| #include <stdio.h> |
| #include <stdlib.h> |
| #include <memory.h> |
| #include <math.h> |
| #include "sound.h" |
| |
| #define SOUND_BUFSIZE_MSEC 20 |
| // Linux needs a buffer with a size of a factor of 2 |
| // 512 1024 2048 4096 |
| #define FRAGMENT_SIZE (int(double(sampleFrq * SOUND_BUFSIZE_MSEC) / 1000.0 / 1024.0 + 0.5 ) * 1024) |
| |
| //#define LOG_AUDIO |
| #define SND_BUF_MAX_READ_AHEAD 6 |
| #define SND_LATENCY_IN_FRAGS 2 |
| |
| static SDL_AudioDeviceID dev; |
| static SDL_AudioSpec obtained, *audiohwspec; |
| |
| static int MixingFreq; |
| static unsigned int BufferLength; |
| static unsigned int sndBufferPos; |
| |
| static short *mixingBuffer; |
| static short *sndRingBuffer; |
| static unsigned int sndWriteBufferIndex; |
| static unsigned int sndPlayBufferIndex; |
| static short *sndWriteBufferPtr; |
| static short *sndPlayBufferPtr; |
| static short lastSample; |
| static ClockCycle lastUpdateCycle; |
| static unsigned int lastSamplePos; |
| |
| template<> unsigned int LinkedList<SoundSource>::count = 0; |
| template<> SoundSource* LinkedList<SoundSource>::root = 0; |
| template<> SoundSource* LinkedList<SoundSource>::last = 0; |
| unsigned int SoundSource::sampleRate = SAMPLE_FREQ; |
| |
| void SoundSource::bufferFill(unsigned int nrsamples, short *buffer) |
| { |
| SoundSource *cb = SoundSource::getRoot(); |
| if (cb) { |
| cb->calcSamples(buffer, nrsamples); |
| cb = cb->getNext(); |
| } |
| // multiple sources |
| while (cb) { |
| int i = nrsamples - 1; |
| cb->calcSamples(mixingBuffer, nrsamples); |
| do { |
| buffer[i] += mixingBuffer[i]; |
| } while(i--); |
| cb = cb->getNext(); |
| } |
| } |
| |
| static void add_new_frag() |
| { |
| sndWriteBufferIndex++; |
| sndWriteBufferPtr = sndRingBuffer |
| + (BufferLength * (sndWriteBufferIndex % SND_BUF_MAX_READ_AHEAD)); |
| } |
| |
| static void delete_frag() |
| { |
| sndPlayBufferIndex++; |
| sndPlayBufferPtr = sndRingBuffer |
| + (BufferLength * (sndPlayBufferIndex % SND_BUF_MAX_READ_AHEAD)); |
| } |
| |
| static int getLeadInFrags() |
| { |
| return (int) (sndWriteBufferIndex - sndPlayBufferIndex); |
| } |
| |
| static void fragmentDone() |
| { |
| int lead_in_frags = getLeadInFrags(); |
| #ifdef LOG_AUDIO |
| fprintf(stderr, "Lead in frags: %i\n", lead_in_frags); |
| #endif |
| |
| while (lead_in_frags < SND_LATENCY_IN_FRAGS) { |
| #ifdef LOG_AUDIO |
| fprintf(stderr, " adding an extra frag.\n"); |
| #endif |
| SoundSource::bufferFill(BufferLength, sndWriteBufferPtr); |
| add_new_frag(); |
| lead_in_frags++; |
| } |
| } |
| |
| static void audioCallback(void *userdata, Uint8 *stream, int len) |
| { |
| if (sndPlayBufferIndex < sndWriteBufferIndex) { |
| if (len > (int)(BufferLength*2)) |
| len = BufferLength * 2; |
| memcpy(stream, sndPlayBufferPtr, len); |
| lastSample = sndPlayBufferPtr[len/2 - 1]; |
| delete_frag(); |
| } else { |
| short *buf = (short *) stream; |
| len /= 2; |
| do { |
| *buf++ = lastSample; |
| } while(--len); |
| } |
| #ifdef LOG_AUDIO |
| fprintf(stderr, "Playing a frag (%i).\n", getLeadInFrags()); |
| #endif |
| } |
| |
| void updateAudio(unsigned int nrsamples) |
| { |
| // SDL openaudio failed? |
| if (!sndWriteBufferPtr) |
| return; |
| |
| if (sndBufferPos + nrsamples >= BufferLength) { |
| if (getLeadInFrags() > SND_LATENCY_IN_FRAGS) { |
| #ifdef LOG_AUDIO |
| fprintf(stderr, "Skipping a frag.\n"); |
| #endif |
| } else { |
| // Finish pending buffer... |
| SoundSource::bufferFill(BufferLength - sndBufferPos, sndWriteBufferPtr + sndBufferPos); |
| add_new_frag(); |
| if ((sndBufferPos = (sndBufferPos + nrsamples) % BufferLength)) |
