mirror of
https://github.com/wheremyfoodat/Panda3DS.git
synced 2025-04-06 06:05:40 +12:00
Merge pull request #649 from wheremyfoodat/volume
Audio: Add more settings
This commit is contained in:
commit
547d47d9dc
16 changed files with 119 additions and 48 deletions
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@ -18,7 +18,8 @@ namespace Audio::AAC {
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public:
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// Decode function. Takes in a reference to the AAC response & request, and a callback for paddr -> pointer conversions
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void decode(AAC::Message& response, const AAC::Message& request, PaddrCallback paddrCallback);
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// We also allow for optionally muting the AAC output (setting all of it to 0) instead of properly decoding it, for debug/research purposes
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void decode(AAC::Message& response, const AAC::Message& request, PaddrCallback paddrCallback, bool enableAudio = true);
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~Decoder();
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};
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} // namespace Audio::AAC
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@ -14,6 +14,7 @@
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// The DSP core must have access to the DSP service to be able to trigger interrupts properly
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class DSPService;
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class Memory;
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struct EmulatorConfig;
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namespace Audio {
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// There are 160 stereo samples in 1 audio frame, so 320 samples total
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@ -31,6 +32,7 @@ namespace Audio {
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Memory& mem;
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Scheduler& scheduler;
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DSPService& dspService;
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EmulatorConfig& settings;
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Samples sampleBuffer;
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bool audioEnabled = false;
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@ -39,7 +41,8 @@ namespace Audio {
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public:
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enum class Type { Null, Teakra, HLE };
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DSPCore(Memory& mem, Scheduler& scheduler, DSPService& dspService) : mem(mem), scheduler(scheduler), dspService(dspService) {}
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DSPCore(Memory& mem, Scheduler& scheduler, DSPService& dspService, EmulatorConfig& settings)
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: mem(mem), scheduler(scheduler), dspService(dspService), settings(settings) {}
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virtual ~DSPCore() {}
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virtual void reset() = 0;
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@ -62,5 +65,5 @@ namespace Audio {
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virtual void setAudioEnabled(bool enable) { audioEnabled = enable; }
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};
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std::unique_ptr<DSPCore> makeDSPCore(DSPCore::Type type, Memory& mem, Scheduler& scheduler, DSPService& dspService);
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std::unique_ptr<DSPCore> makeDSPCore(EmulatorConfig& config, Memory& mem, Scheduler& scheduler, DSPService& dspService);
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} // namespace Audio
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@ -206,7 +206,7 @@ namespace Audio {
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SampleBuffer decodeADPCM(const u8* data, usize sampleCount, Source& source);
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public:
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HLE_DSP(Memory& mem, Scheduler& scheduler, DSPService& dspService);
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HLE_DSP(Memory& mem, Scheduler& scheduler, DSPService& dspService, EmulatorConfig& config);
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~HLE_DSP() override {}
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void reset() override;
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@ -3,6 +3,7 @@
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#include <string>
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#include <vector>
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#include "config.hpp"
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#include "helpers.hpp"
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#include "miniaudio.h"
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#include "ring_buffer.hpp"
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@ -12,12 +13,13 @@ class MiniAudioDevice {
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static constexpr ma_uint32 sampleRate = 32768; // 3DS sample rate
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static constexpr ma_uint32 channelCount = 2; // Audio output is stereo
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ma_device device;
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ma_context context;
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ma_device_config deviceConfig;
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ma_device device;
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ma_resampler resampler;
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Samples* samples = nullptr;
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const AudioDeviceConfig& audioSettings;
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bool initialized = false;
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bool running = false;
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@ -26,7 +28,8 @@ class MiniAudioDevice {
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std::vector<std::string> audioDevices;
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public:
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MiniAudioDevice();
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MiniAudioDevice(const AudioDeviceConfig& audioSettings);
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// If safe is on, we create a null audio device
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void init(Samples& samples, bool safe = false);
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void close();
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@ -23,7 +23,7 @@ namespace Audio {
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bool loaded = false; // Have we loaded a component?
