ref: 0f1b1ba5cde71429f3404dd28d87488052573ec5
dir: /src/Sound.cpp/
#include <string> #include <math.h> #include <algorithm> #include <stdint.h> #include <SDL_audio.h> #include "Sound.h" #include "Organya.h" #include "PixTone.h" #define FREQUENCY 44100 #define STREAM_SIZE (FREQUENCY / 200) #define clamp(x, y, z) ((x > z) ? z : (x < y) ? y : x) //Audio device SDL_AudioDeviceID audioDevice; //Keep track of all existing sound buffers SOUNDBUFFER *soundBuffers; //Sound buffer code SOUNDBUFFER::SOUNDBUFFER(size_t bufSize) { //Lock audio buffer SDL_LockAudioDevice(audioDevice); //Set parameters size = bufSize; playing = false; looping = false; looped = false; frequency = 0.0; volume = 1.0; volume_l = 1.0; volume_r = 1.0; samplePosition = 0.0; //Create waveform buffer data = new uint8_t[bufSize]; memset(data, 0x80, bufSize); //Add to buffer list this->next = soundBuffers; soundBuffers = this; //Unlock audio buffer SDL_UnlockAudioDevice(audioDevice); } SOUNDBUFFER::~SOUNDBUFFER() { //Lock audio buffer SDL_LockAudioDevice(audioDevice); //Free buffer if (data) delete[] data; //Remove from buffer list for (SOUNDBUFFER **soundBuffer = &soundBuffers; *soundBuffer != nullptr; soundBuffer = &(*soundBuffer)->next) { if (*soundBuffer == this) { *soundBuffer = this->next; break; } } //Unlock audio buffer SDL_UnlockAudioDevice(audioDevice); } void SOUNDBUFFER::Release() { //TODO: find a better and more stable(?) way to handle this function delete this; } void SOUNDBUFFER::Lock(uint8_t **outBuffer, size_t *outSize) { SDL_LockAudioDevice(audioDevice); if (outBuffer != nullptr) *outBuffer = data; if (outSize != nullptr) *outSize = size; } void SOUNDBUFFER::Unlock() { SDL_UnlockAudioDevice(audioDevice); } void SOUNDBUFFER::SetCurrentPosition(uint32_t dwNewPosition) { SDL_LockAudioDevice(audioDevice); samplePosition = dwNewPosition; SDL_UnlockAudioDevice(audioDevice); } void SOUNDBUFFER::SetFrequency(uint32_t dwFrequency) { SDL_LockAudioDevice(audioDevice); frequency = (double)dwFrequency; SDL_UnlockAudioDevice(audioDevice); } float MillibelToVolume(int32_t lVolume) { //Volume is in hundredths of decibels, from 0 to -10000 lVolume = clamp(lVolume, (decltype(lVolume))-10000, (decltype(lVolume))0); return pow(10.0, lVolume / 2000.0); } void SOUNDBUFFER::SetVolume(int32_t lVolume) { SDL_LockAudioDevice(audioDevice); volume = MillibelToVolume(lVolume); SDL_UnlockAudioDevice(audioDevice); } void SOUNDBUFFER::SetPan(int32_t lPan) { SDL_LockAudioDevice(audioDevice); volume_l = MillibelToVolume(-lPan); volume_r = MillibelToVolume(lPan); SDL_UnlockAudioDevice(audioDevice); } void SOUNDBUFFER::Play(bool bLooping) { SDL_LockAudioDevice(audioDevice); playing = true; looping = bLooping; SDL_UnlockAudioDevice(audioDevice); } void SOUNDBUFFER::Stop() { SDL_LockAudioDevice(audioDevice); playing = false; SDL_UnlockAudioDevice(audioDevice); } void SOUNDBUFFER::Mix(float (*buffer)[2], size_t samples) { if (!playing) //This sound buffer isn't playing return; for (size_t sample = 0; sample < samples; sample++) { const double freqPosition = frequency / FREQUENCY; //This is added to position at the end //Get the in-between sample this is (linear interpolation) const float sample1 = ((looped || ((size_t)samplePosition) >= 1) ? data[(size_t)samplePosition] : 0x80); const float sample2 = ((looping || (((size_t)samplePosition) + 1) < size) ? data[(((size_t)samplePosition) + 1) % size] : 0x80); //Interpolate sample const float subPos = std::fmod(samplePosition, 1.0); const float sampleA = sample1 + (sample2 - sample1) * subPos; //Convert sample to float32 const float sampleConvert = (sampleA - 128.0f) / 128.0f; //Mix buffer[sample][0] += sampleConvert * volume * volume_l; buffer[sample][1] += sampleConvert * volume * volume_r; //Increment position samplePosition += freqPosition; if (samplePosition >= size) { if (looping) { samplePosition = std::fmod(samplePosition, size); looped = true; } else { samplePosition = 0.0; playing = false; looped = false; break; } } } } //Sound mixer void AudioCallback(void *userdata, uint8_t *stream, int len) { (void)userdata; float (*buffer)[2] = (float(*)[2])stream; const size_t samples = len / (sizeof(float) * 2); //Clear stream for (size_t sample = 0; sample < samples; ++sample) { buffer[sample][0] = 0.0f; buffer[sample][1] = 0.0f; } //Mix sounds to primary buffer for (SOUNDBUFFER *sound = soundBuffers; sound != nullptr; sound = sound->next) sound->Mix(buffer, samples); } //Sound things SOUNDBUFFER* lpSECONDARYBUFFER[SOUND_NO]; bool InitDirectSound() { SDL_AudioSpec want, have; //Set specifications we want SDL_memset(&want, 0, sizeof(want)); want.freq = FREQUENCY; want.format = AUDIO_F32; want.channels = 2; want.samples = STREAM_SIZE; want.callback = AudioCallback; audioDevice = SDL_OpenAudioDevice(NULL, 0, &want, &have, 0); if (audioDevice == 0) { printf("Failed to open audio device\nSDL Error: %s\n", SDL_GetError()); return false; } //Unpause audio device SDL_PauseAudioDevice(audioDevice, 0); //Start organya StartOrganya(); return true; } void EndDirectSound() { //Close audio device SDL_CloseAudioDevice(audioDevice); //End organya EndOrganya(); } //Sound effects playing void PlaySoundObject(int no, int mode) { if (lpSECONDARYBUFFER[no]) { if (mode == -1) { lpSECONDARYBUFFER[no]->Play(true); } else if ( mode ) { if ( mode == 1 ) { lpSECONDARYBUFFER[no]->Stop(); lpSECONDARYBUFFER[no]->SetCurrentPosition(0); lpSECONDARYBUFFER[no]->Play(false); } } else { lpSECONDARYBUFFER[no]->Stop(); } } } void ChangeSoundFrequency(int no, uint32_t rate) { if (lpSECONDARYBUFFER[no]) lpSECONDARYBUFFER[no]->SetFrequency(10 * rate + 100); } void ChangeSoundVolume(int no, int32_t volume) { if (lpSECONDARYBUFFER[no]) lpSECONDARYBUFFER[no]->SetVolume(8 * volume - 2400); } void ChangeSoundPan(int no, int32_t pan) { if (lpSECONDARYBUFFER[no]) lpSECONDARYBUFFER[no]->SetPan(10 * (pan - 256)); }