4); // Not Interested
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+ is.read(dump, sizeof(char) * 4); // Read "WAVE"
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+ if (strncmp(dump, "WAVE", 4) != 0) return olc::FAIL;
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+
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+ // Read Wave description chunk
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+ is.read(dump, sizeof(char) * 4); // Read "fmt "
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+ unsigned int nHeaderSize = 0;
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+ is.read((char*)&nHeaderSize, sizeof(unsigned int)); // Not Interested
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+ is.read((char*)&wavHeader, nHeaderSize);// sizeof(WAVEFORMATEX)); // Read Wave Format Structure chunk
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+ // Note the -2, because the structure has 2 bytes to indicate its own size
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+ // which are not in the wav file
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+
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+ // Just check if wave format is compatible with olcPGE
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+ if (wavHeader.wBitsPerSample != 16 || wavHeader.nSamplesPerSec != 44100)
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+ return olc::FAIL;
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+
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+ // Search for audio data chunk
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+ uint32_t nChunksize = 0;
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+ is.read(dump, sizeof(char) * 4); // Read chunk header
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+ is.read((char*)&nChunksize, sizeof(uint32_t)); // Read chunk size
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+ while (strncmp(dump, "data", 4) != 0)
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+ {
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+ // Not audio data, so just skip it
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+ //std::fseek(f, nChunksize, SEEK_CUR);
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+ is.seekg(nChunksize, std::istream::cur);
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+ is.read(dump, sizeof(char) * 4);
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+ is.read((char*)&nChunksize, sizeof(uint32_t));
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+ }
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+
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+ // Finally got to data, so read it all in and convert to float samples
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+ nSamples = nChunksize / (wavHeader.nChannels * (wavHeader.wBitsPerSample >> 3));
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+ nChannels = wavHeader.nChannels;
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+
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+ // Create floating point buffer to hold audio sample
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+ fSample = new float[nSamples * nChannels];
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+ float *pSample = fSample;
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+
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+ // Read in audio data and normalise
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+ for (long i = 0; i < nSamples; i++)
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+ {
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+ for (int c = 0; c < nChannels; c++)
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+ {
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+ short s = 0;
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+ if (!is.eof())
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+ {
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+ is.read((char*)&s, sizeof(short));
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+
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+ *pSample = (float)s / (float)(SHRT_MAX);
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+ pSample++;
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+ }
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+ }
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+ }
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+
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+ // All done, flag sound as valid
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+ bSampleValid = true;
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+ return olc::OK;
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+ };
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+
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+ if (pack != nullptr)
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+ {
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+ olc::ResourcePack::sEntry entry = pack->GetStreamBuffer(sWavFile);
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+ std::istream is(&entry);
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+ return ReadWave(is);
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+ }
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+ else
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+ {
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+ // Read from file