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public:
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NullDSP(Memory& mem, Scheduler& scheduler, DSPService& dspService) : DSPCore(mem, scheduler, dspService) {}
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NullDSP(Memory& mem, Scheduler& scheduler, DSPService& dspService, EmulatorConfig& config) : DSPCore(mem, scheduler, dspService, config) {}
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~NullDSP() override {}
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void reset() override;
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@ -77,7 +77,7 @@ namespace Audio {
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}
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public:
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TeakraDSP(Memory& mem, Scheduler& scheduler, DSPService& dspService);
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TeakraDSP(Memory& mem, Scheduler& scheduler, DSPService& dspService, EmulatorConfig& config);
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~TeakraDSP() override {}
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void reset() override;
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@ -4,6 +4,19 @@
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#include "audio/dsp_core.hpp"
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#include "renderer.hpp"
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struct AudioDeviceConfig {
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float volumeRaw = 1.0f;
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bool muteAudio = false;
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float getVolume() const {
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if (muteAudio) {
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return 0.0f;
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}
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return volumeRaw;
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}
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};
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// Remember to initialize every field here to its default value otherwise bad things will happen
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struct EmulatorConfig {
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// Only enable the shader JIT by default on platforms where it's completely tested
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@ -41,6 +54,7 @@ struct EmulatorConfig {
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bool audioEnabled = false;
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bool vsyncEnabled = true;
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bool aacEnabled = true; // Enable AAC audio?
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bool enableRenderdoc = false;
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bool printAppVersion = true;
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@ -70,6 +84,7 @@ struct EmulatorConfig {
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};
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WindowSettings windowSettings;
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AudioDeviceConfig audioDeviceConfig;
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EmulatorConfig(const std::filesystem::path& path);
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void load();
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@ -96,7 +96,14 @@ void EmulatorConfig::load() {
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auto dspCoreName = toml::find_or<std::string>(audio, "DSPEmulation", "HLE");
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dspType = Audio::DSPCore::typeFromString(dspCoreName);
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audioEnabled = toml::find_or<toml::boolean>(audio, "EnableAudio", false);
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aacEnabled = toml::find_or<toml::boolean>(audio, "EnableAACAudio", true);
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audioDeviceConfig.muteAudio = toml::find_or<toml::boolean>(audio, "MuteAudio", false);
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// Our volume ranges from 0.0 (muted) to 2.0 (boosted, using a logarithmic scale). 1.0 is the "default" volume, ie we don't adjust the PCM
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// samples at all.
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audioDeviceConfig.volumeRaw = float(std::clamp(toml::find_or<toml::floating>(audio, "AudioVolume", 1.0), 0.0, 2.0));
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}
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}
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@ -167,6 +174,9 @@ void EmulatorConfig::save() {
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data["Audio"]["DSPEmulation"] = std::string(Audio::DSPCore::typeToString(dspType));
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data["Audio"]["EnableAudio"] = audioEnabled;
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data["Audio"]["EnableAACAudio"] = aacEnabled;
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data["Audio"]["MuteAudio"] = audioDeviceConfig.muteAudio;
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data["Audio"]["AudioVolume"] = double(audioDeviceConfig.volumeRaw);
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data["Battery"]["ChargerPlugged"] = chargerPlugged;
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data["Battery"]["BatteryPercentage"] = batteryPercentage;
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@ -370,12 +370,11 @@ void ShaderEmitter::storeRegister(Xmm source, const PICAShader& shader, u32 dest
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} else if (haveSSE4_1) {
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// Bit reverse the write mask because that is what blendps expects
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u32 adjustedMask = ((writeMask >> 3) & 0b1) | ((writeMask >> 1) & 0b10) | ((writeMask << 1) & 0b100) | ((writeMask << 3) & 0b1000);
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// Don't accidentally overwrite scratch1 if that is what we're writing derp
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Xmm temp = (source == scratch1) ? scratch2 : scratch1;
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movaps(temp, xword[statePointer + offset]); // Read current value of dest
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blendps(temp, source, adjustedMask); // Blend with source
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movaps(xword[statePointer + offset], temp); // Write back
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// Blend current value of dest with source. We have to invert the bits of the mask, as we do blendps source, dest instead of dest, source
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// Note: This destroys source
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blendps(source, xword[statePointer + offset], adjustedMask ^ 0xF);
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movaps(xword[statePointer + offset], source); // Write back
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} else {
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// Blend algo referenced from Citra
