more voice stuff
This commit is contained in:
parent
f14f77724d
commit
3b38825b34
@ -1,7 +1,8 @@
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{
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"opensb": {
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"groups": {
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"camera": { "name": "Camera" }
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"camera": { "name": "Camera" },
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"voice": { "name": "Voice" }
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},
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"name": "Open^#ebd74a;Starbound",
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"binds": {
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@ -21,13 +22,10 @@
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"group" : "camera",
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"name": "Zoom Out"
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},
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"test": {
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"default": [{
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"type": "key",
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"value": "C",
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"mods": ["LShift"]
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}],
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"name": "Test Bind"
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"pushToTalk": {
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"default": [],
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"group" : "voice",
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"name": "Push To Talk"
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}
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}
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}
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@ -344,8 +344,10 @@ public:
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SDL_AudioSpec obtained = {};
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m_sdlAudioInputDevice = SDL_OpenAudioDevice(name, 1, &desired, &obtained, 0);
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if (m_sdlAudioInputDevice)
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if (m_sdlAudioInputDevice) {
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Logger::info("Opened audio input device '{}'", SDL_GetAudioDeviceName(m_sdlAudioInputDevice, 1));
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SDL_PauseAudioDevice(m_sdlAudioInputDevice, 0);
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}
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else
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Logger::info("Failed to open audio input device: {}", SDL_GetError());
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@ -375,6 +375,14 @@ void ClientApplication::update() {
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else if (m_state > MainAppState::Title)
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updateRunning();
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{ // testing
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m_voice->setLocalSpeaker(0);
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m_voice->setInput(m_input->bindHeld("opensb", "pushToTalk"));
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DataStreamBuffer data;
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if (m_voice->send(data, 5000))
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m_voice->receive(m_voice->speaker(0), std::string_view(data.ptr(), data.size()));
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}
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m_guiContext->cleanup();
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m_edgeKeyEvents.clear();
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m_input->reset();
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@ -3,9 +3,9 @@
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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 <queue>
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#include "SDL.h"
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constexpr int VOICE_SAMPLE_RATE = 48000;
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@ -59,45 +59,10 @@ float getAudioLoudness(int16_t* data, size_t samples) {
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return highest;
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}
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struct VoiceAudioChunk {
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std::unique_ptr<int16_t[]> data;
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size_t remaining;
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size_t offset = 0;
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VoiceAudioChunk(int16_t* ptr, size_t size) {
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data.reset(ptr);
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remaining = size;
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offset = 0;
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}
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inline size_t takeSamples(std::vector<int16_t>& out, size_t count) {
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size_t toRead = std::min<size_t>(count, remaining);
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int16_t* start = data.get() + offset;
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out.insert(out.end(), start, start + toRead);
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offset += toRead;
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remaining -= toRead;
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return toRead;
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}
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//this one's unsafe
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inline int16_t takeSample() {
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--remaining;
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return *(data.get() + offset++);
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}
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inline bool exhausted() {
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return remaining == 0;
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}
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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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atomic<bool> muted = false;
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atomic<bool> playing = false;
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atomic<float> decibelLevel = 0.0f;
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atomic<Array<float, 2>> channelVolumes = Array<float, 2>::filled(1.0f);
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Mutex mutex;
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@ -225,8 +190,8 @@ void Voice::save() const {
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}
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void Voice::scheduleSave() {
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if (nextSaveTime == 0.0)
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nextSaveTime = Time::monotonicTime() + 2.0;
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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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@ -248,19 +213,130 @@ Voice::SpeakerPtr Voice::speaker(SpeakerId speakerId) {
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}
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}
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void Voice::getAudioData(uint8_t* stream, int len) {
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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);
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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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std::vector<opus_int16> takenSamples;
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while (m_capturedChunksFrames >= VOICE_FRAME_SIZE) {
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takenSamples.clear();
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size_t samplesToTake = VOICE_FRAME_SIZE * (size_t)m_deviceChannels;
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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 if ((samplesToTake -= front.takeSamples(takenSamples, samplesToTake)) == 0)
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break;
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}
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m_capturedChunksFrames -= VOICE_FRAME_SIZE;
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ByteArray encodedData(VOICE_MAX_PACKET_SIZE, 0);
