518 lines
14 KiB
C++
518 lines
14 KiB
C++
#include "StarVoice.hpp"
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#include "StarFormat.hpp"
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#include "StarApplicationController.hpp"
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#include "StarTime.hpp"
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#include "StarRoot.hpp"
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#include "StarLogging.hpp"
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#include "opus/include/opus.h"
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#include "SDL.h"
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constexpr int VOICE_SAMPLE_RATE = 48000;
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constexpr int VOICE_FRAME_SIZE = 960;
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constexpr int VOICE_MAX_FRAME_SIZE = 6 * VOICE_FRAME_SIZE;
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constexpr int VOICE_MAX_PACKET_SIZE = 3 * 1276;
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constexpr uint16_t VOICE_VERSION = 1;
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namespace Star {
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EnumMap<VoiceInputMode> const VoiceInputModeNames{
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{VoiceInputMode::VoiceActivity, "VoiceActivity"},
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{VoiceInputMode::PushToTalk, "PushToTalk"}
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};
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EnumMap<VoiceChannelMode> const VoiceChannelModeNames{
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{VoiceChannelMode::Mono, "Mono"},
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{VoiceChannelMode::Stereo, "Stereo"}
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};
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float getAudioChunkLoudness(int16_t* data, size_t samples) {
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if (!samples)
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return 0.f;
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double rms = 0.;
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for (size_t i = 0; i != samples; ++i) {
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float sample = (float)data[i] / 32767.f;
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rms += (double)(sample * sample);
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}
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float fRms = sqrtf((float)(rms / samples));
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if (fRms > 0)
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return std::clamp<float>(20.f * log10f(fRms), -127.f, 0.f);
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else
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return -127.f;
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}
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float getAudioLoudness(int16_t* data, size_t samples) {
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constexpr size_t CHUNK_SIZE = 50;
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float highest = -127.f;
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for (size_t i = 0; i < samples; i += CHUNK_SIZE) {
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float level = getAudioChunkLoudness(data + i, std::min<size_t>(i + CHUNK_SIZE, samples) - i);
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if (level > highest)
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highest = level;
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}
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return highest;
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}
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struct VoiceAudioStream {
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// TODO: This should really be a ring buffer instead.
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std::queue<VoiceAudioChunk> chunks{};
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size_t samples = 0;
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Mutex mutex;
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inline int16_t getSample() {
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int16_t sample = 0;
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while (!chunks.empty()) {
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auto& front = chunks.front();
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if (front.exhausted()) {
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chunks.pop();
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continue;
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}
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--samples;
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return front.takeSample();
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}
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return 0;
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}
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void nukeSamples(size_t count) {
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while (!chunks.empty() && count > 0) {
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auto& front = chunks.front();
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if (count >= front.remaining) {
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count -= front.remaining;
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samples -= front.remaining;
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chunks.pop();
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}
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else {
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for (size_t i = 0; i != count; ++i) {
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--samples;
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front.takeSample();
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}
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break;
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}
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}
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}
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inline bool empty() { return chunks.empty(); }
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void take(int16_t* ptr, size_t size) {
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MutexLocker lock(mutex);
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while (samples > 22050 && !chunks.empty()) {
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samples -= chunks.front().remaining;
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chunks.pop();
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}
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chunks.emplace(ptr, size);
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samples += size;
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}
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};
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Voice::Speaker::Speaker(SpeakerId id)
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: decoderMono (createDecoder(1), opus_decoder_destroy)
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, decoderStereo(createDecoder(2), opus_decoder_destroy) {
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speakerId = id;
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audioStream = make_shared<VoiceAudioStream>();
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}
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Voice* Voice::s_singleton;
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Voice* Voice::singletonPtr() {
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return s_singleton;
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}
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Voice& Voice::singleton() {
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if (!s_singleton)
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throw VoiceException("Voice::singleton() called with no Voice instance available");
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else
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return *s_singleton;
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}
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Voice::Voice(ApplicationControllerPtr appController) : m_encoder(nullptr, opus_encoder_destroy) {
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if (s_singleton)
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throw VoiceException("Singleton Voice has been constructed twice");
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m_clientSpeaker = make_shared<Speaker>(m_speakerId);
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m_inputMode = VoiceInputMode::PushToTalk;
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m_channelMode = VoiceChannelMode::Mono;