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+ std::ifstream ifs(sWavFile, std::ifstream::binary);
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+ if (ifs.is_open())
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+ {
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+ return ReadWave(ifs);
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+ }
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+ else
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+ return olc::FAIL;
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+ }
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+ }
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+
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+ // This vector holds all loaded sound samples in memory
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+ std::vector<olc::SOUND::AudioSample> vecAudioSamples;
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+
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+ // This structure represents a sound that is currently playing. It only
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+ // holds the sound ID and where this instance of it is up to for its
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+ // current playback
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+
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+ void SOUND::SetUserSynthFunction(std::function<float(int, float, float)> func)
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+ {
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+ funcUserSynth = func;
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+ }
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+
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+ void SOUND::SetUserFilterFunction(std::function<float(int, float, float)> func)
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+ {
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+ funcUserFilter = func;
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+ }
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+
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+ // Load a 16-bit WAVE file @ 44100Hz ONLY into memory. A sample ID
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+ // number is returned if successful, otherwise -1
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+ int SOUND::LoadAudioSample(std::string sWavFile, olc::ResourcePack *pack)
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+ {
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+
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+ olc::SOUND::AudioSample a(sWavFile, pack);
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+ if (a.bSampleValid)
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+ {
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+ vecAudioSamples.push_back(a);
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+ return (unsigned int)vecAudioSamples.size();
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+ }
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+ else
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+ return -1;
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+ }
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+
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+ // Add sample 'id' to the mixers sounds to play list
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+ void SOUND::PlaySample(int id, bool bLoop)
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+ {
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+ olc::SOUND::sCurrentlyPlayingSample a;
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+ a.nAudioSampleID = id;
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+ a.nSamplePosition = 0;
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+ a.bFinished = false;
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+ a.bFlagForStop = false;
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+ a.bLoop = bLoop;
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+ SOUND::listActiveSamples.push_back(a);
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+ }
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+
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+ void SOUND::StopSample(int id)
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+ {
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+ // Find first occurence of sample id
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361
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+ auto s = std::find_if(listActiveSamples.begin(), listActiveSamples.end(), [&](const olc::SOUND::sCurrentlyPlayingSample &s) { return s.nAudioSampleID == id; });
|
|
362
|
+ if (s != listActiveSamples.end())
|
|
363
|
+ s->bFlagForStop = true;
|
|
364
|
+ }
|
|
365
|
+
|
|
366
|
+ void SOUND::StopAll()
|
|
367
|
+ {
|
|
368
|
+ for (auto &s : listActiveSamples)
|
|
369
|
+ {
|
|
370
|
+ s.bFlagForStop = true;
|
|
371
|
+ }
|
|
372
|
+ }
|
|
373
|
+
|
|
374
|
+ float SOUND::GetMixerOutput(int nChannel, float fGlobalTime, float fTimeStep)
|
|
375
|
+ {
|
|
376
|
+ // Accumulate sample for this channel
|
|
377
|
+ float fMixerSample = 0.0f;
|
|
378
|
+
|
|
379
|
+ for (auto &s : listActiveSamples)
|
|
380
|
+ {
|
|
381
|
+ if (m_bAudioThreadActive)
|
|
382
|
+ {
|
|
383
|
+ if (s.bFlagForStop)
|
|
384
|
+ {
|
|
385
|
+ s.bLoop = false;
|
|
386
|
+ s.bFinished = true;
|
|
387
|
+ }
|
|
388
|
+ else
|
|
389
|
+ {
|
|
390
|
