Merge branch 'voice'

1 files changed, 716 insertions, 0 deletions

diff --git a/source/frontend/StarVoice.cpp b/source/frontend/StarVoice.cpp
new file mode 100644
index 0000000..82205bd
--- /dev/null
+++ b/source/frontend/StarVoice.cpp
@@ -0,0 +1,716 @@
+#include "StarVoice.hpp"
+#include "StarFormat.hpp"
+#include "StarJsonExtra.hpp"
+#include "StarApplicationController.hpp"
+#include "StarTime.hpp"
+#include "StarRoot.hpp"
+#include "StarLogging.hpp"
+#include "StarInterpolation.hpp"
+#include "opus/include/opus.h"
+
+#include "SDL.h"
+
+constexpr int VOICE_SAMPLE_RATE = 48000;
+constexpr int VOICE_FRAME_SIZE = 960;
+
+constexpr int VOICE_MAX_FRAME_SIZE = 6 * VOICE_FRAME_SIZE;
+constexpr int VOICE_MAX_PACKET_SIZE = 3 * 1276;
+
+constexpr uint16_t VOICE_VERSION = 1;
+
+namespace Star {
+
+EnumMap<VoiceInputMode> const VoiceInputModeNames{
+  {VoiceInputMode::VoiceActivity, "VoiceActivity"},
+  {VoiceInputMode::PushToTalk, "PushToTalk"}
+};
+
+EnumMap<VoiceChannelMode> const VoiceChannelModeNames{
+  {VoiceChannelMode::Mono, "Mono"},
+  {VoiceChannelMode::Stereo, "Stereo"}
+};
+
+inline float getAudioChunkLoudness(int16_t* data, size_t samples, float volume) {
+	if (!samples)
+		return 0.f;
+
+	double rms = 0.;
+	for (size_t i = 0; i != samples; ++i) {
+		float sample = ((float)data[i] / 32767.f) * volume;
+		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, float volume = 1.0f) {
+	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, volume);
+		if (level > highest)
+      highest = level;
+	}
+
+	return highest;
+}
+
+struct VoiceAudioStream {
+  // TODO: This should really be a ring buffer instead.
+  std::queue<int16_t> samples;
+	SDL_AudioStream* sdlAudioStreamMono;
+	SDL_AudioStream* sdlAudioStreamStereo;
+  Mutex mutex;
+
+	VoiceAudioStream()
+		: sdlAudioStreamMono  (SDL_NewAudioStream(AUDIO_S16, 1, 48000, AUDIO_S16SYS, 1, 44100))
+	  , sdlAudioStreamStereo(SDL_NewAudioStream(AUDIO_S16, 2, 48000, AUDIO_S16SYS, 2, 44100)) {};
+	~VoiceAudioStream() {
+		SDL_FreeAudioStream(sdlAudioStreamMono);
+		SDL_FreeAudioStream(sdlAudioStreamStereo);
+	}
+
+	inline int16_t take() {
+		int16_t sample = 0;
+		if (!samples.empty()) {
+			sample = samples.front();
+			samples.pop();
+		}
+		return sample;
+	}
+
+	size_t resample(int16_t* in, size_t inSamples, std::vector<int16_t>& out, bool mono) {
+		SDL_AudioStream* stream = mono ? sdlAudioStreamMono : sdlAudioStreamStereo;
+		SDL_AudioStreamPut(stream, in, inSamples * sizeof(int16_t));
+		if (int available = SDL_AudioStreamAvailable(stream)) {
+			out.resize(available / 2);
+			SDL_AudioStreamGet(stream, out.data(), available);
+			return available;
+		}
+		return 0;
+	}
+};
+
+Voice::Speaker::Speaker(SpeakerId id)
+  : decoderMono  (createDecoder(1), opus_decoder_destroy)
+  , decoderStereo(createDecoder(2), opus_decoder_destroy) {
+  speakerId = id;
+  audioStream = make_shared<VoiceAudioStream>();
+}
+
+Json Voice::Speaker::toJson() const {
+	return JsonObject{
+		{"speakerId",      speakerId},
+		{"entityId",       entityId },
+		{"name",           name     },
+		{"playing",        (bool)playing},
+		{"muted",          (bool)muted  },
+		{"decibels",       (float)decibelLevel},
+		{"smoothDecibels", (float)smoothDb    },
+		{"position",       jsonFromVec2F(position)}
+	};
+}
+
+Voice* Voice::s_singleton;
+
+Voice* Voice::singletonPtr() {
+  return s_singleton;
+}
+
+Voice& Voice::singleton() {
