mirror of https://github.com/bsnes-emu/bsnes.git
Update to v098r05 release.
byuu says: Changelog: - WS/WSC: re-added support for screen rotation (code is inside WS core) - ruby: changed sample(uint16_t left, uint16_t right) to sample(int16_t left, int16_t right); - requires casting to uint prior to shifting in each driver, but I felt it was misleading to use uint16_t just to avoid that - ruby: WASAPI is now built in by default; has wareya's improvements, and now supports latency adjust - tomoko: audio settings panel has new "Exclusive Mode" checkbox for WASAPI driver only - note: although the setting *does* take effect in real-time, I'd suggest restarting the emulator after changing it - tomoko: audio latency can now be set to 0ms (which in reality means "the minimum supported by the driver") - all: increased cothread size from 512KiB to 2MiB to see if it fixes bullshit AMD driver crashes - this appears to cause a slight speed penalty due to cache locality going down between threads, though
This commit is contained in:
parent
a2d3b8ba15
commit
55e507d5df
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@ -8,7 +8,7 @@ using namespace nall;
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namespace Emulator {
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static const string Name = "higan";
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static const string Version = "098.04";
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static const string Version = "098.05";
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static const string Author = "byuu";
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static const string License = "GPLv3";
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static const string Website = "http://byuu.org/";
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@ -22,7 +22,7 @@ namespace Famicom {
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auto create(auto (*entrypoint)() -> void, uint frequency) -> void {
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if(thread) co_delete(thread);
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thread = co_create(65536 * sizeof(void*), entrypoint);
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thread = co_create(262'144 * sizeof(void*), entrypoint);
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this->frequency = frequency;
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clock = 0;
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}
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@ -22,7 +22,7 @@ namespace GameBoy {
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auto create(auto (*entrypoint)() -> void, uint frequency) -> void {
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if(thread) co_delete(thread);
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thread = co_create(65536 * sizeof(void*), entrypoint);
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thread = co_create(262'144 * sizeof(void*), entrypoint);
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this->frequency = frequency;
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clock = 0;
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}
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@ -34,7 +34,7 @@ namespace GameBoyAdvance {
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auto create(auto (*entrypoint)() -> void, uint frequency) -> void {
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if(thread) co_delete(thread);
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thread = co_create(65536 * sizeof(void*), entrypoint);
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thread = co_create(262'144 * sizeof(void*), entrypoint);
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this->frequency = frequency;
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clock = 0;
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}
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@ -31,7 +31,7 @@ namespace SuperFamicom {
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auto create(auto (*entrypoint)() -> void, uint frequency) -> void {
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if(thread) co_delete(thread);
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thread = co_create(65536 * sizeof(void*), entrypoint);
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thread = co_create(262'144 * sizeof(void*), entrypoint);
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this->frequency = frequency;
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clock = 0;
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}
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@ -16,7 +16,7 @@ ui_objects += $(if $(call streq,$(platform),windows),ui-resource)
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# platform
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ifeq ($(platform),windows)
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ruby += video.direct3d video.wgl video.directdraw video.gdi
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ruby += audio.xaudio2 audio.directsound
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ruby += audio.wasapi audio.xaudio2 audio.directsound
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ruby += input.windows
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else ifeq ($(platform),macosx)
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ruby += video.cgl
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@ -29,6 +29,7 @@ Settings::Settings() {
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set("Audio/Driver", ruby::Audio::optimalDriver());
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set("Audio/Device", "");
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set("Audio/Exclusive", false);
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set("Audio/Synchronize", true);
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set("Audio/Mute", false);