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const u8 selector = (((writeMask & 0b1000) ? 1 : 0) << 0) |
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@ -5,7 +5,7 @@
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#include <vector>
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using namespace Audio;
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void AAC::Decoder::decode(AAC::Message& response, const AAC::Message& request, AAC::Decoder::PaddrCallback paddrCallback) {
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void AAC::Decoder::decode(AAC::Message& response, const AAC::Message& request, AAC::Decoder::PaddrCallback paddrCallback, bool enableAudio) {
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// Copy the command and mode fields of the request to the response
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response.command = request.command;
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response.mode = request.mode;
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@ -95,9 +95,16 @@ void AAC::Decoder::decode(AAC::Message& response, const AAC::Message& request, A
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}
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}
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for (int sample = 0; sample < info->frameSize; sample++) {
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if (enableAudio) {
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for (int sample = 0; sample < info->frameSize; sample++) {
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for (int stream = 0; stream < channels; stream++) {
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audioStreams[stream].push_back(frame[(sample * channels) + stream]);
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}
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}
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} else {
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// If audio is not enabled, push 0s
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for (int stream = 0; stream < channels; stream++) {
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audioStreams[stream].push_back(frame[(sample * channels) + stream]);
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audioStreams[stream].resize(audioStreams[stream].size() + info->frameSize, 0);
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}
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}
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} else {
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@ -8,17 +8,17 @@
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#include "audio/null_core.hpp"
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#include "audio/teakra_core.hpp"
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std::unique_ptr<Audio::DSPCore> Audio::makeDSPCore(DSPCore::Type type, Memory& mem, Scheduler& scheduler, DSPService& dspService) {
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std::unique_ptr<Audio::DSPCore> Audio::makeDSPCore(EmulatorConfig& config, Memory& mem, Scheduler& scheduler, DSPService& dspService) {
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std::unique_ptr<DSPCore> core;
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switch (type) {
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case DSPCore::Type::Null: core = std::make_unique<NullDSP>(mem, scheduler, dspService); break;
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case DSPCore::Type::Teakra: core = std::make_unique<TeakraDSP>(mem, scheduler, dspService); break;
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case DSPCore::Type::HLE: core = std::make_unique<HLE_DSP>(mem, scheduler, dspService); break;
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switch (config.dspType) {
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case DSPCore::Type::Null: core = std::make_unique<NullDSP>(mem, scheduler, dspService, config); break;
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case DSPCore::Type::Teakra: core = std::make_unique<TeakraDSP>(mem, scheduler, dspService, config); break;
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case DSPCore::Type::HLE: core = std::make_unique<HLE_DSP>(mem, scheduler, dspService, config); break;
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default:
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Helpers::warn("Invalid DSP core selected!");
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core = std::make_unique<NullDSP>(mem, scheduler, dspService);
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core = std::make_unique<NullDSP>(mem, scheduler, dspService, config);
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break;
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}
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@ -8,6 +8,7 @@
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#include "audio/aac_decoder.hpp"
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#include "audio/dsp_simd.hpp"
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#include "config.hpp"
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#include "services/dsp.hpp"
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namespace Audio {
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@ -20,7 +21,8 @@ namespace Audio {
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};
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}
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HLE_DSP::HLE_DSP(Memory& mem, Scheduler& scheduler, DSPService& dspService) : DSPCore(mem, scheduler, dspService) {
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HLE_DSP::HLE_DSP(Memory& mem, Scheduler& scheduler, DSPService& dspService, EmulatorConfig& config)
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: DSPCore(mem, scheduler, dspService, config) {
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// Set up source indices
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for (int i = 0; i < sources.size(); i++) {
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sources[i].index = i;
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@ -702,24 +704,9 @@ namespace Audio {
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AAC::Message response;
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switch (request.command) {
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case AAC::Command::EncodeDecode: {
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// Dummy response to stop games from hanging
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response.resultCode = AAC::ResultCode::Success;
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response.decodeResponse.channelCount = 2;
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response.decodeResponse.sampleCount = 1024;
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response.decodeResponse.size = 0;
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response.decodeResponse.sampleRate = AAC::SampleRate::Rate48000;
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response.command = request.command;
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response.mode = request.mode;
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// TODO: Make this a toggle in config.toml. Currently we have it on by default.