+  if (!s_singleton)
+    throw VoiceException("Voice::singleton() called with no Voice instance available");
+  else
+    return *s_singleton;
+}
+
+Voice::Voice(ApplicationControllerPtr appController) : m_encoder(nullptr, opus_encoder_destroy) {
+  if (s_singleton)
+    throw VoiceException("Singleton Voice has been constructed twice");
+
+  m_clientSpeaker = make_shared<Speaker>(m_speakerId);
+  m_inputMode = VoiceInputMode::PushToTalk;
+  m_channelMode = VoiceChannelMode::Mono;
+  m_applicationController = appController;
+
+	m_stopThread = false;
+	m_thread = Thread::invoke("Voice::thread", mem_fn(&Voice::thread), this);
+
+  s_singleton = this;
+}
+
+Voice::~Voice() {
+	m_stopThread = true;
+
+	{
+		MutexLocker locker(m_threadMutex);
+		m_threadCond.broadcast();
+	}
+
+	m_thread.finish();
+
+	if (m_nextSaveTime)
+		save();
+
+	closeDevice();
+
+  s_singleton = nullptr;
+}
+
+void Voice::init() {
+	resetEncoder();
+	resetDevice();
+}
+
+
+template <typename T>
+inline bool change(T& value, T newValue, bool& out) {
+	bool changed = value != newValue;
+	out |= changed;
+	value = move(newValue);
+	return changed;
+}
+
+void Voice::loadJson(Json const& config, bool skipSave) {
+  // Not all keys are required
+
+	bool changed = false;
+
+	{
+		bool enabled = shouldEnableInput();
+		m_enabled      = config.getBool("enabled", m_enabled);
+		m_inputEnabled = config.getBool("inputEnabled", m_inputEnabled);
+		if (shouldEnableInput() != enabled) {
+			changed = true;
+			resetDevice();
+		}
+	}
+
+	if (config.contains("deviceName") // Make sure null-type key exists
+	&& change(m_deviceName, config.optString("deviceName"), changed))
+		resetDevice();
+
+  m_threshold    = config.getFloat("threshold", m_threshold);
+  m_inputVolume  = config.getFloat("inputVolume", m_inputVolume);
+  m_outputVolume = config.getFloat("outputVolume",   m_outputVolume);
+	
+	if (change(m_loopback, config.getBool("loopback", m_loopback), changed))
+		m_clientSpeaker->playing = false;
+
+	if (auto inputMode = config.optString("inputMode")) {
+		if (change(m_inputMode, VoiceInputModeNames.getLeft(*inputMode), changed))
+			m_lastInputTime = 0;
+	}
+
+	if (auto channelMode = config.optString("channelMode")) {
+		if (change(m_channelMode, VoiceChannelModeNames.getLeft(*channelMode), changed)) {
+			closeDevice();
+			resetEncoder();
+			resetDevice();
+		}
+	}
+
+	if (changed && !skipSave)
+		scheduleSave();
+}
+
+
+
+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)},
+		{"loopback",     m_loopback},
+    {"version",      1}
+  };
+}
+
+void Voice::save() const {
+  if (Root* root = Root::singletonPtr()) {
+    if (auto config = root->configuration())
+      config->set("voice", saveJson());
+  }
+}
+
+void Voice::scheduleSave() {
+  if (!m_nextSaveTime)
+		m_nextSaveTime = Time::monotonicMilliseconds() + 2000;
+}
+
+Voice::SpeakerPtr Voice::setLocalSpeaker(SpeakerId speakerId) {
+  if (m_speakers.contains(m_speakerId))
+    m_speakers.remove(m_speakerId);
+
+  m_clientSpeaker->speakerId = m_speakerId = speakerId;
+  return m_speakers.insert(m_speakerId, m_clientSpeaker).first->second;
+}
+
+Voice::SpeakerPtr Voice::localSpeaker() {
+	return m_clientSpeaker;
+}
+
+Voice::SpeakerPtr Voice::speaker(SpeakerId speakerId) {
+  if (m_speakerId == speakerId)
+    return m_clientSpeaker;
+  else {
+    if (SpeakerPtr const* ptr = m_speakers.ptr(speakerId))