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m_applicationController = appController;
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s_singleton = this;
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}
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Voice::~Voice() {
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save();
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closeDevice();
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s_singleton = nullptr;
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}
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void Voice::init() {
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resetEncoder();
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if (m_inputEnabled)
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openDevice();
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}
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void Voice::loadJson(Json const& config) {
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m_enabled = config.getBool("enabled", m_enabled);
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m_inputEnabled = config.getBool("inputEnabled", m_inputEnabled);
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m_deviceName = config.optQueryString("inputDevice");
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m_threshold = config.getFloat("threshold", m_threshold);
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m_inputVolume = config.getFloat("inputVolume", m_inputVolume);
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m_outputVolume = config.getFloat("outputVolume", m_outputVolume);
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m_inputMode = VoiceInputModeNames.getLeft(config.getString("inputMode", "PushToTalk"));
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m_channelMode = VoiceChannelModeNames.getLeft(config.getString("channelMode", "Mono"));
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}
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Json Voice::saveJson() const {
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return JsonObject{
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{"enabled", m_enabled},
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{"inputEnabled", m_inputEnabled},
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{"inputDevice", m_deviceName ? *m_deviceName : Json()},
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{"threshold", m_threshold},
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{"inputVolume", m_inputVolume},
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{"outputVolume", m_outputVolume},
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{"inputMode", VoiceInputModeNames.getRight(m_inputMode)},
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{"channelMode", VoiceChannelModeNames.getRight(m_channelMode)},
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{"version", 1}
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};
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}
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void Voice::save() const {
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if (Root* root = Root::singletonPtr()) {
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if (auto config = root->configuration())
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config->set("voice", saveJson());
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}
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}
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void Voice::scheduleSave() {
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if (!m_nextSaveTime)
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m_nextSaveTime = Time::monotonicMilliseconds() + 2000;
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}
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Voice::SpeakerPtr Voice::setLocalSpeaker(SpeakerId speakerId) {
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if (m_speakers.contains(m_speakerId))
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m_speakers.remove(m_speakerId);
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m_clientSpeaker->speakerId = m_speakerId = speakerId;
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return m_speakers.insert(m_speakerId, m_clientSpeaker).first->second;
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}
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Voice::SpeakerPtr Voice::speaker(SpeakerId speakerId) {
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if (m_speakerId == speakerId)
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return m_clientSpeaker;
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else {
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if (SpeakerPtr const* ptr = m_speakers.ptr(speakerId))
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return *ptr;
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else
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return m_speakers.emplace(speakerId, make_shared<Speaker>(speakerId)).first->second;
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}
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}
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void Voice::readAudioData(uint8_t* stream, int len) {
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auto now = Time::monotonicMilliseconds();
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if (!m_encoder || m_inputMode == VoiceInputMode::PushToTalk && now > m_lastInputTime)
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return;
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// Stop encoding if 2048 bytes have been encoded and not taken by the game thread yet
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if (m_encodedChunksLength > 2048)
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return;
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size_t samples = len / 2;
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float decibels = getAudioLoudness((int16_t*)stream, samples);
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m_clientSpeaker->decibelLevel = decibels;
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bool active = true;
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if (m_inputMode == VoiceInputMode::VoiceActivity) {
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bool aboveThreshold = decibels > m_threshold;
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if (aboveThreshold)
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m_lastThresholdTime = now;
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active = now - m_lastThresholdTime < 50;
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}
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if (active) {
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m_capturedChunksFrames += samples / m_deviceChannels;
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auto data = (opus_int16*)malloc(len);
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memcpy(data, stream, len);
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m_capturedChunks.emplace(data, samples); // takes ownership
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}
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else { // Clear out any residual data so they don't manifest at the start of the next encode, whenever that is
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while (!m_capturedChunks.empty())
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m_capturedChunks.pop();
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m_capturedChunksFrames = 0;
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}
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while (m_capturedChunksFrames >= VOICE_FRAME_SIZE) {
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size_t samplesToTake = VOICE_FRAME_SIZE * (size_t)m_deviceChannels;
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std::vector<opus_int16> takenSamples;
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takenSamples.reserve(samplesToTake);
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while (!m_capturedChunks.empty()) {
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auto& front = m_capturedChunks.front();
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if (front.exhausted())
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m_capturedChunks.pop();
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else {
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samplesToTake -= front.takeSamples(takenSamples, samplesToTake);
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if (samplesToTake == 0)
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break;
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}
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}
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m_capturedChunksFrames -= VOICE_FRAME_SIZE;