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set("Audio/Volume", 100);
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@ -43,7 +43,7 @@ Program::Program(lstring args) {
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dsp.setPrecision(16);
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dsp.setBalance(0.0);
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dsp.setFrequency(32040);
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dsp.setFrequency(44100);
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dsp.setResampler(DSP::ResampleEngine::Sinc);
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dsp.setResamplerFrequency(audio->get(Audio::Frequency).get<uint>());
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@ -55,7 +55,7 @@ Program::Program(lstring args) {
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new ToolsManager;
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updateVideoShader();
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updateAudioVolume();
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updateAudio();
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args.takeFirst(); //ignore program location in argument parsing
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for(auto& argument : args) {
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@ -35,6 +35,7 @@ struct Program : Emulator::Interface::Bind {
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auto updateStatusText() -> void;
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auto updateVideoShader() -> void;
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auto updateAudio() -> void;
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auto updateAudioMode() -> void;
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auto updateAudioVolume() -> void;
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auto updateDSP() -> void;
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@ -71,10 +71,15 @@ auto Program::updateAudio() -> void {
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if(settings["Audio/Resampler"].text() == "Hermite") dsp.setResampler(DSP::ResampleEngine::Hermite);
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if(settings["Audio/Resampler"].text() == "Sinc" ) dsp.setResampler(DSP::ResampleEngine::Sinc);
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dsp.setResamplerFrequency(audio->get(Audio::Frequency).get<uint>());
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updateAudioMode();
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updateAudioVolume();
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updateDSP();
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}
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auto Program::updateAudioMode() -> void {
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audio->set(Audio::Exclusive, settings["Audio/Exclusive"].boolean());
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}
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auto Program::updateAudioVolume() -> void {
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dsp.setVolume(settings["Audio/Mute"].boolean() ? 0.0 : settings["Audio/Volume"].natural() * 0.01);
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}
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@ -10,17 +10,19 @@ AudioSettings::AudioSettings(TabFrame* parent) : TabFrameItem(parent) {
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frequencyCombo.setEnabled(false); //not user configurable
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latencyLabel.setText("Latency:");
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latencyCombo.append(ComboButtonItem().setText("0ms"));
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latencyCombo.append(ComboButtonItem().setText("20ms"));
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latencyCombo.append(ComboButtonItem().setText("40ms"));
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latencyCombo.append(ComboButtonItem().setText("60ms"));
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latencyCombo.append(ComboButtonItem().setText("80ms"));
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latencyCombo.append(ComboButtonItem().setText("100ms"));
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switch(settings["Audio/Latency"].natural()) {
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case 20: latencyCombo.item(0)->setSelected(); break;
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case 40: latencyCombo.item(1)->setSelected(); break;
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case 60: latencyCombo.item(2)->setSelected(); break;
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case 80: latencyCombo.item(3)->setSelected(); break;
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case 100: latencyCombo.item(4)->setSelected(); break;
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case 0: latencyCombo.item(0)->setSelected(); break;
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case 20: latencyCombo.item(1)->setSelected(); break;
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case 40: latencyCombo.item(2)->setSelected(); break;
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case 60: latencyCombo.item(3)->setSelected(); break;
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case 80: latencyCombo.item(4)->setSelected(); break;
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case 100: latencyCombo.item(5)->setSelected(); break;
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}
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latencyCombo.onChange([&] { update(); });
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@ -36,17 +38,21 @@ AudioSettings::AudioSettings(TabFrame* parent) : TabFrameItem(parent) {
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volumeLabel.setText("Volume:");