+ // Calculate sample position
|
|
391
|
+ s.nSamplePosition += roundf((float)vecAudioSamples[s.nAudioSampleID - 1].wavHeader.nSamplesPerSec * fTimeStep);
|
|
392
|
+
|
|
393
|
+ // If sample position is valid add to the mix
|
|
394
|
+ if (s.nSamplePosition < vecAudioSamples[s.nAudioSampleID - 1].nSamples)
|
|
395
|
+ fMixerSample += vecAudioSamples[s.nAudioSampleID - 1].fSample[(s.nSamplePosition * vecAudioSamples[s.nAudioSampleID - 1].nChannels) + nChannel];
|
|
396
|
+ else
|
|
397
|
+ {
|
|
398
|
+ if (s.bLoop)
|
|
399
|
+ {
|
|
400
|
+ s.nSamplePosition = 0;
|
|
401
|
+ }
|
|
402
|
+ else
|
|
403
|
+ s.bFinished = true; // Else sound has completed
|
|
404
|
+ }
|
|
405
|
+ }
|
|
406
|
+ }
|
|
407
|
+ else
|
|
408
|
+ return 0.0f;
|
|
409
|
+ }
|
|
410
|
+
|
|
411
|
+ // If sounds have completed then remove them
|
|
412
|
+ listActiveSamples.remove_if([](const sCurrentlyPlayingSample &s) {return s.bFinished; });
|
|
413
|
+
|
|
414
|
+ // The users application might be generating sound, so grab that if it exists
|
|
415
|
+ if (funcUserSynth != nullptr)
|
|
416
|
+ fMixerSample += funcUserSynth(nChannel, fGlobalTime, fTimeStep);
|
|
417
|
+
|
|
418
|
+ // Return the sample via an optional user override to filter the sound
|
|
419
|
+ if (funcUserFilter != nullptr)
|
|
420
|
+ return funcUserFilter(nChannel, fGlobalTime, fMixerSample);
|
|
421
|
+ else
|
|
422
|
+ return fMixerSample;
|
|
423
|
+ }
|
|
424
|
+
|
|
425
|
+ std::thread SOUND::m_AudioThread;
|
|
426
|
+ std::atomic<bool> SOUND::m_bAudioThreadActive{ false };
|
|
427
|
+ std::atomic<float> SOUND::m_fGlobalTime{ 0.0f };
|
|
428
|
+ std::list<SOUND::sCurrentlyPlayingSample> SOUND::listActiveSamples;
|
|
429
|
+ std::function<float(int, float, float)> SOUND::funcUserSynth = nullptr;
|
|
430
|
+ std::function<float(int, float, float)> SOUND::funcUserFilter = nullptr;
|
|
431
|
+}
|
|
432
|
+
|
|
433
|
+// Implementation, Windows-specific
|
|
434
|
+#ifdef USE_WINDOWS
|
|
435
|
+#pragma comment(lib, "winmm.lib")
|
|
436
|
+
|
|
437
|
+namespace olc
|
|
438
|
+{
|
|
439
|
+ bool SOUND::InitialiseAudio(unsigned int nSampleRate, unsigned int nChannels, unsigned int nBlocks, unsigned int nBlockSamples)
|
|
440
|
+ {
|
|
441
|
+ // Initialise Sound Engine
|
|
442
|
+ m_bAudioThreadActive = false;
|
|
443
|
+ m_nSampleRate = nSampleRate;
|
|
444
|
+ m_nChannels = nChannels;
|
|
445
|
+ m_nBlockCount = nBlocks;
|
|
446
|
+ m_nBlockSamples = nBlockSamples;
|
|
447
|
+ m_nBlockFree = m_nBlockCount;
|
|
448
|
+ m_nBlockCurrent = 0;
|
|
449
|
+ m_pBlockMemory = nullptr;
|
|
450
|
+ m_pWaveHeaders = nullptr;
|
|
451
|
+
|
|
452
|
+ // Device is available
|
|
453
|
+ WAVEFORMATEX waveFormat;
|
|
454
|
+ waveFormat.wFormatTag = WAVE_FORMAT_PCM;
|
|
455
|
+ waveFormat.nSamplesPerSec = m_nSampleRate;
|
|
456
|
+ waveFormat.wBitsPerSample = sizeof(short) * 8;
|
|
457
|
+ waveFormat.nChannels = m_nChannels;
|
|
458
|
+ waveFormat.nBlockAlign = (waveFormat.wBitsPerSample / 8) * waveFormat.nChannels;
|
|
459
|
+ waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
|
|
460
|
+ waveFormat.cbSize = 0;
|
|
461
|
+
|
|
462
|
+ listActiveSamples.clear();
|
|
463
|
+
|
|
464
|
+ // Open Device if valid
|
|
465
|
+ if (waveOutOpen(&m_hwDevice, WAVE_MAPPER, &waveFormat, (DWORD_PTR)SOUND::waveOutProc, (DWORD_PTR)0, CALLBACK_FUNCTION) != S_OK)
|
|
466
|
+ return DestroyAudio();
|
|
467
|
+
|
|
468
|
+ // Allocate Wave|Block Memory
|
|
469
|
+ m_pBlockMemory = new short[m_nBlockCount * m_nBlockSamples];
|
|
470
|
+ if (m_pBlockMemory == nullptr)
|
|
471
|
+ return DestroyAudio();
|
|
472
|
+ ZeroMemory(m_pBlockMemory, sizeof(short) * m_nBlockCount * m_nBlockSamples);
|
|
473
|
+
|
|
474
|
+ m_pWaveHeaders = new WAVEHDR[m_nBlockCount];
|
|
475
|
+ if (m_pWaveHeaders == nullptr)
|
|
476
|
+ return DestroyAudio();
|
|
477
|
+ ZeroMemory(m_pWaveHeaders, sizeof(WAVEHDR) * m_nBlockCount);
|
|
478
|
+
|
|
479
|
+ // Link headers to block memory
|
|
480
|
+ for (unsigned int n = 0; n < m_nBlockCount; n++)
|
|
481
|
+ {
|
|
482
|
+ m_pWaveHeaders[n].dwBufferLength = m_nBlockSamples * sizeof(short);
|
|
483
|
+ m_pWaveHeaders[n].lpData = (LPSTR)(m_pBlockMemory + (n * m_nBlockSamples));
|
|
484
|
+ }
|
|
485
|
+
|
|
486
|
+ m_bAudioThreadActive = true;
|
|
487
|
+ m_AudioThread = std::thread(&SOUND::AudioThread);
|
|
488
|
+
|
|
489
|
+ // Start the ball rolling with the sound delivery thread
|
|
490
|
+ std::unique_lock<std::mutex> lm(m_muxBlockNotZero);
|
|
491
|
+ m_cvBlockNotZero.notify_one();
|
|
492
|
+ return true;
|
|
493
|
+ }
|
|
494
|
+
|
|
495
|
+ // Stop and clean up audio system
|
|
496
|
+ bool SOUND::DestroyAudio()
|
|
497
|
+ {
|
|
498
|
+ m_bAudioThreadActive = false;
|
|
499
|
+ m_AudioThread.join();
|
|
500
|
+ return false;
|
|
501
|
+ }
|
|
502
|
+
|
|
503
|
+ // Handler for soundcard request for more data
|
|
504
|
+ void CALLBACK SOUND::waveOutProc(HWAVEOUT hWaveOut, UINT uMsg, DWORD dwParam1, DWORD dwParam2)
|
|
505
|
+ {
|
|
506
|
+ if (uMsg != WOM_DONE) return;
|
|
507
|
+ m_nBlockFree++;
|
|
508
|
+ std::unique_lock<std::mutex> lm(m_muxBlockNotZero);
|
|
509
|
+ m_cvBlockNotZero.notify_one();
|
|
510
|
+ }
|
|
511
|
+
|
|
512
|
+ // Audio thread. This loop responds to requests from the soundcard to fill 'blocks'
|
|
513
|
+ // with audio data. If no requests are available it goes dormant until the sound
|
|
514
|
+ // card is ready for more data. The block is fille by the "user" in some manner
|
|
515
|
+ // and then issued to the soundcard.