+      return *ptr;
+    else
+      return m_speakers.emplace(speakerId, make_shared<Speaker>(speakerId)).first->second;
+  }
+}
+
+HashMap<Voice::SpeakerId, Voice::SpeakerPtr>& Voice::speakers() {
+	return m_speakers;
+}
+
+List<Voice::SpeakerPtr> Voice::sortedSpeakers(bool onlyPlaying) {
+	List<SpeakerPtr> result;
+
+	auto sorter = [](SpeakerPtr const& a, SpeakerPtr const& b) -> bool {
+		if (a->lastPlayTime != b->lastPlayTime)
+			return a->lastPlayTime < b->lastPlayTime;
+		else
+			return a->speakerId < b->speakerId;
+	};
+
+	for (auto& p : m_speakers) {
+		if (!onlyPlaying || p.second->playing)
+			result.insertSorted(p.second, sorter);
+	}
+
+	return result;
+}
+
+void Voice::clearSpeakers() {
+	auto it = m_speakers.begin();
+	while (it != m_speakers.end()) {
+		if (it->second == m_clientSpeaker)
+			it = ++it;
+		else
+			it = m_speakers.erase(it);
+	}
+}
+
+void Voice::readAudioData(uint8_t* stream, int len) {
+	auto now = Time::monotonicMilliseconds();
+	bool active = m_encoder && m_encodedChunksLength < 2048
+       && (m_inputMode == VoiceInputMode::VoiceActivity || now < m_lastInputTime);
+
+	size_t sampleCount = len / 2;
+
+	if (active) {
+		float decibels = getAudioLoudness((int16_t*)stream, sampleCount);
+
+		if (m_inputMode == VoiceInputMode::VoiceActivity) {
+			if (decibels > m_threshold)
+				m_lastThresholdTime = now;
+			active = now - m_lastThresholdTime < 50;
+		}
+	}
+
+	m_clientSpeaker->decibelLevel = getAudioLoudness((int16_t*)stream, sampleCount, m_inputVolume);
+
+	if (!m_loopback) {
+		if (active && !m_clientSpeaker->playing)
+			m_clientSpeaker->lastPlayTime = now;
+
+		m_clientSpeaker->playing = active;
+	}
+
+	MutexLocker captureLock(m_captureMutex);
+	if (active) {
+		m_capturedChunksFrames += sampleCount / m_deviceChannels;
+		auto data = (opus_int16*)malloc(len);
+		memcpy(data, stream, len);
+		m_capturedChunks.emplace(data, sampleCount); // takes ownership
+		m_threadCond.signal();
+	}
+	else { // Clear out any residual data so they don't manifest at the start of the next encode, whenever that is
+		while (!m_capturedChunks.empty())
+			m_capturedChunks.pop();
+
+		m_capturedChunksFrames = 0;
+	}
+}
+
+void Voice::mix(int16_t* buffer, size_t frameCount, unsigned channels) {
+	size_t samples = frameCount * channels;
+	static std::vector<int16_t> finalBuffer, speakerBuffer;
+	static std::vector<int32_t> sharedBuffer; //int32 to reduce clipping
+	speakerBuffer.resize(samples);
+	sharedBuffer.resize(samples);
+
+	bool mix = false;
+	{
+		MutexLocker lock(m_activeSpeakersMutex);
+		auto it = m_activeSpeakers.begin();
+		while (it != m_activeSpeakers.end()) {
+			SpeakerPtr const& speaker = *it;
+			VoiceAudioStream* audio = speaker->audioStream.get();
+			MutexLocker audioLock(audio->mutex);
+			if (speaker->playing && !audio->samples.empty()) {
+				for (size_t i = 0; i != samples; ++i)
+					speakerBuffer[i] = audio->take();
+
+				if (speaker != m_clientSpeaker)
+					speaker->decibelLevel = getAudioLoudness(speakerBuffer.data(), samples);
+
+				if (!speaker->muted) {
+					mix = true;
+
+					float volume = speaker->volume;
+					Array2F levels = speaker->channelVolumes;
+					for (size_t i = 0; i != samples; ++i)
+						sharedBuffer[i] += (int32_t)(speakerBuffer[i]) * levels[i % 2] * volume;
+					//Blends the weaker channel into the stronger one,
+					/* unused, is a bit too strong on stereo music.