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volumeSlider.setLength(201).setPosition(settings["Audio/Volume"].natural()).onChange([&] { updateVolume(); });
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exclusiveMode.setText("Exclusive Mode").setVisible(audio->cap(Audio::Exclusive));
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exclusiveMode.setChecked(settings["Audio/Exclusive"].boolean()).onToggle([&] { updateMode(); });
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update();
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}
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auto AudioSettings::update() -> void {
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if(auto item = latencyCombo.selected()) {
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uint latency = 60;
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if(item->offset() == 0) latency = 20;
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if(item->offset() == 1) latency = 40;
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if(item->offset() == 2) latency = 60;
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if(item->offset() == 3) latency = 80;
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if(item->offset() == 4) latency = 100;
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if(item->offset() == 0) latency = 0;
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if(item->offset() == 1) latency = 20;
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if(item->offset() == 2) latency = 40;
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if(item->offset() == 3) latency = 60;
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if(item->offset() == 4) latency = 80;
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if(item->offset() == 5) latency = 100;
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settings["Audio/Latency"].setValue(latency);
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}
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if(auto item = resamplerCombo.selected()) {
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settings["Audio/Resampler"].setValue(resampler);
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}
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updateVolume();
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program->updateAudio();
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updateMode();
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}
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auto AudioSettings::updateMode() -> void {
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settings["Audio/Exclusive"].setValue(exclusiveMode.checked());
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program->updateAudioMode();
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}
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auto AudioSettings::updateVolume() -> void {
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@ -30,8 +30,10 @@ struct AudioSettings : TabFrameItem {
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Label volumeLabel{&volumeLayout, Size{80, 0}};
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Label volumeValue{&volumeLayout, Size{80, 0}};
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HorizontalSlider volumeSlider{&volumeLayout, Size{~0, 0}};
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CheckLabel exclusiveMode{&layout, Size{~0, 0}};
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auto update() -> void;
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auto updateMode() -> void;
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auto updateVolume() -> void;
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};
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@ -11,7 +11,7 @@ Interface::Interface() {
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information.manufacturer = "Bandai";
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information.name = "WonderSwan";
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information.width = 224; //note: technically 224x144; but screen can be rotated
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information.height = 144; //by using a square size; this can be done in the core
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information.height = 224; //by using a square size; this can be done in the core
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information.overscan = false;
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information.aspectRatio = 1.0;
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information.resettable = false;
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@ -28,7 +28,10 @@ auto PPU::main() -> void {
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if(l.screenTwoEnable) renderScreenTwo();
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if(l.spriteEnable) renderSprite();
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}
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output[s.vclk * 224 + s.hclk] = s.pixel.color;
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switch(l.orientation) {
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case 0: output[(s.vclk + 40) * 224 + s.hclk] = s.pixel.color; break;
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case 1: output[(223 - s.hclk) * 224 + (s.vclk + 40)] = s.pixel.color; break;
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}
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step(1);
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}
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step(32);
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auto PPU::frame() -> void {
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s.field = !s.field;
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s.vclk = 0;
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Emulator::video.refresh(output, 224 * sizeof(uint32), 224, 144);