| SoundSource::bufferFill(sndBufferPos, sndWriteBufferPtr); |
| fragmentDone(); |
| } |
| } else if (nrsamples) { |
| SoundSource::bufferFill(nrsamples, sndWriteBufferPtr + sndBufferPos); |
| sndBufferPos += nrsamples; |
| } |
| //_ASSERT(sndBufferPos <= BufferLength); |
| } |
| |
| static inline unsigned int getNrOfSamplesToGenerate(ClockCycle clock, unsigned int deviceFrq) |
| { |
| // OK this should really be INT but I'm tired right now |
| unsigned int samplePos = (unsigned int) ((double) clock * (double) MixingFreq / deviceFrq); |
| unsigned int samplesToDo = samplePos - lastSamplePos; |
| //fprintf( stderr, "Sound: %i cycles/%f samples\n", clock, (double) clock * (double) MixingFreq / deviceFrq); |
| // 'clock' might have been reset but 'lastSamplePos' not! |
| if (lastSamplePos > samplePos) |
| samplesToDo = 0; |
| lastSamplePos = samplePos; |
| return samplesToDo; |
| } |
| |
| void flushBuffer(ClockCycle cycle, unsigned int frq) |
| { |
| updateAudio(getNrOfSamplesToGenerate(cycle, frq)); |
| lastUpdateCycle = cycle; |
| } |
| |
| void init_audio(unsigned int sampleFrq) |
| { |
| SDL_AudioSpec desired; |
| |
| MixingFreq = sampleFrq; |
| |
| BufferLength = FRAGMENT_SIZE; |
| |
| desired.freq = MixingFreq; |
| desired.format = AUDIO_S16; |
| desired.channels = 1; |
| desired.samples = BufferLength; |
| desired.callback = audioCallback; |
| desired.userdata = NULL; |
| desired.size = desired.channels * desired.samples * sizeof(Uint8); |
| desired.silence = 0x00; |
| |
| mixingBuffer = new short[FRAGMENT_SIZE]; |
| if (!mixingBuffer) |
| return; |
| dev = SDL_OpenAudioDevice(NULL, 0, &desired, &obtained, SDL_AUDIO_ALLOW_FREQUENCY_CHANGE); |
| if (!dev) { |
| fprintf(stderr, "SDL_OpenAudioDevice failed!\n"); |
| return; |
| } else { |
| fprintf(stderr, "SDL_OpenAudioDevice success!\n"); |
| fprintf(stderr, "Using audio driver : %s\n", SDL_GetCurrentAudioDriver()); |
| audiohwspec = &obtained; |
| } |
| MixingFreq = audiohwspec->freq; |
| BufferLength = audiohwspec->samples; |
| |
| fprintf(stderr, "Obtained mixing frequency: %u\n", audiohwspec->freq); |
| fprintf(stderr, "Obtained audio format: %04X\n", audiohwspec->format); |
| fprintf(stderr, "Obtained channel number: %u\n", audiohwspec->channels); |
| fprintf(stderr, "Obtained audio buffer size: %u\n", audiohwspec->size); |
| fprintf(stderr, "Obtained sample buffer size: %u\n", audiohwspec->samples); |
| fprintf(stderr, "Obtained silence value: %u\n", audiohwspec->silence); |
| |
| sndRingBuffer = new short[SND_BUF_MAX_READ_AHEAD * BufferLength]; |
| for(unsigned int i = 0; i < SND_BUF_MAX_READ_AHEAD * BufferLength; i++) |
| sndRingBuffer[i] = audiohwspec->silence; |
| sndWriteBufferIndex = SND_LATENCY_IN_FRAGS; |
| sndPlayBufferIndex = 0; |
| sndWriteBufferPtr = sndRingBuffer + BufferLength * SND_LATENCY_IN_FRAGS; |
| sndPlayBufferPtr = sndRingBuffer; |
| |
| sndBufferPos = 0; |
| lastSample = 0; |
| lastUpdateCycle = 0; |
| lastSamplePos = 0; |
| SDL_PauseAudioDevice(dev, 0); |
| } |
| |
| void sound_pause() |
| { |
| SDL_PauseAudioDevice(dev, 1); |
| } |
| |
| void sound_resume() |
| { |
| SDL_PauseAudioDevice(dev, 0); |
| } |
| |
| void sound_change_freq(unsigned int &newFreq) |
| { |
| close_audio(); |
| SoundSource *cb = SoundSource::getRoot(); |
| while (cb) { |
| cb->setSampleRate(newFreq); |
| cb = cb->getNext(); |
| } |
| init_audio(newFreq); |
| newFreq = audiohwspec->freq; |
| } |
| |
| void close_audio() |
| { |
| SDL_CloseAudioDevice(dev); |
| delete[] sndRingBuffer; |
| delete[] mixingBuffer; |
| } |