+					float maxLevel = max(levels[0], levels[1]);
+					float leftToRight = maxLevel != 0.0f ? 1.0f - (levels[0] / maxLevel) : 0.0f;
+					float rightToLeft = maxLevel != 0.0f ? 1.0f - (levels[1] / maxLevel) : 0.0f;
+
+					int16_t* speakerData = speakerBuffer.data();
+					int32_t* sharedData  = sharedBuffer.data();
+					for (size_t i = 0; i != frameCount; ++i) {
+						auto leftSample  = (float)*speakerData++;
+						auto rightSample = (float)*speakerData++;
+						
+					  if (rightToLeft != 0.0f)
+							leftSample  = ( leftSample + rightSample * rightToLeft) / (1.0f + rightToLeft);
+						if (leftToRight != 0.0f)
+						  rightSample = (rightSample +  leftSample * leftToRight) / (1.0f + leftToRight);
+
+						*sharedData++ += (int32_t)leftSample  * levels[0];
+						*sharedData++ += (int32_t)rightSample * levels[1];
+					}
+					//*/
+				}
+				++it;
+			}
+			else {
+				speaker->playing = false;
+				if (speaker != m_clientSpeaker)
+					speaker->decibelLevel = -96.0f;
+				it = m_activeSpeakers.erase(it);
+			}
+		}
+	}
+
+	if (mix) {
+		finalBuffer.resize(sharedBuffer.size(), 0);
+
+		float vol = m_outputVolume;
+		for (size_t i = 0; i != sharedBuffer.size(); ++i)
+			finalBuffer[i] = (int16_t)clamp<int>(sharedBuffer[i] * vol, INT16_MIN, INT16_MAX);
+
+		SDL_MixAudioFormat((Uint8*)buffer, (Uint8*)finalBuffer.data(), AUDIO_S16, finalBuffer.size() * sizeof(int16_t), SDL_MIX_MAXVOLUME);
+		memset(sharedBuffer.data(), 0, sharedBuffer.size() * sizeof(int32_t));
+	}
+}
+
+void Voice::update(PositionalAttenuationFunction positionalAttenuationFunction) {
+  for (auto& entry : m_speakers) {
+    if (SpeakerPtr& speaker = entry.second) {
+			if (positionalAttenuationFunction) {
+				speaker->channelVolumes = {
+					1.0f - positionalAttenuationFunction(0, speaker->position, 1.0f),
+					1.0f - positionalAttenuationFunction(1, speaker->position, 1.0f)
+				};
+			}
+			else
+				speaker->channelVolumes = Vec2F::filled(1.0f);
+				
+			auto& dbHistory = speaker->dbHistory;
+			memcpy(&dbHistory[1], &dbHistory[0], (dbHistory.size() - 1) * sizeof(float));
+			dbHistory[0] = speaker->decibelLevel;
+			float smoothDb = 0.0f;
+			for (float dB : dbHistory)
+				smoothDb += dB;
+
+			speaker->smoothDb = smoothDb / dbHistory.size();
+    }
+  }
+
+  if (m_nextSaveTime && Time::monotonicMilliseconds() > m_nextSaveTime) {
+		m_nextSaveTime = 0;
+    save();
+  }
+}
+
+
+void Voice::setDeviceName(Maybe<String> deviceName) {
+  if (m_deviceName == deviceName)
+    return;
+
+  m_deviceName = deviceName;
+  if (m_deviceOpen)
+    openDevice();
+}
+
+StringList Voice::availableDevices() {
+	int devices = SDL_GetNumAudioDevices(1);
+	StringList deviceList;
+	if (devices > 0) {
+		deviceList.reserve(devices);
+		for (size_t i = 0; i != devices; ++i)
+			deviceList.emplace_back(SDL_GetAudioDeviceName(i, 1));
+	}
+	deviceList.sort();
+	return deviceList;
+}
+
+int Voice::send(DataStreamBuffer& out, size_t budget) {
+	out.setByteOrder(ByteOrder::LittleEndian);
+	out.write<uint16_t>(VOICE_VERSION);
+
+	MutexLocker encodeLock(m_encodeMutex);
+
+	if (m_encodedChunks.empty())
+		return 0;
+
+	std::vector<ByteArray> encodedChunks = move(m_encodedChunks);
+	size_t encodedChunksLength = m_encodedChunksLength;
+	m_encodedChunksLength = 0;
+
+	encodeLock.unlock();
+
+	for (auto& chunk : encodedChunks) {
+		out.write<uint32_t>(chunk.size());