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Emulator::video.refresh(output, 224 * sizeof(uint32), 224, 224);
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if(l.orientation != system.orientation()) {
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l.orientation = system.orientation();
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memory::fill(output, 224 * 224 * sizeof(uint32));
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}
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scheduler.exit(Scheduler::Event::Frame);
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}
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@ -34,7 +34,7 @@ struct PPU : Thread, IO {
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uint12 color;
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};
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uint32 output[224 * 144];
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uint32 output[224 * 224];
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struct State {
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bool field;
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} s;
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struct Latches {
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//frame(), power()
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bool orientation;
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//latchRegisters()
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uint8 backColor;
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@ -7,6 +7,7 @@ auto PPU::serialize(serializer& s) -> void {
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s.integer((uint&)this->s.pixel.source);
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s.integer(this->s.pixel.color);
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s.integer(l.orientation);
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s.integer(l.backColor);
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s.integer(l.screenOneEnable);
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s.integer(l.screenOneMapBase);
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@ -30,7 +30,7 @@ namespace WonderSwan {
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auto create(auto (*entrypoint)() -> void, uint frequency) -> void {
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if(thread) co_delete(thread);
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thread = co_create(65536 * sizeof(void*), entrypoint);
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thread = co_create(262'144 * sizeof(void*), entrypoint);
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this->frequency = frequency;
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clock = 0;
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}
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@ -65,10 +65,10 @@ struct AudioALSA : Audio {
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return false;
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}
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auto sample(uint16_t left, uint16_t right) -> void {
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auto sample(int16_t left, int16_t right) -> void {
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if(!device.handle) return;
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buffer.data[buffer.length++] = left + (right << 16);
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buffer.data[buffer.length++] = (uint16_t)left << 0 | (uint16_t)right << 16;
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if(buffer.length < device.period_size) return;
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snd_pcm_sframes_t avail;
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@ -31,8 +31,8 @@ struct AudioAO : Audio {
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return false;
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}
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auto sample(uint16_t l_sample, uint16_t r_sample) -> void {
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uint32_t samp = (l_sample << 0) + (r_sample << 16);
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auto sample(int16_t left, int16_t right) -> void {
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uint32_t samp = (uint16_t)left << 0 | (uint16_t)right << 0;
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ao_play(audio_device, (char*)&samp, 4); //This may need to be byte swapped for Big Endian
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}
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@ -72,8 +72,8 @@ struct AudioDS : Audio {
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return false;
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}
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auto sample(uint16_t left, uint16_t right) -> void {
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device.buffer[device.bufferoffset++] = left + (right << 16);
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auto sample(int16_t left, int16_t right) -> void {
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device.buffer[device.bufferoffset++] = (uint16_t)left << 0 | (uint16_t)right << 16;
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if(device.bufferoffset < device.latency) return;
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device.bufferoffset = 0;
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@ -66,8 +66,8 @@ struct AudioOpenAL : Audio {
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return false;
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}
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auto sample(uint16_t left, uint16_t right) -> void {
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buffer.data[buffer.length++] = left << 0 | right << 16;
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auto sample(int16_t left, int16_t right) -> void {
