more voice work!!!
This commit is contained in:
Kae
parent
212de6b876
commit
f14f77724d
@ -248,9 +248,15 @@ public:
|
||||
if (SDL_InitSubSystem(SDL_INIT_GAMECONTROLLER))
|
||||
throw ApplicationException(strf("Couldn't initialize SDL Controller: {}", SDL_GetError()));
|
||||
|
||||
Logger::info("Application: Initializing SDL Sound");
|
||||
#ifdef STAR_SYSTEM_WINDOWS // Newer SDL is defaulting to xaudio2, which does not support audio capture
|
||||
SDL_setenv("SDL_AUDIODRIVER", "directsound", 1);
|
||||
#endif
|
||||
|
||||
Logger::info("Application: Initializing SDL Audio");
|
||||
if (SDL_InitSubSystem(SDL_INIT_AUDIO))
|
||||
throw ApplicationException(strf("Couldn't initialize SDL Sound: {}", SDL_GetError()));
|
||||
throw ApplicationException(strf("Couldn't initialize SDL Audio: {}", SDL_GetError()));
|
||||
|
||||
Logger::info("Application: using Audio Driver '{}'", SDL_GetCurrentAudioDriver());
|
||||
|
||||
SDL_JoystickEventState(SDL_ENABLE);
|
||||
|
||||
@ -336,17 +342,23 @@ public:
|
||||
closeAudioInputDevice();
|
||||
|
||||
SDL_AudioSpec obtained = {};
|
||||
return (m_sdlAudioInputDevice = SDL_OpenAudioDevice(name, 1, &desired, &obtained, 0)) != 0;
|
||||
m_sdlAudioInputDevice = SDL_OpenAudioDevice(name, 1, &desired, &obtained, 0);
|
||||
|
||||
if (m_sdlAudioInputDevice)
|
||||
Logger::info("Opened audio input device '{}'", SDL_GetAudioDeviceName(m_sdlAudioInputDevice, 1));
|
||||
else
|
||||
Logger::info("Failed to open audio input device: {}", SDL_GetError());
|
||||
|
||||
return m_sdlAudioInputDevice != 0;
|
||||
}
|
||||
|
||||
bool closeAudioInputDevice() {
|
||||
if (m_sdlAudioInputDevice) {
|
||||
Logger::info("Closing audio input device '{}'", SDL_GetAudioDeviceName(m_sdlAudioInputDevice, 1));
|
||||
SDL_CloseAudioDevice(m_sdlAudioInputDevice);
|
||||
m_sdlAudioInputDevice = 0;
|
||||
|
||||
return true;
|
||||
}
|
||||
|
||||
return false;
|
||||
}
|
||||
|
||||
|
||||
@ -208,6 +208,11 @@ void ClientApplication::applicationInit(ApplicationControllerPtr appController)
|
||||
|
||||
appController->setMaxFrameSkip(assets->json("/client.config:maxFrameSkip").toUInt());
|
||||
appController->setUpdateTrackWindow(assets->json("/client.config:updateTrackWindow").toFloat());
|
||||
|
||||
if (auto jVoice = configuration->get("voice"))
|
||||
m_voice->loadJson(jVoice.toObject());
|
||||
|
||||
m_voice->init();
|
||||
}
|
||||
|
||||
void ClientApplication::renderInit(RendererPtr renderer) {
|
||||
|
||||
@ -20,8 +20,6 @@ struct KeySet {
|
||||
secret[31] |= 64;
|
||||
|
||||
ed25519_CreateKeyPair(publicKey.data(), privateKey.data(), nullptr, secret.data());
|
||||
|
||||
Logger::info("Generated Curve25519 key-pair");
|
||||
}
|
||||
};
|
||||
|
||||
|
||||
@ -1,8 +1,11 @@
|
||||
#include "StarVoice.hpp"
|
||||
#include "StarFormat.hpp"
|
||||
#include "StarApplicationController.hpp"
|
||||
#include "StarTime.hpp"
|
||||
#include "StarRoot.hpp"
|
||||
#include "opus/include/opus.h"
|
||||
|
||||
#include <queue>
|
||||
#include "SDL.h"
|
||||
|
||||
constexpr int VOICE_SAMPLE_RATE = 48000;
|
||||
@ -25,10 +28,129 @@ EnumMap<VoiceChannelMode> const VoiceChannelModeNames{
|
||||
{VoiceChannelMode::Stereo, "Stereo"}
|
||||
};
|
||||
|
||||
float getAudioChunkLoudness(int16_t* data, size_t samples) {
|
||||
if (!samples)
|
||||
return 0.f;
|
||||
|
||||
double rms = 0.;
|
||||
for (size_t i = 0; i != samples; ++i) {