+		out.writeBytes(chunk);
+		if (budget && (budget -= min<size_t>(budget, chunk.size())) == 0)
+			break;
+	}
+
+	m_lastSentTime = Time::monotonicMilliseconds();
+	if (m_loopback)
+		receive(m_clientSpeaker, { out.ptr(), out.size() });
+	return 1;
+}
+
+bool Voice::receive(SpeakerPtr speaker, std::string_view view) {
+	if (!m_enabled || !speaker || view.empty())
+		return false;
+
+	try {
+		DataStreamExternalBuffer reader(view.data(), view.size());
+		reader.setByteOrder(ByteOrder::LittleEndian);
+
+		if (reader.read<uint16_t>() > VOICE_VERSION)
+			return false;
+
+		uint32_t opusLength = 0;
+		while (!reader.atEnd()) {
+			reader >> opusLength;
+			if (reader.pos() + opusLength > reader.size())
+				throw VoiceException("Opus packet length goes past end of buffer"s, false);
+			auto opusData = (unsigned char*)reader.ptr() + reader.pos();
+			reader.seek(opusLength, IOSeek::Relative);
+
+			int channels = opus_packet_get_nb_channels(opusData);
+			if (channels == OPUS_INVALID_PACKET)
+				continue;
+
+			bool mono = channels == 1;
+			OpusDecoder* decoder = mono ? speaker->decoderMono.get() : speaker->decoderStereo.get();
+			int samples = opus_decoder_get_nb_samples(decoder, opusData, opusLength);
+			if (samples < 0)
+				throw VoiceException(strf("Decoder error: {}", opus_strerror(samples)), false);
+
+			m_decodeBuffer.resize(samples * (size_t)channels);
+
+			int decodedSamples = opus_decode(decoder, opusData, opusLength, m_decodeBuffer.data(), m_decodeBuffer.size() * sizeof(int16_t), 0);
+			if (decodedSamples <= 0) {
+				if (decodedSamples < 0)
+				  throw VoiceException(strf("Decoder error: {}", opus_strerror(samples)), false);
+				return true;
+			}
+
+			//Logger::info("Voice: decoded Opus chunk {} bytes -> {} samples", opusLength, decodedSamples * channels);
+
+			speaker->audioStream->resample(m_decodeBuffer.data(), (size_t)decodedSamples * channels, m_resampleBuffer, mono);
+
+			{
+			  MutexLocker lock(speaker->audioStream->mutex);
+				auto& samples = speaker->audioStream->samples;
+
+				auto now = Time::monotonicMilliseconds();
+				if (now - speaker->lastReceiveTime < 1000) {
+					auto limit = (size_t)speaker->minimumPlaySamples + 22050;
+					if (samples.size() > limit) { // skip ahead if we're getting too far
+						for (size_t i = samples.size(); i >= limit; --i)
+							samples.pop();
+					}
+				}
+				else
+					samples = std::queue<int16_t>();
+
+				speaker->lastReceiveTime = now;
+
+				if (mono) {
+					for (int16_t sample : m_resampleBuffer) {
+						samples.push(sample);
+						samples.push(sample);
+					}
+				}
+				else {
+					for (int16_t sample : m_resampleBuffer)
+						samples.push(sample);
+				}
+			}
+			playSpeaker(speaker, channels);
+		}
+		return true;
+	}
+	catch (StarException const& e) {
+		Logger::error("Voice: Error receiving voice data for speaker #{} ('{}'): {}", speaker->speakerId, speaker->name, e.what());
+		return false;
+	}
+}
+
+void Voice::setInput(bool input) {
+  m_lastInputTime = (m_deviceOpen && input) ? Time::monotonicMilliseconds() + 1000 : 0;
+}
+
+OpusDecoder* Voice::createDecoder(int channels) {
+  int error;
+  OpusDecoder* decoder = opus_decoder_create(VOICE_SAMPLE_RATE, channels, &error);
+  if (error != OPUS_OK)
+    throw VoiceException::format("Could not create decoder: {}", opus_strerror(error));
+  else