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buffer.data[buffer.length++] = (uint16_t)left << 0 | (uint16_t)right << 16;
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if(buffer.length < buffer.size) return;
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ALuint albuffer = 0;
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@ -69,8 +69,8 @@ struct AudioOSS : Audio {
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return false;
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}
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auto sample(uint16_t left, uint16_t right) -> void {
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uint32_t sample = left << 0 | right << 16;
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auto sample(int16_t left, int16_t right) -> void {
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uint32_t sample = (uint16_t)left << 0 | (uint16_t)right << 16;
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auto unused = write(device.fd, &sample, 4);
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}
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@ -64,9 +64,9 @@ struct AudioPulseAudio : Audio {
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return false;
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}
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auto sample(uint16_t left, uint16_t right) -> void {
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auto sample(int16_t left, int16_t right) -> void {
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pa_stream_begin_write(device.stream, (void**)&buffer.data, &buffer.size);
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buffer.data[buffer.offset++] = left + (right << 16);
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buffer.data[buffer.offset++] = (uint16_t)left << 0 | (uint16_t)right << 16;
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if((buffer.offset + 1) * pa_frame_size(&device.spec) <= buffer.size) return;
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while(true) {
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@ -38,10 +38,10 @@ struct AudioPulseAudioSimple : Audio {
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return false;
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}
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auto sample(uint16_t left, uint16_t right) -> void {
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auto sample(int16_t left, int16_t right) -> void {
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if(!device.handle) return;
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buffer.data[buffer.offset++] = left + (right << 16);
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buffer.data[buffer.offset++] = (uint16_t)left << 0 | (uint16_t)right << 16;
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if(buffer.offset >= 64) {
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int error;
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pa_simple_write(device.handle, (const void*)buffer.data, buffer.offset * sizeof(uint32_t), &error);
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@ -5,19 +5,25 @@
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#include <devicetopology.h>
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#include <endpointvolume.h>
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#include <nall/dsp.hpp>
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struct AudioWASAPI : Audio {
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~AudioWASAPI() { term(); }
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struct {
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bool exclusive = false;
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bool synchronize = false;
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uint frequency = 48000;
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uint latency = 80;
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bool synchronize = true;
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} settings;
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struct {
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uint channels = 0;
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uint frequency = 0;
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uint mode = 0;
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uint precision = 0;
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} device;
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auto cap(const string& name) -> bool {
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if(name == Audio::Exclusive) return true;
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if(name == Audio::Latency) return true;
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if(name == Audio::Synchronize) return true;
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if(name == Audio::Frequency) return true;
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return false;
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@ -25,145 +31,132 @@ struct AudioWASAPI : Audio {
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auto get(const string& name) -> any {
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if(name == Audio::Exclusive) return settings.exclusive;
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if(name == Audio::Latency) return settings.latency;
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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::Frequency) return device.frequency;
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return {};
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}
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auto set(const string& name, const any& value) -> bool {
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if(name == Audio::Exclusive && value.get<bool>()) {
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if(audioDevice) term(), init();
