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/*
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olcPGEX_Sound.h
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+-------------------------------------------------------------+
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| OneLoneCoder Pixel Game Engine Extension |
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| Sound - v0.4 |
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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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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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License (OLC-3)
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~~~~~~~~~~~~~~~
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Copyright 2018 - 2019 OneLoneCoder.com
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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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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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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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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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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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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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Author
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~~~~~~
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David Barr, aka javidx9, <EFBFBD>OneLoneCoder 2019
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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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#include <istream>
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#include <cstring>
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#include <climits>
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#include <algorithm>
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#undef min
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#undef max
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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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#ifdef __EMSCRIPTEN__
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#define USE_OPENAL
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#endif
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#ifdef _WIN32
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#define USE_WINDOWS
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#endif
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#endif
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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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#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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#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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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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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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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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static std::list<sCurrentlyPlayingSample> listActiveSamples;
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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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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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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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#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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#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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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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// Implementation, platform-independent
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#ifdef OLC_PGEX_SOUND
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#undef OLC_PGEX_SOUND
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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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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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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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// 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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// 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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// 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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// 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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// 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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// 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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*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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// 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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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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// This vector holds all loaded sound samples in memory
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std::vector<olc::SOUND::AudioSample> vecAudioSamples;
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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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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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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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// 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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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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// 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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void SOUND::StopSample(int id)
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{
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// Find first occurence of sample id
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auto s = std::find_if(listActiveSamples.begin(), listActiveSamples.end(), [&](const olc::SOUND::sCurrentlyPlayingSample &s) { return s.nAudioSampleID == id; });
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if (s != listActiveSamples.end())
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s->bFlagForStop = true;
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}
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void SOUND::StopAll()
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{
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for (auto &s : listActiveSamples)
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{
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s.bFlagForStop = true;
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}
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}
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float SOUND::GetMixerOutput(int nChannel, float fGlobalTime, float fTimeStep)
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{
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// Accumulate sample for this channel
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float fMixerSample = 0.0f;
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for (auto &s : listActiveSamples)
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{
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if (m_bAudioThreadActive)
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{
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if (s.bFlagForStop)
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{
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s.bLoop = false;
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s.bFinished = true;
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}
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else
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{
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// Calculate sample position
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s.nSamplePosition += roundf((float)vecAudioSamples[s.nAudioSampleID - 1].wavHeader.nSamplesPerSec * fTimeStep);
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// If sample position is valid add to the mix
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if (s.nSamplePosition < vecAudioSamples[s.nAudioSampleID - 1].nSamples)
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fMixerSample += vecAudioSamples[s.nAudioSampleID - 1].fSample[(s.nSamplePosition * vecAudioSamples[s.nAudioSampleID - 1].nChannels) + nChannel];
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else
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{
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if (s.bLoop)
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{
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s.nSamplePosition = 0;
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}
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else
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s.bFinished = true; // Else sound has completed
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}
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}
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}
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else
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return 0.0f;
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}
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// If sounds have completed then remove them
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listActiveSamples.remove_if([](const sCurrentlyPlayingSample &s) {return s.bFinished; });
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// The users application might be generating sound, so grab that if it exists
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if (funcUserSynth != nullptr)
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fMixerSample += funcUserSynth(nChannel, fGlobalTime, fTimeStep);
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// Return the sample via an optional user override to filter the sound
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if (funcUserFilter != nullptr)
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return funcUserFilter(nChannel, fGlobalTime, fMixerSample);
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else
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return fMixerSample;
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}
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std::thread SOUND::m_AudioThread;
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std::atomic<bool> SOUND::m_bAudioThreadActive{ false };
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std::atomic<float> SOUND::m_fGlobalTime{ 0.0f };
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std::list<SOUND::sCurrentlyPlayingSample> SOUND::listActiveSamples;
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std::function<float(int, float, float)> SOUND::funcUserSynth = nullptr;
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std::function<float(int, float, float)> SOUND::funcUserFilter = nullptr;
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}
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// Implementation, Windows-specific
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#ifdef USE_WINDOWS
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#pragma comment(lib, "winmm.lib")
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namespace olc
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{
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bool SOUND::InitialiseAudio(unsigned int nSampleRate, unsigned int nChannels, unsigned int nBlocks, unsigned int nBlockSamples)
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{
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// Initialise Sound Engine
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m_bAudioThreadActive = false;
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m_nSampleRate = nSampleRate;
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|
|
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;
|
|
|
if(m_AudioThread.joinable())
|
|
|
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;
|
|
|
|
|
|
auto tp1 = std::chrono::system_clock::now();
|
|
|
auto tp2 = std::chrono::system_clock::now();
|
|
|
|
|
|
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);
|
|
|
};
|
|
|
|
|
|
tp2 = std::chrono::system_clock::now();
|
|
|
std::chrono::duration<float> elapsedTime = tp2 - tp1;
|
|
|
tp1 = tp2;
|
|
|
|
|
|
// Our time per frame coefficient
|
|
|
float fElapsedTime = elapsedTime.count();
|
|
|
|
|
|
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 * (float)n, fTimeStep), 1.0) * fMaxSample);
|
|
|
m_pBlockMemory[nCurrentBlock + n + c] = nNewSample;
|
|
|
nPreviousSample = nNewSample;
|
|
|
}
|
|
|
}
|
|
|
|
|
|
m_fGlobalTime = m_fGlobalTime + fTimeStep * (float)m_nBlockSamples;
|
|
|
|
|
|
// 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(¶ms);
|
|
|
snd_pcm_hw_params_any(m_pPCM, params);
|
|
|
|
|
|
// Set other parameters
|
|
|
snd_pcm_hw_params_set_access(m_pPCM, params, SND_PCM_ACCESS_RW_INTERLEAVED);
|
|
|
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;
|
|
|
if(m_AudioThread.joinable())
|
|
|
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;
|
|
|
short nPreviousSample = 0;
|
|
|
|
|
|
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)(GetMixerOutput(c, m_fGlobalTime + fTimeStep * (float)n, fTimeStep), 1.0) * fMaxSample;
|
|
|
m_pBlockMemory[n + c] = nNewSample;
|
|
|
nPreviousSample = nNewSample;
|
|
|
}
|
|
|
}
|
|
|
|
|
|
m_fGlobalTime = m_fGlobalTime + fTimeStep * (float)m_nBlockSamples;
|
|
|
|
|
|
// 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;
|
|
|
if(m_AudioThread.joinable())
|
|
|
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
|
|
|
#endif
|
|
|
#endif // OLC_PGEX_SOUND
|