+    return decoder;
+}
+
+OpusEncoder* Voice::createEncoder(int channels) {
+  int error;
+  OpusEncoder* encoder = opus_encoder_create(VOICE_SAMPLE_RATE, channels, OPUS_APPLICATION_AUDIO, &error);
+  if (error != OPUS_OK)
+    throw VoiceException::format("Could not create encoder: {}", opus_strerror(error));
+  else
+    return encoder;
+}
+
+void Voice::resetEncoder() {
+  int channels = encoderChannels();
+	MutexLocker locker(m_threadMutex);
+  m_encoder.reset(createEncoder(channels));
+  opus_encoder_ctl(m_encoder.get(), OPUS_SET_BITRATE(channels == 2 ? 50000 : 24000));
+}
+
+void Voice::resetDevice() {
+	if (shouldEnableInput())
+		openDevice();
+	else
+		closeDevice();
+}
+
+void Voice::openDevice() {
+  closeDevice();
+
+  m_applicationController->openAudioInputDevice(
+		m_deviceName ? m_deviceName->utf8Ptr() : nullptr,
+		VOICE_SAMPLE_RATE,
+		m_deviceChannels = encoderChannels(),
+		this,
+		[](void* userdata, uint8_t* stream, int len) {
+      ((Voice*)(userdata))->readAudioData(stream, len);
+    }
+	);
+
+  m_deviceOpen = true;
+}
+
+void Voice::closeDevice() {
+  if (!m_deviceOpen)
+    return;
+
+  m_applicationController->closeAudioInputDevice();
+	m_clientSpeaker->playing = false;
+	m_clientSpeaker->decibelLevel = -96.0f;
+  m_deviceOpen = false;
+}
+
+bool Voice::playSpeaker(SpeakerPtr const& speaker, int channels) {
+	if (speaker->playing || speaker->audioStream->samples.size() < speaker->minimumPlaySamples)
+		return false;
+
+	if (!speaker->playing) {
+		speaker->lastPlayTime = Time::monotonicMilliseconds();
+		speaker->playing = true;
+		MutexLocker lock(m_activeSpeakersMutex);
+		m_activeSpeakers.insert(speaker);
+	}
+	return true;
+}
+
+void Voice::thread() {
+	while (true) {
+		MutexLocker locker(m_threadMutex);
+
+		m_threadCond.wait(m_threadMutex);
+		if (m_stopThread)
+			return;
+
+		{
+			MutexLocker locker(m_captureMutex);
+		  ByteArray encoded(VOICE_MAX_PACKET_SIZE, 0);
+			size_t frameSamples = VOICE_FRAME_SIZE * (size_t)m_deviceChannels;
+			while (m_capturedChunksFrames >= VOICE_FRAME_SIZE) {
+				std::vector<opus_int16> samples;
+				samples.reserve(frameSamples);
+				size_t samplesLeft = frameSamples;
+				while (samplesLeft && !m_capturedChunks.empty()) {
+					auto& front = m_capturedChunks.front();
+					if (front.exhausted())
+						m_capturedChunks.pop();
+					else
+						samplesLeft -= front.takeSamples(samples, samplesLeft);
+				}
+				m_capturedChunksFrames -= VOICE_FRAME_SIZE;
+
+				if (m_inputVolume != 1.0f) {
+					for (size_t i = 0; i != samples.size(); ++i)
+						samples[i] *= m_inputVolume;
+				}
+
+				if (int encodedSize = opus_encode(m_encoder.get(), samples.data(), VOICE_FRAME_SIZE, (unsigned char*)encoded.ptr(), encoded.size())) {
+					if (encodedSize == 1)
+						continue;
+
+					encoded.resize(encodedSize);
+
+					{
+						MutexLocker lock(m_encodeMutex);
+						m_encodedChunks.emplace_back(move(encoded));
+						m_encodedChunksLength += encodedSize;
+
+						encoded = ByteArray(VOICE_MAX_PACKET_SIZE, 0);
+					}
+
+					//Logger::info("Voice: encoded Opus chunk {} samples -> {} bytes", frameSamples, encodedSize);
+				}
+				else if (encodedSize < 0)
+					Logger::error("Voice: Opus encode error {}", opus_strerror(encodedSize));
+			}
+		}
+
+		continue;
+
+		locker.unlock();
+		Thread::yield();
+	}
+	return;
+}
+
+}
\ No newline at end of file