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settings.exclusive = value.get<bool>();
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return true;
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}
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if(name == Audio::Latency && value.get<uint>()) {
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if(audioDevice) term(), init();
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settings.latency = value.get<uint>();
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return true;
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}
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if(name == Audio::Synchronize && value.is<bool>()) {
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settings.synchronize = value.get<bool>();
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return true;
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}
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if(name == Audio::Frequency && value.is<uint>()) {
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settings.frequency = value.get<uint>();
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dsp.setFrequency(settings.frequency);
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return true;
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}
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return false;
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}
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auto sample(uint16 left, uint16 right) -> void {
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int samples[] = {(int16)left, (int16)right};
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dsp.sample(samples);
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while(dsp.pending()) {
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dsp.read(samples);
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write(samples);
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auto sample(int16_t left, int16_t right) -> void {
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queuedFrames.append((uint16_t)left << 0 | (uint16_t)right << 16);
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if(!available() && queuedFrames.size() >= bufferSize) {
|
||||
if(settings.synchronize) while(!available()); //wait for free sample slot
|
||||
else queuedFrames.takeFirst(); //drop sample (run ahead)
|
||||
}
|
||||
|
||||
uint32_t cachedFrame = 0;
|
||||
for(auto n : range(available())) {
|
||||
if(queuedFrames) cachedFrame = queuedFrames.takeFirst();
|
||||
write(cachedFrame >> 0, cachedFrame >> 16);
|
||||
}
|
||||
}
|
||||
|
||||
auto clear() -> void {
|
||||
audioClient->Stop();
|
||||
renderClient->GetBuffer(bufferFrameCount, &bufferData);
|
||||
|
||||
renderClient->ReleaseBuffer(bufferFrameCount, 0);
|
||||
audioClient->Reset();
|
||||
for(auto n : range(available())) write(0, 0);
|
||||
audioClient->Start();
|
||||
}
|
||||
|
||||
auto init() -> bool {
|
||||
if(CoCreateInstance(CLSID_MMDeviceEnumerator, nullptr, CLSCTX_ALL, IID_IMMDeviceEnumerator, (void**)&enumerator) != S_OK) return false;
|
||||
if(enumerator->GetDefaultAudioEndpoint(eRender, eConsole, &device) != S_OK) return false;
|
||||
if(device->Activate(IID_IAudioClient, CLSCTX_ALL, nullptr, (void**)&audioClient) != S_OK) return false;
|
||||
if(enumerator->GetDefaultAudioEndpoint(eRender, eConsole, &audioDevice) != S_OK) return false;
|
||||
if(audioDevice->Activate(IID_IAudioClient, CLSCTX_ALL, nullptr, (void**)&audioClient) != S_OK) return false;
|
||||
|
||||
if(settings.exclusive) {
|
||||
if(device->OpenPropertyStore(STGM_READ, &propertyStore) != S_OK) return false;
|
||||
if(audioDevice->OpenPropertyStore(STGM_READ, &propertyStore) != S_OK) return false;
|
||||
if(propertyStore->GetValue(PKEY_AudioEngine_DeviceFormat, &propVariant) != S_OK) return false;
|
||||
waveFormat = (WAVEFORMATEX*)propVariant.blob.pBlobData;
|
||||
if(audioClient->GetDevicePeriod(nullptr, &devicePeriod) != S_OK) return false;
|
||||
if(audioClient->Initialize(AUDCLNT_SHAREMODE_EXCLUSIVE, 0, devicePeriod, devicePeriod, waveFormat, nullptr) != S_OK) return false;
|
||||
auto latency = max(devicePeriod, (REFERENCE_TIME)settings.latency * 10'000); //1ms to 100ns units
|
||||
if(audioClient->Initialize(AUDCLNT_SHAREMODE_EXCLUSIVE, 0, latency, latency, waveFormat, nullptr) != S_OK) return false;
|
||||
DWORD taskIndex = 0;
|
||||
taskHandle = AvSetMmThreadCharacteristics(L"Pro Audio", &taskIndex);
|
||||
} else {
|
||||
if(audioClient->GetMixFormat(&waveFormat) != S_OK) return false;
|
||||
if(audioClient->GetDevicePeriod(&devicePeriod, nullptr)) return false;
|
||||
if(audioClient->Initialize(AUDCLNT_SHAREMODE_SHARED, 0, devicePeriod, 0, waveFormat, nullptr) != S_OK) return false;
|
||||
auto latency = max(devicePeriod, (REFERENCE_TIME)settings.latency * 10'000); //1ms to 100ns units
|
||||
if(audioClient->Initialize(AUDCLNT_SHAREMODE_SHARED, 0, latency, 0, waveFormat, nullptr) != S_OK) return false;
|
||||
}
|
||||
|
||||
if(audioClient->GetService(IID_IAudioRenderClient, (void**)&renderClient) != S_OK) return false;
|
||||
if(audioClient->GetBufferSize(&bufferFrameCount) != S_OK) return false;
|
||||
if(audioClient->GetBufferSize(&bufferSize) != S_OK) return false;
|
||||
|
||||
switch(((WAVEFORMATEXTENSIBLE*)waveFormat)->SubFormat.Data1) {
|
||||
case 1: ieee = false; break; //fixed point
|
||||
case 3: ieee = true; break; //floating point
|
||||
default: return false; //unknown format; abort
|
||||
}
|
||||
|
||||
dsp.setChannels(2);
|
||||
dsp.setPrecision(16);
|
||||
dsp.setFrequency(settings.frequency);
|
||||
|
||||
dsp.setResampler(DSP::ResampleEngine::Linear);
|
||||
dsp.setResamplerFrequency(waveFormat->nSamplesPerSec);
|
||||
dsp.setChannels(waveFormat->nChannels);
|
||||
dsp.setPrecision(waveFormat->wBitsPerSample);
|
||||
|
||||
print("[WASAPI]\n");
|
||||
print("Channels: ", waveFormat->nChannels, "\n");