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float vol = m_inputVolume;
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if (m_inputVolume != 1.0f) {
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for (size_t i = 0; i != takenSamples.size(); ++i)
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takenSamples[i] *= m_inputVolume;
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}
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ByteArray encodedData(VOICE_MAX_FRAME_SIZE, 0);
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opus_int32 encodedSize = opus_encode(m_encoder.get(), takenSamples.data(), VOICE_FRAME_SIZE, (unsigned char*)encodedData.ptr(), encodedData.size());
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if (encodedSize == 1)
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continue;
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else if (encodedSize < 0)
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Logger::error("Voice: Opus encode error {}", opus_strerror(encodedSize));
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else {
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encodedData.resize(encodedSize);
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MutexLocker lock(m_captureMutex);
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m_encodedChunks.emplace_back(move(encodedData)); // reset takes ownership of data buffer
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m_encodedChunksLength += encodedSize;
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Logger::info("Voice: encoded Opus chunk {} bytes big", encodedSize);
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}
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}
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}
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void Voice::mix(int16_t* buffer, size_t frameCount, unsigned channels) {
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size_t samples = frameCount * channels;
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static std::vector<int16_t> finalMixBuffer{};
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static std::vector<int32_t> voiceMixBuffer{};
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finalMixBuffer.resize(samples);
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voiceMixBuffer.resize(samples);
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int32_t* mixBuf = (int32_t*)memset(voiceMixBuffer.data(), 0, samples * sizeof(int32_t));
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//read into buffer now
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bool mix = false;
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{
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MutexLocker lock(m_activeSpeakersMutex);
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auto it = m_activeSpeakers.begin();
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while (it != m_activeSpeakers.end()) {
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SpeakerPtr const& speaker = *it;
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VoiceAudioStream* audio = speaker->audioStream.get();
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MutexLocker audioLock(audio->mutex);
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if (!audio->empty()) {
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if (!speaker->muted) {
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mix = true;
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auto channelVolumes = speaker->channelVolumes.load();
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for (size_t i = 0; i != samples; ++i)
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mixBuf[i] += (int32_t)(audio->getSample()) * channelVolumes[i % 2];
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}
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else {
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for (size_t i = 0; i != samples; ++i)
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audio->getSample();
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}
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++it;
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}
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else {
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speaker->playing = false;
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it = m_activeSpeakers.erase(it);
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}
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}
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}
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if (mix) {
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int16_t* finBuf = finalMixBuffer.data();
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float vol = m_outputVolume;
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for (size_t i = 0; i != samples; ++i)
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finBuf[i] = (int16_t)clamp<int>(mixBuf[i] * vol, INT16_MIN, INT16_MAX);
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SDL_MixAudioFormat((Uint8*)buffer, (Uint8*)finBuf, AUDIO_S16, samples * sizeof(int16_t), SDL_MIX_MAXVOLUME);
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}
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}
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void Voice::update(PositionalAttenuationFunction positionalAttenuationFunction) {
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if (positionalAttenuationFunction) {
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for (auto& entry : m_speakers) {
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if (SpeakerPtr& speaker = entry.second) {
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speaker->channelVolumes = {
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positionalAttenuationFunction(0, speaker->position, 1.0f),
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positionalAttenuationFunction(1, speaker->position, 1.0f)
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};
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}
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}
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}
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if (m_nextSaveTime && Time::monotonicMilliseconds() > m_nextSaveTime) {
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m_nextSaveTime = 0;
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save();
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}
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}
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void Voice::setDeviceName(Maybe<String> deviceName) {
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if (m_deviceName == deviceName)
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return;
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m_deviceName = deviceName;
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if (m_deviceOpen)
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openDevice();
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}
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int Voice::send(DataStreamBuffer& out, size_t budget) {
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out.setByteOrder(ByteOrder::LittleEndian);
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out.write<uint16_t>(VOICE_VERSION);
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MutexLocker captureLock(m_captureMutex);
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if (m_capturedChunks.empty())
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return 0;
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std::vector<ByteArray> encodedChunks = move(m_encodedChunks);
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size_t encodedChunksLength = m_encodedChunksLength;
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m_encodedChunksLength = 0;
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captureLock.unlock();
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for (auto& chunk : encodedChunks) {
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out.write<uint32_t>(chunk.size());
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out.writeBytes(chunk);
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if ((budget -= min<size_t>(budget, chunk.size())) == 0)
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break;
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}
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m_lastSentTime = Time::monotonicMilliseconds();
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return 1;
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}
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bool Voice::receive(SpeakerPtr speaker, std::string_view view) {
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if (!speaker || view.empty())
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return false;