|
||||
float sample = (float)data[i] / 32767.f;
|
||||
rms += (double)(sample * sample);
|
||||
}
|
||||
|
||||
float fRms = sqrtf((float)(rms / samples));
|
||||
|
||||
if (fRms > 0)
|
||||
return std::clamp<float>(20.f * log10f(fRms), -127.f, 0.f);
|
||||
else
|
||||
return -127.f;
|
||||
}
|
||||
|
||||
float getAudioLoudness(int16_t* data, size_t samples) {
|
||||
constexpr size_t CHUNK_SIZE = 50;
|
||||
|
||||
float highest = -127.f;
|
||||
for (size_t i = 0; i < samples; i += CHUNK_SIZE) {
|
||||
float level = getAudioChunkLoudness(data + i, std::min<size_t>(i + CHUNK_SIZE, samples) - i);
|
||||
if (level > highest)
|
||||
highest = level;
|
||||
}
|
||||
|
||||
return highest;
|
||||
}
|
||||
|
||||
struct VoiceAudioChunk {
|
||||
std::unique_ptr<int16_t[]> data;
|
||||
size_t remaining;
|
||||
size_t offset = 0;
|
||||
|
||||
VoiceAudioChunk(int16_t* ptr, size_t size) {
|
||||
data.reset(ptr);
|
||||
remaining = size;
|
||||
offset = 0;
|
||||
}
|
||||
|
||||
inline size_t takeSamples(std::vector<int16_t>& out, size_t count) {
|
||||
size_t toRead = std::min<size_t>(count, remaining);
|
||||
int16_t* start = data.get() + offset;
|
||||
out.insert(out.end(), start, start + toRead);
|
||||
offset += toRead;
|
||||
remaining -= toRead;
|
||||
return toRead;
|
||||
}
|
||||
|
||||
//this one's unsafe
|
||||
inline int16_t takeSample() {
|
||||
--remaining;
|
||||
return *(data.get() + offset++);
|
||||
}
|
||||
|
||||
inline bool exhausted() {
|
||||
return remaining == 0;
|
||||
}
|
||||
};
|
||||
|
||||
struct VoiceAudioStream {
|
||||
// TODO: This should really be a ring buffer instead.
|
||||
std::queue<VoiceAudioChunk> chunks{};
|
||||
size_t samples = 0;
|
||||
atomic<bool> muted = false;
|
||||
atomic<bool> playing = false;
|
||||
atomic<float> decibelLevel = 0.0f;
|
||||
atomic<Array<float, 2>> channelVolumes = Array<float, 2>::filled(1.0f);
|
||||
|
||||
Mutex mutex;
|
||||
|
||||
inline int16_t getSample() {
|
||||
int16_t sample = 0;
|
||||
while (!chunks.empty()) {
|
||||
auto& front = chunks.front();
|
||||
if (front.exhausted()) {
|
||||
chunks.pop();
|
||||
continue;
|
||||
}
|
||||
--samples;
|
||||
return front.takeSample();
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
void nukeSamples(size_t count) {
|
||||
while (!chunks.empty() && count > 0) {
|
||||
auto& front = chunks.front();
|
||||
if (count >= front.remaining) {
|
||||
count -= front.remaining;
|
||||
samples -= front.remaining;
|
||||
chunks.pop();
|
||||
}
|
||||
else {
|
||||
for (size_t i = 0; i != count; ++i) {
|
||||
--samples;
|
||||
front.takeSample();
|
||||
}
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
inline bool empty() { return chunks.empty(); }
|
||||
|
||||
void take(int16_t* ptr, size_t size) {
|
||||
MutexLocker lock(mutex);
|
||||
while (samples > 22050 && !chunks.empty()) {
|
||||
samples -= chunks.front().remaining;
|
||||
chunks.pop();
|
||||
}
|
||||
chunks.emplace(ptr, size);
|
||||
samples += size;
|
||||
}
|
||||
};
|
||||
|
||||
Voice::Speaker::Speaker(SpeakerId id)
|
||||
: decoderMono (createDecoder(1), opus_decoder_destroy)
|
||||
, decoderStereo(createDecoder(2), opus_decoder_destroy) {
|
||||
speakerId = id;
|
||||
audioStream = make_shared<VoiceAudioStream>();
|
||||
}
|
||||
|
||||
Voice* Voice::s_singleton;
|
||||
@ -53,19 +175,58 @@ Voice::Voice(ApplicationControllerPtr appController) : m_encoder(nullptr, opus_e