|
|
516
|
+ void SOUND::AudioThread()
|
|
517
|
+ {
|
|
518
|
+ m_fGlobalTime = 0.0f;
|
|
519
|
+ static float fTimeStep = 1.0f / (float)m_nSampleRate;
|
|
520
|
+
|
|
521
|
+ // Goofy hack to get maximum integer for a type at run-time
|
|
522
|
+ short nMaxSample = (short)pow(2, (sizeof(short) * 8) - 1) - 1;
|
|
523
|
+ float fMaxSample = (float)nMaxSample;
|
|
524
|
+ short nPreviousSample = 0;
|
|
525
|
+
|
|
526
|
+ while (m_bAudioThreadActive)
|
|
527
|
+ {
|
|
528
|
+ // Wait for block to become available
|
|
529
|
+ if (m_nBlockFree == 0)
|
|
530
|
+ {
|
|
531
|
+ std::unique_lock<std::mutex> lm(m_muxBlockNotZero);
|
|
532
|
+ while (m_nBlockFree == 0) // sometimes, Windows signals incorrectly
|
|
533
|
+ m_cvBlockNotZero.wait(lm);
|
|
534
|
+ }
|
|
535
|
+
|
|
536
|
+ // Block is here, so use it
|
|
537
|
+ m_nBlockFree--;
|
|
538
|
+
|
|
539
|
+ // Prepare block for processing
|
|
540
|
+ if (m_pWaveHeaders[m_nBlockCurrent].dwFlags & WHDR_PREPARED)
|
|
541
|
+ waveOutUnprepareHeader(m_hwDevice, &m_pWaveHeaders[m_nBlockCurrent], sizeof(WAVEHDR));
|
|
542
|
+
|
|
543
|
+ short nNewSample = 0;
|
|
544
|
+ int nCurrentBlock = m_nBlockCurrent * m_nBlockSamples;
|
|
545
|
+
|
|
546
|
+ auto clip = [](float fSample, float fMax)
|
|
547
|
+ {
|
|
548
|
+ if (fSample >= 0.0)
|
|
549
|
+ return fmin(fSample, fMax);
|
|
550
|
+ else
|
|
551
|
+ return fmax(fSample, -fMax);
|
|
552
|
+ };
|
|
553
|
+
|
|
554
|
+ for (unsigned int n = 0; n < m_nBlockSamples; n += m_nChannels)
|
|
555
|
+ {
|
|
556
|
+ // User Process
|
|
557
|
+ for (unsigned int c = 0; c < m_nChannels; c++)
|
|
558
|
+ {
|
|
559
|
+ nNewSample = (short)(clip(GetMixerOutput(c, m_fGlobalTime, fTimeStep), 1.0) * fMaxSample);
|
|
560
|
+ m_pBlockMemory[nCurrentBlock + n + c] = nNewSample;
|
|
561
|
+ nPreviousSample = nNewSample;
|
|
562
|
+ }
|
|
563
|
+
|
|
564
|
+ m_fGlobalTime = m_fGlobalTime + fTimeStep;
|
|
565
|
+ }
|
|
566
|
+
|
|
567
|
+ // Send block to sound device
|
|
568
|
+ waveOutPrepareHeader(m_hwDevice, &m_pWaveHeaders[m_nBlockCurrent], sizeof(WAVEHDR));
|
|
569
|
+ waveOutWrite(m_hwDevice, &m_pWaveHeaders[m_nBlockCurrent], sizeof(WAVEHDR));
|
|
570
|
+ m_nBlockCurrent++;
|
|
571
|
+ m_nBlockCurrent %= m_nBlockCount;
|
|
572
|
+ }
|
|
573
|
+ }
|
|
574
|
+
|
|
575
|
+ unsigned int SOUND::m_nSampleRate = 0;
|
|
576
|
+ unsigned int SOUND::m_nChannels = 0;
|
|
577
|
+ unsigned int SOUND::m_nBlockCount = 0;
|
|
578
|
+ unsigned int SOUND::m_nBlockSamples = 0;
|
|
579
|
+ unsigned int SOUND::m_nBlockCurrent = 0;
|
|
580
|
+ short* SOUND::m_pBlockMemory = nullptr;
|
|
581
|
+ WAVEHDR *SOUND::m_pWaveHeaders = nullptr;
|
|
582
|
+ HWAVEOUT SOUND::m_hwDevice;
|
|
583
|
+ std::atomic<unsigned int> SOUND::m_nBlockFree = 0;
|
|
584
|
+ std::condition_variable SOUND::m_cvBlockNotZero;
|
|
585
|
+ std::mutex SOUND::m_muxBlockNotZero;
|
|
586
|
+}
|
|
587
|
+
|
|
588
|