|
||||
print("Precision: ", waveFormat->wBitsPerSample, "\n");
|
||||
print("Frequency: ", waveFormat->nSamplesPerSec, "\n");
|
||||
print("IEEE-754: ", ieee, "\n");
|
||||
print("Exclusive: ", settings.exclusive, "\n\n");
|
||||
device.channels = waveFormat->nChannels;
|
||||
device.frequency = waveFormat->nSamplesPerSec;
|
||||
device.mode = ((WAVEFORMATEXTENSIBLE*)waveFormat)->SubFormat.Data1;
|
||||
device.precision = waveFormat->wBitsPerSample;
|
||||
|
||||
audioClient->Start();
|
||||
return true;
|
||||
}
|
||||
|
||||
auto term() -> void {
|
||||
if(audioClient) {
|
||||
audioClient->Stop();
|
||||
}
|
||||
|
||||
if(taskHandle) {
|
||||
AvRevertMmThreadCharacteristics(taskHandle);
|
||||
taskHandle = nullptr;
|
||||
}
|
||||
if(audioClient) audioClient->Stop();
|
||||
if(renderClient) renderClient->Release(), renderClient = nullptr;
|
||||
if(waveFormat) CoTaskMemFree(waveFormat), waveFormat = nullptr;
|
||||
if(audioClient) audioClient->Release(), audioClient = nullptr;
|
||||
if(audioDevice) audioDevice->Release(), audioDevice = nullptr;
|
||||
if(taskHandle) AvRevertMmThreadCharacteristics(taskHandle), taskHandle = nullptr;
|
||||
}
|
||||
|
||||
private:
|
||||
auto write(int samples[]) -> void {
|
||||
while(true) {
|
||||
uint32 padding = 0;
|
||||
audioClient->GetCurrentPadding(&padding);
|
||||
if(bufferFrameCount - padding < 1) {
|
||||
if(!settings.synchronize) return;
|
||||
continue;
|
||||
}
|
||||
break;
|
||||
auto available() -> uint {
|
||||
uint32_t padding = 0;
|
||||
audioClient->GetCurrentPadding(&padding);
|
||||
return bufferSize - padding;
|
||||
}
|
||||
|
||||
auto write(int16_t left, int16_t right) -> void {
|
||||
if(renderClient->GetBuffer(1, &bufferData) != S_OK) return;
|
||||
|
||||
if(device.channels >= 2 && device.mode == 1 && device.precision == 16) {
|
||||
auto buffer = (int16_t*)bufferData;
|
||||
buffer[0] = left;
|
||||
buffer[1] = right;
|
||||
}
|
||||
|
||||
renderClient->GetBuffer(1, &bufferData);
|
||||
|
||||
if(ieee) {
|
||||
if(device.channels >= 2 && device.mode == 3 && device.precision == 32) {
|
||||
auto buffer = (float*)bufferData;
|
||||
buffer[0] = (int16)samples[0] / 32768.0;
|
||||
buffer[1] = (int16)samples[1] / 32768.0;
|
||||
} else {
|
||||
auto buffer = (int16*)bufferData;
|
||||
buffer[0] = (int16)samples[0];
|
||||
buffer[1] = (int16)samples[1];
|
||||
buffer[0] = left / 32768.0;
|
||||
buffer[1] = right / 32768.0;
|
||||
}
|
||||
|
||||
renderClient->ReleaseBuffer(1, 0);
|
||||
}
|
||||
|
||||
DSP dsp;
|
||||
IMMDeviceEnumerator* enumerator = nullptr;
|
||||
IMMDevice* device = nullptr;
|
||||
IMMDevice* audioDevice = nullptr;
|
||||
IPropertyStore* propertyStore = nullptr;
|
||||
IAudioClient* audioClient = nullptr;
|
||||
IAudioRenderClient* renderClient = nullptr;
|
||||
WAVEFORMATEX* waveFormat = nullptr;
|
||||
PROPVARIANT propVariant;
|
||||
HANDLE taskHandle = nullptr;
|
||||
DWORD taskIndex = 0;
|
||||
REFERENCE_TIME devicePeriod = 0;
|
||||
uint32 bufferFrameCount = 0;
|
||||
uint8* bufferData = nullptr;
|
||||
bool ieee = false;
|
||||
uint32_t bufferSize = 0; //in frames
|
||||
uint8_t* bufferData = nullptr;
|
||||
vector<uint32_t> queuedFrames;
|
||||
};
|
||||
|
|
|
@ -78,8 +78,8 @@ struct AudioXAudio2 : Audio, public IXAudio2VoiceCallback {
|
|||
pSourceVoice->SubmitSourceBuffer(&xa2buffer);
|
||||
}
|
||||
|
||||
auto sample(uint16_t left, uint16_t right) -> void {
|
||||
device.buffer[device.writebuffer * device.latency + device.bufferoffset++] = left + (right << 16);
|
||||
auto sample(int16_t left, int16_t right) -> void {
|
||||
device.buffer[device.writebuffer * device.latency + device.bufferoffset++] = (uint16_t)left << 0 | (uint16_t)right << 16;
|
||||
if(device.bufferoffset < device.latency) return;
|
||||
device.bufferoffset = 0;
|
||||
|
||||
|
|
|
@ -1,12 +1,6 @@
|
|||
/*
|
||||
xaudio2.hpp (2010-08-14)
|
||||
author: OV2
|
||||
|
||||
ruby-specific header to provide mingw-friendly xaudio2 interfaces
|
||||
*/
|
||||
#pragma once
|
||||
|
||||
#ifndef XAUDIO2_RUBY_H
|
||||
#define XAUDIO2_RUBY_H
|
||||
//ruby-specific header to provide mingw-friendly xaudio2 interfaces
|
||||
|
||||
//64-bit GCC fix
|
||||
#define GUID_EXT EXTERN_C
|
||||
|
@ -34,20 +28,6 @@ DECLARE_INTERFACE(IXAudio2Voice);
|
|||
#define XAUDIO2_DEBUG_ENGINE 0x0001
|
||||
#define XAUDIO2_VOICE_NOSRC 0x0004
|
||||
|
||||
typedef struct
|
||||
{
|
||||
WAVEFORMATEX Format;
|
||||
union
|
||||
{
|
||||
WORD wValidBitsPerSample;
|
||||
WORD wSamplesPerBlock;
|
||||
WORD wReserved;
|
||||
} Samples;
|
||||
DWORD dwChannelMask;
|
||||
GUID SubFormat;
|
||||
} WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE, *LPPWAVEFORMATEXTENSIBLE;
|
||||
typedef const WAVEFORMATEXTENSIBLE* LPCWAVEFORMATEXTENSIBLE;
|
||||
|
||||
typedef enum XAUDIO2_DEVICE_ROLE
|
||||
{
|
||||
NotDefaultDevice = 0x0,
|
||||
|
@ -337,4 +317,3 @@ __inline HRESULT XAudio2Create(IXAudio2** ppXAudio2, UINT32 Flags X2DEFAULT(0),
|
|||
}
|
||||
return hr;
|
||||
}
|
||||
#endif
|
||||
|
|
|
@ -3,10 +3,10 @@
|
|||
/* ruby
|
||||
* author: byuu
|
||||
* license: ISC
|
||||
* version: 0.14 (2015-11-19)
|
||||
* version: 0.15 (2016-04-18)
|
||||
*
|
||||
* ruby is a cross-platform hardware abstraction layer
|
||||
* it provides a common interface to video, audio and input devices
|
||||
* ruby is a cross-platform hardware abstraction layer.
|
||||
* it provides a common interface to video, audio and input devices.
|
||||
*/
|
||||
|
||||
#include <nall/nall.hpp>
|
||||
|
@ -62,7 +62,7 @@ struct Audio {
|
|||
virtual auto get(const nall::string& name) -> nall::any { return false; }
|
||||
virtual auto set(const nall::string& name, const nall::any& value) -> bool { return false; }
|
||||
|
||||
virtual auto sample(uint16_t left, uint16_t right) -> void {}
|
||||
virtual auto sample(int16_t left, int16_t right) -> void {}
|
||||
virtual auto clear() -> void {}
|
||||
|
||||
virtual auto init() -> bool { return true; }
|
||||
|
|
Loading…
Reference in New Issue