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constexpr bool enableAAC = true;
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if (enableAAC) {
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aacDecoder->decode(response, request, [this](u32 paddr) { return getPointerPhys<u8>(paddr); });
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}
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case AAC::Command::EncodeDecode:
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aacDecoder->decode(response, request, [this](u32 paddr) { return getPointerPhys<u8>(paddr); }, settings.aacEnabled);
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break;
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}
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case AAC::Command::Init:
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case AAC::Command::Shutdown:
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@ -1,10 +1,14 @@
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#include "audio/miniaudio_device.hpp"
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#include <algorithm>
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#include <cmath>
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#include <cstring>
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#include <limits>
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#include "helpers.hpp"
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MiniAudioDevice::MiniAudioDevice() : initialized(false), running(false), samples(nullptr) {}
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MiniAudioDevice::MiniAudioDevice(const AudioDeviceConfig& audioSettings)
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: initialized(false), running(false), samples(nullptr), audioSettings(audioSettings) {}
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void MiniAudioDevice::init(Samples& samples, bool safe) {
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this->samples = &samples;
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@ -106,6 +110,40 @@ void MiniAudioDevice::init(Samples& samples, bool safe) {
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std::memcpy(&self->lastStereoSample[0], &output[(samplesWritten - 1) * 2], sizeof(lastStereoSample));
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}
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// Adjust the volume of our samples based on the emulator's volume slider
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float audioVolume = self->audioSettings.getVolume();
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// If volume is 1.0 we don't need to do anything
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if (audioVolume != 1.0f) {
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s16* sample = output;
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// If our volume is > 1.0 then we boost samples using a logarithmic scale,
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// In this case we also have to clamp samples to make sure they don't wrap around
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if (audioVolume > 1.0f) {
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audioVolume = 0.6 + 20 * std::log10(audioVolume);
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constexpr s32 min = s32(std::numeric_limits<s16>::min());
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constexpr s32 max = s32(std::numeric_limits<s16>::max());
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for (usize i = 0; i < samplesWritten; i += 2) {
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s16 l = s16(std::clamp<s32>(s32(float(sample[0]) * audioVolume), min, max));
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s16 r = s16(std::clamp<s32>(s32(float(sample[1]) * audioVolume), min, max));
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*sample++ = l;
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*sample++ = r;
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}
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} else {
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// If our volume is in [0.0, 1.0) then just multiply by the volume. No need to clamp, since there is no danger of our samples wrapping
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// around due to overflow
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for (usize i = 0; i < samplesWritten; i += 2) {
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s16 l = s16(float(sample[0]) * audioVolume);
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s16 r = s16(float(sample[1]) * audioVolume);
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*sample++ = l;
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*sample++ = r;
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}
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}
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}
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// If underruning, copy the last output sample
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{
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s16* pointer = &output[samplesWritten * 2];
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@ -36,8 +36,8 @@ struct Dsp1 {
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Segment segments[10];
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};