+#elif defined(USE_ALSA)
|
|
589
|
+
|
|
590
|
+namespace olc
|
|
591
|
+{
|
|
592
|
+ bool SOUND::InitialiseAudio(unsigned int nSampleRate, unsigned int nChannels, unsigned int nBlocks, unsigned int nBlockSamples)
|
|
593
|
+ {
|
|
594
|
+ // Initialise Sound Engine
|
|
595
|
+ m_bAudioThreadActive = false;
|
|
596
|
+ m_nSampleRate = nSampleRate;
|
|
597
|
+ m_nChannels = nChannels;
|
|
598
|
+ m_nBlockSamples = nBlockSamples;
|
|
599
|
+ m_pBlockMemory = nullptr;
|
|
600
|
+
|
|
601
|
+ // Open PCM stream
|
|
602
|
+ int rc = snd_pcm_open(&m_pPCM, "default", SND_PCM_STREAM_PLAYBACK, 0);
|
|
603
|
+ if (rc < 0)
|
|
604
|
+ return DestroyAudio();
|
|
605
|
+
|
|
606
|
+
|
|
607
|
+ // Prepare the parameter structure and set default parameters
|
|
608
|
+ snd_pcm_hw_params_t *params;
|
|
609
|
+ snd_pcm_hw_params_alloca(¶ms);
|
|
610
|
+ snd_pcm_hw_params_any(m_pPCM, params);
|
|
611
|
+
|
|
612
|
+ // Set other parameters
|
|
613
|
+ snd_pcm_hw_params_set_format(m_pPCM, params, SND_PCM_FORMAT_S16_LE);
|
|
614
|
+ snd_pcm_hw_params_set_rate(m_pPCM, params, m_nSampleRate, 0);
|
|
615
|
+ snd_pcm_hw_params_set_channels(m_pPCM, params, m_nChannels);
|
|
616
|
+ snd_pcm_hw_params_set_period_size(m_pPCM, params, m_nBlockSamples, 0);
|
|
617
|
+ snd_pcm_hw_params_set_periods(m_pPCM, params, nBlocks, 0);
|
|
618
|
+
|
|
619
|
+ // Save these parameters
|
|
620
|
+ rc = snd_pcm_hw_params(m_pPCM, params);
|
|
621
|
+ if (rc < 0)
|
|
622
|
+ return DestroyAudio();
|
|
623
|
+
|
|
624
|
+ listActiveSamples.clear();
|
|
625
|
+
|
|
626
|
+ // Allocate Wave|Block Memory
|
|
627
|
+ m_pBlockMemory = new short[m_nBlockSamples];
|
|
628
|
+ if (m_pBlockMemory == nullptr)
|
|
629
|
+ return DestroyAudio();
|
|
630
|
+ std::fill(m_pBlockMemory, m_pBlockMemory + m_nBlockSamples, 0);
|
|
631
|
+
|
|
632
|
+ // Unsure if really needed, helped prevent underrun on my setup
|
|
633
|
+ snd_pcm_start(m_pPCM);
|
|
634
|
+ for (unsigned int i = 0; i < nBlocks; i++)
|
|
635
|
+ rc = snd_pcm_writei(m_pPCM, m_pBlockMemory, 512);
|
|
636
|
+
|
|
637
|
+ snd_pcm_start(m_pPCM);
|
|
638
|
+ m_bAudioThreadActive = true;
|
|
639
|
+ m_AudioThread = std::thread(&SOUND::AudioThread);
|
|
640
|
+
|
|
641
|
+ return true;
|
|
642
|
+ }
|
|
643
|
+
|
|
644
|
+ // Stop and clean up audio system
|
|
645
|
+ bool SOUND::DestroyAudio()
|
|
646
|
+ {
|
|
647
|
+ m_bAudioThreadActive = false;
|
|
648
|
+ m_AudioThread.join();
|
|
649
|
+ snd_pcm_drain(m_pPCM);
|
|
650
|
+ snd_pcm_close(m_pPCM);
|
|
651
|
+ return false;
|
|
652
|
+ }
|
|
653
|
+
|
|
654
|
+
|
|
655
|
+ // Audio thread. This loop responds to requests from the soundcard to fill 'blocks'
|
|
656
|
+ // with audio data. If no requests are available it goes dormant until the sound
|
|
657
|
+ // card is ready for more data. The block is fille by the "user" in some manner
|
|
658
|
+ // and then issued to the soundcard.
