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https://github.com/bsnes-emu/bsnes.git
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This represents a major code restructuring. The dot-based and scanline-based renderers are now split into two separate core libraries, asnes and bsnes. For now at least, these are -internal- names. I'm not entirely decided on how I'm going to handle releasing these two separate builds. Regardless, the folders need names. asnes has had all of the processor subfolders collapsed back into their parent folders. In other words, ppu's functions were moved into ppu/sppu, and then ppu was deleted, and then ppu/sppu became the new ppu. Repeat this for the cpu, smp and dsp and there you go. asnes/dsp also removed the DSP_STATE_MACHINE option. This was done for the sake of consistency with the rest of the core. asnes' debugger mode is currently extremely broken, but I will be fixing it in time. And for now, bsnes has kept the processor abstraction layer. I may keep it around, not sure yet. It doesn't hurt speed or anything, so I'm not too worried about making a decision right away. I may throw snesfilter, snesreader and supergameboy into this folder, just to have everything in one place. The alternate GUI forks are definitely going in there as dotnet, cocoa and python. Compiled output goes to the out/ folder now, to prevent conflicts with a file and folder named bsnes, for instance.
210 lines
5 KiB
C++
210 lines
5 KiB
C++
/*
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audio.openal (2007-12-26)
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author: Nach
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contributors: byuu, wertigon, _willow_
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*/
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#if defined(PLATFORM_OSX)
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#include <OpenAL/al.h>
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#include <OpenAL/alc.h>
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#else
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#include <AL/al.h>
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#include <AL/alc.h>
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#endif
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namespace ruby {
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class pAudioOpenAL {
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public:
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struct {
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ALCdevice *handle;
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ALCcontext *context;
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ALuint source;
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ALenum format;
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unsigned latency;
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unsigned queue_length;
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} device;
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struct {
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uint32_t *data;
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unsigned length;
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unsigned size;
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} buffer;
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struct {
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bool synchronize;
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unsigned frequency;
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unsigned latency;
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} settings;
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bool cap(const string& name) {
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if(name == Audio::Synchronize) return true;
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if(name == Audio::Frequency) return true;
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if(name == Audio::Latency) return true;
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return false;
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}
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any get(const string& name) {
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if(name == Audio::Synchronize) return settings.synchronize;
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if(name == Audio::Frequency) return settings.frequency;
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if(name == Audio::Latency) return settings.latency;
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return false;
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}
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bool set(const string& name, const any& value) {
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if(name == Audio::Synchronize) {
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settings.synchronize = any_cast<bool>(value);
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return true;
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}
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if(name == Audio::Frequency) {
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settings.frequency = any_cast<unsigned>(value);
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return true;
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}
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if(name == Audio::Latency) {
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if(settings.latency != any_cast<unsigned>(value)) {
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settings.latency = any_cast<unsigned>(value);
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update_latency();
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}
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return true;
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}
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return false;
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}
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void sample(uint16_t sl, uint16_t sr) {
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buffer.data[buffer.length++] = sl + (sr << 16);
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if(buffer.length < buffer.size) return;
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ALuint albuffer = 0;
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int processed = 0;
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while(true) {
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alGetSourcei(device.source, AL_BUFFERS_PROCESSED, &processed);
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while(processed--) {
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alSourceUnqueueBuffers(device.source, 1, &albuffer);
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alDeleteBuffers(1, &albuffer);
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device.queue_length--;
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}
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//wait for buffer playback to catch up to sample generation if not synchronizing
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if(settings.synchronize == false || device.queue_length < 3) break;
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}
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if(device.queue_length < 3) {
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alGenBuffers(1, &albuffer);
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alBufferData(albuffer, device.format, buffer.data, buffer.size * 4, settings.frequency);
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alSourceQueueBuffers(device.source, 1, &albuffer);
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device.queue_length++;
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}
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ALint playing;
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alGetSourcei(device.source, AL_SOURCE_STATE, &playing);
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if(playing != AL_PLAYING) alSourcePlay(device.source);
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buffer.length = 0;
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}
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void clear() {
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}
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void update_latency() {
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if(buffer.data) delete[] buffer.data;
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buffer.size = settings.frequency * settings.latency / 1000.0 + 0.5;
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buffer.data = new uint32_t[buffer.size];
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}
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bool init() {
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update_latency();
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device.queue_length = 0;
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bool success = false;
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if(device.handle = alcOpenDevice(NULL)) {
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if(device.context = alcCreateContext(device.handle, NULL)) {
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alcMakeContextCurrent(device.context);
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alGenSources(1, &device.source);
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//alSourcef (device.source, AL_PITCH, 1.0);
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//alSourcef (device.source, AL_GAIN, 1.0);
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//alSource3f(device.source, AL_POSITION, 0.0, 0.0, 0.0);
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//alSource3f(device.source, AL_VELOCITY, 0.0, 0.0, 0.0);
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//alSource3f(device.source, AL_DIRECTION, 0.0, 0.0, 0.0);
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//alSourcef (device.source, AL_ROLLOFF_FACTOR, 0.0);
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//alSourcei (device.source, AL_SOURCE_RELATIVE, AL_TRUE);
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alListener3f(AL_POSITION, 0.0, 0.0, 0.0);
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alListener3f(AL_VELOCITY, 0.0, 0.0, 0.0);
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ALfloat listener_orientation[] = { 0.0, 0.0, 0.0, 0.0, 0.0, 0.0 };
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alListenerfv(AL_ORIENTATION, listener_orientation);
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success = true;
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}
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}
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if(success == false) {
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term();
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return false;
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}
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return true;
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}
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void term() {
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if(alIsSource(device.source) == AL_TRUE) {
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int playing = 0;
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alGetSourcei(device.source, AL_SOURCE_STATE, &playing);
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if(playing == AL_PLAYING) {
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alSourceStop(device.source);
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int queued = 0;
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alGetSourcei(device.source, AL_BUFFERS_QUEUED, &queued);
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while(queued--) {
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ALuint albuffer = 0;
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alSourceUnqueueBuffers(device.source, 1, &albuffer);
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alDeleteBuffers(1, &albuffer);
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device.queue_length--;
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}
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}
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alDeleteSources(1, &device.source);
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device.source = 0;
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}
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if(device.context) {
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alcMakeContextCurrent(NULL);
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alcDestroyContext(device.context);
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device.context = 0;
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}
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if(device.handle) {
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alcCloseDevice(device.handle);
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device.handle = 0;
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}
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if(buffer.data) {
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delete[] buffer.data;
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buffer.data = 0;
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}
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}
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pAudioOpenAL() {
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device.source = 0;
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device.handle = 0;
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device.context = 0;
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device.format = AL_FORMAT_STEREO16;
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device.queue_length = 0;
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buffer.data = 0;
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buffer.length = 0;
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buffer.size = 0;
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settings.synchronize = true;
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settings.frequency = 22050;
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settings.latency = 40;
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}
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~pAudioOpenAL() {
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term();
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}
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};
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DeclareAudio(OpenAL)
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};
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