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try {
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DataStreamExternalBuffer reader(view.data(), view.size());
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reader.setByteOrder(ByteOrder::LittleEndian);
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if (reader.read<uint16_t>() > VOICE_VERSION)
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return false;
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uint32_t opusLength = 0;
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while (!reader.atEnd()) {
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reader >> opusLength;
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auto opusData = (unsigned char*)reader.ptr() + reader.pos();
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reader.seek(opusLength, IOSeek::Relative);
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int channels = opus_packet_get_nb_channels(opusData);
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if (channels == OPUS_INVALID_PACKET)
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continue;
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bool mono = channels == 1;
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OpusDecoder* decoder = mono ? speaker->decoderMono.get() : speaker->decoderStereo.get();
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int samples = opus_decoder_get_nb_samples(decoder, opusData, opusLength);
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if (samples < 0)
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throw VoiceException(strf("Decoder error: {}", opus_strerror(samples)), false);
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size_t decodeBufferSize = samples * sizeof(opus_int16) * (size_t)channels;
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opus_int16* decodeBuffer = (opus_int16*)malloc(decodeBufferSize);
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int decodedSamples = opus_decode(decoder, opusData, opusLength, decodeBuffer, decodeBufferSize, 0);
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if (decodedSamples < 0) {
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free(decodeBuffer);
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throw VoiceException(strf("Decoder error: {}", opus_strerror(samples)), false);
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}
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Logger::info("Voice: decoded Opus chunk {} bytes big", opusLength);
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static auto getCVT = [](int channels) -> SDL_AudioCVT {
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SDL_AudioCVT cvt;
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SDL_BuildAudioCVT(&cvt, AUDIO_S16SYS, channels, VOICE_SAMPLE_RATE, AUDIO_S16, 2, 44100);
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return cvt;
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};
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//TODO: This isn't the best way to resample to 44100 hz because SDL_ConvertAudio is not for streamed audio.
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static SDL_AudioCVT monoCVT = getCVT(1);
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static SDL_AudioCVT stereoCVT = getCVT(2);
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SDL_AudioCVT& cvt = mono ? monoCVT : stereoCVT;
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cvt.len = decodedSamples * sizeof(opus_int16) * (size_t)channels;
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cvt.buf = (Uint8*)realloc(decodeBuffer, (size_t)(cvt.len * cvt.len_mult));
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SDL_ConvertAudio(&cvt);
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size_t reSamples = (size_t)cvt.len_cvt / 2;
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speaker->decibelLevel = getAudioLoudness((int16_t*)cvt.buf, reSamples);
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speaker->audioStream->take((opus_int16*)realloc(cvt.buf, cvt.len_cvt), reSamples);
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playSpeaker(speaker, channels);
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}
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return true;
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}
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catch (StarException const& e) {
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Logger::error("Voice: Error receiving voice data for speaker #{} ('{}'): {}", speaker->speakerId, speaker->name, e.what());
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return false;
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}
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}
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void Voice::setInput(bool input) {
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m_lastInputTime = input ? Time::monotonicMilliseconds() + 1000 : 0;
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}
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OpusDecoder* Voice::createDecoder(int channels) {
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int error;
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OpusDecoder* decoder = opus_decoder_create(VOICE_SAMPLE_RATE, channels, &error);
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if (error != OPUS_OK)
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throw VoiceException::format("Could not create decoder: {}", opus_strerror(error));
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else
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return decoder;
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}
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OpusEncoder* Voice::createEncoder(int channels) {
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int error;
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OpusEncoder* encoder = opus_encoder_create(VOICE_SAMPLE_RATE, channels, OPUS_APPLICATION_AUDIO, &error);
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if (error != OPUS_OK)
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throw VoiceException::format("Could not create encoder: {}", opus_strerror(error));
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else
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return encoder;
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}
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void Voice::resetEncoder() {
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int channels = encoderChannels();
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m_encoder.reset(createEncoder(channels));
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opus_encoder_ctl(m_encoder.get(), OPUS_SET_BITRATE(channels == 2 ? 50000 : 24000));
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}
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void Voice::openDevice() {
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closeDevice();
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m_applicationController->openAudioInputDevice(
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m_deviceName ? m_deviceName->utf8Ptr() : nullptr,
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VOICE_SAMPLE_RATE,
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m_deviceChannels = encoderChannels(),
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this,
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[](void* userdata, uint8_t* stream, int len) {
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((Voice*)(userdata))->readAudioData(stream, len);
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}
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);
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m_deviceOpen = true;
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}
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void Voice::closeDevice() {
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if (!m_deviceOpen)
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return;
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m_applicationController->closeAudioInputDevice();
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m_deviceOpen = false;
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}
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bool Voice::playSpeaker(SpeakerPtr const& speaker, int channels) {
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unsigned int minSamples = speaker->minimumPlaySamples * channels;
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if (speaker->playing || speaker->audioStream->samples < minSamples)
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return false;
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speaker->playing = true;
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MutexLocker lock(m_activeSpeakersMutex);
|
|
m_activeSpeakers.insert(speaker);
|
|
return true;
|
|
}
|
|
|
|
} |