|
|
659
|
+ void SOUND::AudioThread()
|
|
660
|
+ {
|
|
661
|
+ m_fGlobalTime = 0.0f;
|
|
662
|
+ static float fTimeStep = 1.0f / (float)m_nSampleRate;
|
|
663
|
+
|
|
664
|
+ // Goofy hack to get maximum integer for a type at run-time
|
|
665
|
+ short nMaxSample = (short)pow(2, (sizeof(short) * 8) - 1) - 1;
|
|
666
|
+ float fMaxSample = (float)nMaxSample;
|
|
667
|
+ short nPreviousSample = 0;
|
|
668
|
+
|
|
669
|
+ while (m_bAudioThreadActive)
|
|
670
|
+ {
|
|
671
|
+ short nNewSample = 0;
|
|
672
|
+
|
|
673
|
+ auto clip = [](float fSample, float fMax)
|
|
674
|
+ {
|
|
675
|
+ if (fSample >= 0.0)
|
|
676
|
+ return fmin(fSample, fMax);
|
|
677
|
+ else
|
|
678
|
+ return fmax(fSample, -fMax);
|
|
679
|
+ };
|
|
680
|
+
|
|
681
|
+ for (unsigned int n = 0; n < m_nBlockSamples; n += m_nChannels)
|
|
682
|
+ {
|
|
683
|
+ // User Process
|
|
684
|
+ for (unsigned int c = 0; c < m_nChannels; c++)
|
|
685
|
+ {
|
|
686
|
+ nNewSample = (short)(clip(GetMixerOutput(c, m_fGlobalTime, fTimeStep), 1.0) * fMaxSample);
|
|
687
|
+ m_pBlockMemory[n + c] = nNewSample;
|
|
688
|
+ nPreviousSample = nNewSample;
|
|
689
|
+ }
|
|
690
|
+
|
|
691
|
+ m_fGlobalTime = m_fGlobalTime + fTimeStep;
|
|
692
|
+ }
|
|
693
|
+
|
|
694
|
+ // Send block to sound device
|
|
695
|
+ snd_pcm_uframes_t nLeft = m_nBlockSamples;
|
|
696
|
+ short *pBlockPos = m_pBlockMemory;
|
|
697
|
+ while (nLeft > 0)
|
|
698
|
+ {
|
|
699
|
+ int rc = snd_pcm_writei(m_pPCM, pBlockPos, nLeft);
|
|
700
|
+ if (rc > 0)
|
|
701
|
+ {
|
|
702
|
+ pBlockPos += rc * m_nChannels;
|
|
703
|
+ nLeft -= rc;
|
|
704
|
+ }
|
|
705
|
+ if (rc == -EAGAIN) continue;
|
|
706
|
+ if (rc == -EPIPE) // an underrun occured, prepare the device for more data
|
|
707
|
+ snd_pcm_prepare(m_pPCM);
|
|
708
|
+ }
|
|
709
|
+ }
|
|
710
|
+ }
|
|
711
|
+
|
|
712
|
+ snd_pcm_t* SOUND::m_pPCM = nullptr;
|
|
713
|
+ unsigned int SOUND::m_nSampleRate = 0;
|
|
714
|
+ unsigned int SOUND::m_nChannels = 0;
|
|
715
|
+ unsigned int SOUND::m_nBlockSamples = 0;
|
|
716
|
+ short* SOUND::m_pBlockMemory = nullptr;
|
|
717
|
+}
|
|
718
|
+
|
|
719
|
+#elif defined(USE_OPENAL)
|
|
720
|
+
|
|
721
|
+namespace olc
|
|
722
|
+{
|
|
723
|
+ bool SOUND::InitialiseAudio(unsigned int nSampleRate, unsigned int nChannels, unsigned int nBlocks, unsigned int nBlockSamples)
|
|
724
|
+ {
|
|
725
|
+ // Initialise Sound Engine
|
|
726
|
+ m_bAudioThreadActive = false;
|
|
727
|
+ m_nSampleRate = nSampleRate;
|
|
728
|
+ m_nChannels = nChannels;
|
|
729
|
+ m_nBlockCount = nBlocks;
|
|
730
|
+ m_nBlockSamples = nBlockSamples;
|
|
731
|
+ m_pBlockMemory = nullptr;
|
|
732
|
+
|
|
733
|
+ // Open the device and create the context
|
|
734
|
+ m_pDevice = alcOpenDevice(NULL);
|
|
735
|
+ if (m_pDevice)
|
|
736
|
+ {
|
|
737
|
+ m_pContext = alcCreateContext(m_pDevice, NULL);
|
|
738
|
+ alcMakeContextCurrent(m_pContext);