|
||||
m_channelMode = VoiceChannelMode::Mono;
|
||||
m_applicationController = appController;
|
||||
|
||||
resetEncoder();
|
||||
s_singleton = this;
|
||||
}
|
||||
|
||||
Voice::~Voice() {
|
||||
save();
|
||||
|
||||
s_singleton = nullptr;
|
||||
}
|
||||
|
||||
void Voice::load(Json const& config) {
|
||||
// do stuff
|
||||
void Voice::init() {
|
||||
resetEncoder();
|
||||
if (m_inputEnabled)
|
||||
openDevice();
|
||||
}
|
||||
Json Voice::save() const {
|
||||
return JsonObject{};
|
||||
|
||||
void Voice::loadJson(Json const& config) {
|
||||
m_enabled = config.getBool("enabled", m_enabled);
|
||||
m_inputEnabled = config.getBool("inputEnabled", m_inputEnabled);
|
||||
m_deviceName = config.optQueryString("inputDevice");
|
||||
m_threshold = config.getFloat("threshold", m_threshold);
|
||||
m_inputVolume = config.getFloat("inputVolume", m_inputVolume);
|
||||
m_outputVolume = config.getFloat("outputVolume", m_outputVolume);
|
||||
m_inputMode = VoiceInputModeNames.getLeft(config.getString("inputMode", "pushToTalk"));
|
||||
m_channelMode = VoiceChannelModeNames.getLeft(config.getString("channelMode", "mono"));
|
||||
}
|
||||
|
||||
|
||||
|
||||
Json Voice::saveJson() const {
|
||||
return JsonObject{
|
||||
{"enabled", m_enabled},
|
||||
{"inputEnabled", m_inputEnabled},
|
||||
{"inputDevice", m_deviceName ? *m_deviceName : Json()},
|
||||
{"threshold", m_threshold},
|
||||
{"inputVolume", m_inputVolume},
|
||||
{"outputVolume", m_outputVolume},
|
||||
{"inputMode", VoiceInputModeNames.getRight(m_inputMode)},
|
||||
{"channelMode", VoiceChannelModeNames.getRight(m_channelMode)},
|
||||
{"version", 1}
|
||||
};
|
||||
}
|
||||
|
||||
void Voice::save() const {
|
||||
if (Root* root = Root::singletonPtr()) {
|
||||
if (auto config = root->configuration())
|
||||
config->set("voice", saveJson());
|
||||
}
|
||||
}
|
||||
|
||||
void Voice::scheduleSave() {
|
||||
if (nextSaveTime == 0.0)
|
||||
nextSaveTime = Time::monotonicTime() + 2.0;
|
||||
}
|
||||
|
||||
Voice::SpeakerPtr Voice::setLocalSpeaker(SpeakerId speakerId) {
|
||||
@ -87,6 +248,10 @@ Voice::SpeakerPtr Voice::speaker(SpeakerId speakerId) {
|
||||
}
|
||||
}
|
||||
|
||||
void Voice::getAudioData(uint8_t* stream, int len) {
|
||||
|
||||
}
|
||||
|
||||
void Voice::mix(int16_t* buffer, size_t frames, unsigned channels) {
|
||||
|
||||
}
|
||||
@ -95,13 +260,29 @@ void Voice::update(PositionalAttenuationFunction positionalAttenuationFunction)
|
||||
if (positionalAttenuationFunction) {
|
||||
for (auto& entry : m_speakers) {
|
||||
if (SpeakerPtr& speaker = entry.second) {
|
||||
speaker->channelVolumes = {
|
||||
speaker->audioStream->channelVolumes = {
|
||||
positionalAttenuationFunction(0, speaker->position, 1.0f),
|
||||
positionalAttenuationFunction(1, speaker->position, 1.0f)
|
||||
};
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
auto now = Time::monotonicTime();
|
||||
if (now > nextSaveTime) {
|
||||
nextSaveTime = 0.0;
|
||||
save();
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
void Voice::setDeviceName(Maybe<String> deviceName) {
|
||||
if (m_deviceName == deviceName)
|
||||
return;
|
||||
|
||||
m_deviceName = deviceName;
|
||||
if (m_deviceOpen)
|
||||
openDevice();
|
||||
}
|
||||
|
||||
OpusDecoder* Voice::createDecoder(int channels) {
|
||||
@ -131,6 +312,10 @@ void Voice::resetEncoder() {