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TeakraDSP::TeakraDSP(Memory& mem, Scheduler& scheduler, DSPService& dspService)
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: DSPCore(mem, scheduler, dspService), pipeBaseAddr(0), running(false) {
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TeakraDSP::TeakraDSP(Memory& mem, Scheduler& scheduler, DSPService& dspService, EmulatorConfig& config)
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: DSPCore(mem, scheduler, dspService, config), pipeBaseAddr(0), running(false) {
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// Set up callbacks for Teakra
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Teakra::AHBMCallback ahbm;
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|
|
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@ -20,7 +20,7 @@ __declspec(dllexport) DWORD AmdPowerXpressRequestHighPerformance = 1;
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Emulator::Emulator()
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: config(getConfigPath()), kernel(cpu, memory, gpu, config), cpu(memory, kernel, *this), gpu(memory, config), memory(cpu.getTicksRef(), config),
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cheats(memory, kernel.getServiceManager().getHID()), lua(*this), running(false)
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cheats(memory, kernel.getServiceManager().getHID()), audioDevice(config.audioDeviceConfig), lua(*this), running(false)
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#ifdef PANDA3DS_ENABLE_HTTP_SERVER
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,
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httpServer(this)
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@ -28,7 +28,7 @@ Emulator::Emulator()
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{
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DSPService& dspService = kernel.getServiceManager().getDSP();
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dsp = Audio::makeDSPCore(config.dspType, memory, scheduler, dspService);
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dsp = Audio::makeDSPCore(config, memory, scheduler, dspService);
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dspService.setDSPCore(dsp.get());
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audioDevice.init(dsp->getSamples());
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|
|
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@ -172,12 +172,16 @@ static void configInit() {
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{"panda3ds_use_vsync", "Enable VSync; enabled|disabled"},
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{"panda3ds_dsp_emulation", "DSP emulation; HLE|LLE|Null"},
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{"panda3ds_use_audio", "Enable audio; disabled|enabled"},
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{"panda3ds_audio_volume", "Audio volume; 100|0|10|20|40|60|80|90|100|120|140|150|180|200"},
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{"panda3ds_mute_audio", "Mute audio; disabled|enabled"},
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{"panda3ds_enable_aac", "Enable AAC audio; enabled|disabled"},
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{"panda3ds_ubershader_lighting_override", "Force shadergen when rendering lights; enabled|disabled"},
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{"panda3ds_ubershader_lighting_override_threshold", "Light threshold for forcing shadergen; 1|2|3|4|5|6|7|8"},
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{"panda3ds_use_virtual_sd", "Enable virtual SD card; enabled|disabled"},
|
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{"panda3ds_write_protect_virtual_sd", "Write protect virtual SD card; disabled|enabled"},
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{"panda3ds_battery_level", "Battery percentage; 5|10|20|30|50|70|90|100"},
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{"panda3ds_use_charger", "Charger plugged; enabled|disabled"},
|
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{"panda3ds_ubershader_lighting_override", "Force shadergen when rendering lights; enabled|disabled"},
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{"panda3ds_ubershader_lighting_override_threshold", "Light threshold for forcing shadergen; 1|2|3|4|5|6|7|8"},
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{nullptr, nullptr},
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};
|
||||
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||||
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@ -194,6 +198,10 @@ static void configUpdate() {
|
|||
config.batteryPercentage = fetchVariableRange("panda3ds_battery_level", 5, 100);
|
||||
config.dspType = Audio::DSPCore::typeFromString(fetchVariable("panda3ds_dsp_emulation", "null"));
|
||||
config.audioEnabled = fetchVariableBool("panda3ds_use_audio", false);
|
||||
config.aacEnabled = fetchVariableBool("panda3ds_enable_aac", true);
|
||||
config.audioDeviceConfig.muteAudio = fetchVariableBool("panda3ds_mute_audio", false);
|
||||
config.audioDeviceConfig.volumeRaw = float(fetchVariableRange("panda3ds_audio_volume", 0, 200)) / 100.0f;
|
||||
|
||||
config.sdCardInserted = fetchVariableBool("panda3ds_use_virtual_sd", true);
|
||||
config.sdWriteProtected = fetchVariableBool("panda3ds_write_protect_virtual_sd", false);
|
||||
config.accurateShaderMul = fetchVariableBool("panda3ds_accurate_shader_mul", false);
|
||||
|
|
Loading…
Add table
Reference in a new issue