|
|
739
|
+ }
|
|
740
|
+ else
|
|
741
|
+ return DestroyAudio();
|
|
742
|
+
|
|
743
|
+ // Allocate memory for sound data
|
|
744
|
+ alGetError();
|
|
745
|
+ m_pBuffers = new ALuint[m_nBlockCount];
|
|
746
|
+ alGenBuffers(m_nBlockCount, m_pBuffers);
|
|
747
|
+ alGenSources(1, &m_nSource);
|
|
748
|
+
|
|
749
|
+ for (unsigned int i = 0; i < m_nBlockCount; i++)
|
|
750
|
+ m_qAvailableBuffers.push(m_pBuffers[i]);
|
|
751
|
+
|
|
752
|
+ listActiveSamples.clear();
|
|
753
|
+
|
|
754
|
+ // Allocate Wave|Block Memory
|
|
755
|
+ m_pBlockMemory = new short[m_nBlockSamples];
|
|
756
|
+ if (m_pBlockMemory == nullptr)
|
|
757
|
+ return DestroyAudio();
|
|
758
|
+ std::fill(m_pBlockMemory, m_pBlockMemory + m_nBlockSamples, 0);
|
|
759
|
+
|
|
760
|
+ m_bAudioThreadActive = true;
|
|
761
|
+ m_AudioThread = std::thread(&SOUND::AudioThread);
|
|
762
|
+ return true;
|
|
763
|
+ }
|
|
764
|
+
|
|
765
|
+ // Stop and clean up audio system
|
|
766
|
+ bool SOUND::DestroyAudio()
|
|
767
|
+ {
|
|
768
|
+ m_bAudioThreadActive = false;
|
|
769
|
+ m_AudioThread.join();
|
|
770
|
+
|
|
771
|
+ alDeleteBuffers(m_nBlockCount, m_pBuffers);
|
|
772
|
+ delete[] m_pBuffers;
|
|
773
|
+ alDeleteSources(1, &m_nSource);
|
|
774
|
+
|
|
775
|
+ alcMakeContextCurrent(NULL);
|
|
776
|
+ alcDestroyContext(m_pContext);
|
|
777
|
+ alcCloseDevice(m_pDevice);
|
|
778
|
+ return false;
|
|
779
|
+ }
|
|
780
|
+
|
|
781
|
+
|
|
782
|
+ // Audio thread. This loop responds to requests from the soundcard to fill 'blocks'
|
|
783
|
+ // with audio data. If no requests are available it goes dormant until the sound
|
|
784
|
+ // card is ready for more data. The block is fille by the "user" in some manner
|
|
785
|
+ // and then issued to the soundcard.
|
|
786
|
+ void SOUND::AudioThread()
|
|
787
|
+ {
|
|
788
|
+ m_fGlobalTime = 0.0f;
|
|
789
|
+ static float fTimeStep = 1.0f / (float)m_nSampleRate;
|
|
790
|
+
|
|
791
|
+ // Goofy hack to get maximum integer for a type at run-time
|
|
792
|
+ short nMaxSample = (short)pow(2, (sizeof(short) * 8) - 1) - 1;
|
|
793
|
+ float fMaxSample = (float)nMaxSample;
|
|
794
|
+ short nPreviousSample = 0;
|
|
795
|
+
|
|
796
|
+ std::vector<ALuint> vProcessed;
|
|
797
|
+
|
|
798
|
+ while (m_bAudioThreadActive)
|
|
799
|
+ {
|
|
800
|
+ ALint nState, nProcessed;
|
|
801
|
+ alGetSourcei(m_nSource, AL_SOURCE_STATE, &nState);
|
|
802
|
+ alGetSourcei(m_nSource, AL_BUFFERS_PROCESSED, &nProcessed);
|
|
803
|
+
|
|
804
|
+ // Add processed buffers to our queue
|
|
805
|
+ vProcessed.resize(nProcessed);
|
|
806
|
+ alSourceUnqueueBuffers(m_nSource, nProcessed, vProcessed.data());
|
|
807
|
+ for (ALint nBuf : vProcessed) m_qAvailableBuffers.push(nBuf);
|
|
808
|
+
|
|
809
|
+ // Wait until there is a free buffer (ewww)
|
|
810
|
+ if (m_qAvailableBuffers.empty()) continue;
|
|
811
|
+
|
|
812
|
+ short nNewSample = 0;
|