|
||||
void Voice::openDevice() {
|
||||
closeDevice();
|
||||
|
||||
m_applicationController->openAudioInputDevice(m_deviceName ? m_deviceName->utf8Ptr() : nullptr, VOICE_SAMPLE_RATE, encoderChannels(), this, [](void* userdata, uint8_t* stream, int len) {
|
||||
((Voice*)(userdata))->getAudioData(stream, len);
|
||||
});
|
||||
|
||||
m_deviceOpen = true;
|
||||
}
|
||||
|
||||
|
||||
@ -5,6 +5,7 @@
|
||||
#include "StarException.hpp"
|
||||
#include "StarGameTypes.hpp"
|
||||
#include "StarMaybe.hpp"
|
||||
#include "StarThread.hpp"
|
||||
#include "StarApplicationController.hpp"
|
||||
|
||||
struct OpusDecoder;
|
||||
@ -23,6 +24,7 @@ enum class VoiceChannelMode: uint8_t { Mono = 1, Stereo = 2 };
|
||||
extern EnumMap<VoiceChannelMode> const VoiceChannelModeNames;
|
||||
|
||||
STAR_CLASS(Voice);
|
||||
STAR_CLASS(VoiceAudioStream);
|
||||
STAR_CLASS(ApplicationController);
|
||||
|
||||
class Voice {
|
||||
@ -30,7 +32,8 @@ public:
|
||||
// Individual speakers are represented by their connection ID.
|
||||
typedef ConnectionId SpeakerId;
|
||||
|
||||
struct Speaker {
|
||||
class Speaker {
|
||||
public:
|
||||
SpeakerId speakerId = 0;
|
||||
EntityId entityId = 0;
|
||||
|
||||
@ -39,10 +42,8 @@ public:
|
||||
|
||||
OpusDecoderPtr decoderMono;
|
||||
OpusDecoderPtr decoderStereo;
|
||||
|
||||
atomic<bool> active = false;
|
||||
atomic<float> currentLoudness = 0.0f;
|
||||
atomic<Array<float, 2>> channelVolumes = Array<float, 2>::filled(1.0f);
|
||||
VoiceAudioStreamPtr audioStream;
|
||||
Mutex mutex;
|
||||
|
||||
Speaker(SpeakerId speakerId);
|
||||
};
|
||||
@ -63,19 +64,29 @@ public:
|
||||
Voice(Voice const&) = delete;
|
||||
Voice& operator=(Voice const&) = delete;
|
||||
|
||||
void load(Json const& config);
|
||||
Json save() const;
|
||||
void init();
|
||||
|
||||
void loadJson(Json const& config);
|
||||
Json saveJson() const;
|
||||
|
||||
void save() const;
|
||||
void scheduleSave();
|
||||
|
||||
// Sets the local speaker ID and returns the local speaker. Must be called upon loading into a world.
|
||||
SpeakerPtr setLocalSpeaker(SpeakerId speakerId);
|
||||
SpeakerPtr speaker(SpeakerId speakerId);
|
||||
|
||||
// Called when receiving input audio data from SDL, on its own thread.
|
||||
void getAudioData(uint8_t* stream, int len);
|
||||
|
||||
// Called to mix voice audio with the game.
|
||||
void mix(int16_t* buffer, size_t frames, unsigned channels);
|
||||
|
||||
typedef function<float(unsigned, Vec2F, float)> PositionalAttenuationFunction;
|
||||
void update(PositionalAttenuationFunction positionalAttenuationFunction = {});
|
||||
|
||||
void setDeviceName(Maybe<String> device);
|
||||
|
||||
inline int encoderChannels() const {
|
||||
return m_channelMode == VoiceChannelMode::Mono ? 1 : 2;
|
||||
}
|
||||
@ -96,12 +107,21 @@ private:
|
||||
|
||||
OpusEncoderPtr m_encoder;
|
||||
|
||||
float m_outputVolume = 1.0f;
|
||||
float m_inputVolume = 1.0f;
|
||||
float m_threshold = -50.0f;
|
||||
|
||||
bool m_enabled = true;
|
||||
bool m_inputEnabled = true;
|
||||
|
||||
bool m_deviceOpen = false;
|
||||
Maybe<String> m_deviceName;
|
||||
VoiceInputMode m_inputMode;
|
||||
VoiceChannelMode m_channelMode;
|
||||
|
||||
ApplicationControllerPtr m_applicationController;
|
||||
|
||||
double nextSaveTime = 0.0f;
|
||||
};
|
||||
|
||||
}
|
||||
|
||||
Loading…
Reference in New Issue
Block a user