|
813
|
+
|
|
814
|
+ auto clip = [](float fSample, float fMax)
|
|
815
|
+ {
|
|
816
|
+ if (fSample >= 0.0)
|
|
817
|
+ return fmin(fSample, fMax);
|
|
818
|
+ else
|
|
819
|
+ return fmax(fSample, -fMax);
|
|
820
|
+ };
|
|
821
|
+
|
|
822
|
+ for (unsigned int n = 0; n < m_nBlockSamples; n += m_nChannels)
|
|
823
|
+ {
|
|
824
|
+ // User Process
|
|
825
|
+ for (unsigned int c = 0; c < m_nChannels; c++)
|
|
826
|
+ {
|
|
827
|
+ nNewSample = (short)(clip(GetMixerOutput(c, m_fGlobalTime, fTimeStep), 1.0) * fMaxSample);
|
|
828
|
+ m_pBlockMemory[n + c] = nNewSample;
|
|
829
|
+ nPreviousSample = nNewSample;
|
|
830
|
+ }
|
|
831
|
+
|
|
832
|
+ m_fGlobalTime = m_fGlobalTime + fTimeStep;
|
|
833
|
+ }
|
|
834
|
+
|
|
835
|
+ // Fill OpenAL data buffer
|
|
836
|
+ alBufferData(
|
|
837
|
+ m_qAvailableBuffers.front(),
|
|
838
|
+ m_nChannels == 1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16,
|
|
839
|
+ m_pBlockMemory,
|
|
840
|
+ 2 * m_nBlockSamples,
|
|
841
|
+ m_nSampleRate
|
|
842
|
+ );
|
|
843
|
+ // Add it to the OpenAL queue
|
|
844
|
+ alSourceQueueBuffers(m_nSource, 1, &m_qAvailableBuffers.front());
|
|
845
|
+ // Remove it from ours
|
|
846
|
+ m_qAvailableBuffers.pop();
|
|
847
|
+
|
|
848
|
+ // If it's not playing for some reason, change that
|
|
849
|
+ if (nState != AL_PLAYING)
|
|
850
|
+ alSourcePlay(m_nSource);
|
|
851
|
+ }
|
|
852
|
+ }
|
|
853
|
+
|
|
854
|
+ std::queue<ALuint> SOUND::m_qAvailableBuffers;
|
|
855
|
+ ALuint *SOUND::m_pBuffers = nullptr;
|
|
856
|
+ ALuint SOUND::m_nSource = 0;
|
|
857
|
+ ALCdevice *SOUND::m_pDevice = nullptr;
|
|
858
|
+ ALCcontext *SOUND::m_pContext = nullptr;
|
|
859
|
+ unsigned int SOUND::m_nSampleRate = 0;
|
|
860
|
+ unsigned int SOUND::m_nChannels = 0;
|
|
861
|
+ unsigned int SOUND::m_nBlockCount = 0;
|
|
862
|
+ unsigned int SOUND::m_nBlockSamples = 0;
|
|
863
|
+ short* SOUND::m_pBlockMemory = nullptr;
|
|
864
|
+}
|
|
865
|
+
|
|
866
|
+#else // Some other platform
|
|
867
|
+
|
|
868
|
+namespace olc
|
|
869
|
+{
|
|
870
|
+ bool SOUND::InitialiseAudio(unsigned int nSampleRate, unsigned int nChannels, unsigned int nBlocks, unsigned int nBlockSamples)
|
|
871
|
+ {
|
|
872
|
+ return true;
|
|
873
|
+ }
|
|
874
|
+
|
|
875
|
+ // Stop and clean up audio system
|
|
876
|
+ bool SOUND::DestroyAudio()
|
|
877
|
+ {
|
|
878
|
+ return false;
|
|
879
|
+ }
|
|
880
|
+
|
|
881
|
+
|
|
882
|
+ // Audio thread. This loop responds to requests from the soundcard to fill 'blocks'
|
|
883
|
+ // with audio data. If no requests are available it goes dormant until the sound
|
|
884
|
+ // card is ready for more data. The block is fille by the "user" in some manner
|
|
885
|
+ // and then issued to the soundcard.
|
|
886
|
+ void SOUND::AudioThread()
|
|
887
|
+ { }
|
|
888
|
+}
|
|
889
|
+
|
|
890
|
+#endif
|
|
891
|
+#endif
|
|
892
|
+#endif // OLC_PGEX_SOUND
|