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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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if (opus_int32 size = opus_encode(m_encoder.get(), takenSamples.data(), VOICE_FRAME_SIZE, (unsigned char*)encodedData.ptr(), VOICE_MAX_PACKET_SIZE)) {
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if (size == 1)
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continue;
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encodedData.resize(size);
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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 += size;
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Logger::info("Voice: encoded Opus chunk {} bytes big", size);
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}
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else if (size < 0) {
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Logger::error("Voice: Opus encode error {}", opus_strerror(size));
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}
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}
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}
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void Voice::mix(int16_t* buffer, size_t frames, unsigned channels) {
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void Voice::mix(int16_t* buffer, size_t samples, unsigned 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)std::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->audioStream->channelVolumes = {
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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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@ -268,9 +344,8 @@ void Voice::update(PositionalAttenuationFunction positionalAttenuationFunction)
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}
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}
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auto now = Time::monotonicTime();
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if (now > nextSaveTime) {
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nextSaveTime = 0.0;
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if (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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@ -285,6 +360,97 @@ void Voice::setDeviceName(Maybe<String> deviceName) {
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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_encoder || 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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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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@ -312,9 +478,17 @@ void Voice::resetEncoder() {
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void Voice::openDevice() {
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closeDevice();
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m_applicationController->openAudioInputDevice(m_deviceName ? m_deviceName->utf8Ptr() : nullptr, VOICE_SAMPLE_RATE, encoderChannels(), this, [](void* userdata, uint8_t* stream, int len) {
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((Voice*)(userdata))->getAudioData(stream, len);
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});
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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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@ -328,4 +502,15 @@ void Voice::closeDevice() {
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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);
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m_activeSpeakers.insert(speaker);
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return true;
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}
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}
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@ -6,8 +6,11 @@
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#include "StarGameTypes.hpp"
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#include "StarMaybe.hpp"
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#include "StarThread.hpp"
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#include "StarDataStreamDevices.hpp"
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#include "StarApplicationController.hpp"
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#include <queue>
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struct OpusDecoder;
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typedef std::unique_ptr<OpusDecoder, void(*)(OpusDecoder*)> OpusDecoderPtr;
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struct OpusEncoder;
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@ -27,6 +30,36 @@ STAR_CLASS(Voice);
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STAR_CLASS(VoiceAudioStream);
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STAR_CLASS(ApplicationController);
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struct VoiceAudioChunk {
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std::unique_ptr<int16_t[]> data;
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size_t remaining;
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size_t offset = 0;
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VoiceAudioChunk(int16_t* ptr, size_t size) {
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data.reset(ptr);
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remaining = size;
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offset = 0;
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}
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inline size_t takeSamples(std::vector<int16_t>& out, size_t count) {
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size_t toRead = std::min<size_t>(count, remaining);
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int16_t* start = data.get() + offset;
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out.insert(out.end(), start, start + toRead);
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offset += toRead;
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remaining -= toRead;
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return toRead;
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}
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//this one's unsafe
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inline int16_t takeSample() {
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--remaining;
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return *(data.get() + offset++);
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}
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inline bool exhausted() { return remaining == 0; }
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};
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class Voice {
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public:
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// Individual speakers are represented by their connection ID.
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@ -45,6 +78,13 @@ public:
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VoiceAudioStreamPtr audioStream;
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Mutex mutex;
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atomic<bool> muted = false;
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atomic<bool> playing = false;
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atomic<float> decibelLevel = 0.0f;
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atomic<Array<float, 2>> channelVolumes = Array<float, 2>::filled(1.0f);
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unsigned int minimumPlaySamples = 4096;
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Speaker(SpeakerId speakerId);
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};
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@ -77,7 +117,7 @@ public:
|
||||
SpeakerPtr speaker(SpeakerId speakerId);
|
||||
|
||||
// Called when receiving input audio data from SDL, on its own thread.
|
||||
void getAudioData(uint8_t* stream, int len);
|
||||
void readAudioData(uint8_t* stream, int len);
|
||||
|
||||
// Called to mix voice audio with the game.
|
||||
void mix(int16_t* buffer, size_t frames, unsigned channels);
|
||||
@ -87,6 +127,12 @@ public:
|
||||
|
||||
void setDeviceName(Maybe<String> device);
|
||||
|
||||
int send(DataStreamBuffer& out, size_t budget);
|
||||
bool receive(SpeakerPtr speaker, std::string_view view);
|
||||
|
||||
// Must be called every frame with input state, expires after 1s.
|
||||
void setInput(bool input = true);
|
||||
|
||||
inline int encoderChannels() const {
|
||||
return m_channelMode == VoiceChannelMode::Mono ? 1 : 2;
|
||||
}
|
||||
@ -99,10 +145,13 @@ private:
|
||||
void openDevice();
|
||||
void closeDevice();
|
||||
|
||||
bool playSpeaker(SpeakerPtr const& speaker, int channels);
|
||||
|
||||
SpeakerId m_speakerId = 0;
|
||||
SpeakerPtr m_clientSpeaker;
|
||||
HashMap<SpeakerId, SpeakerPtr> m_speakers;
|
||||
|
||||
Mutex m_activeSpeakersMutex;
|
||||
HashSet<SpeakerPtr> m_activeSpeakers;
|
||||
|
||||
OpusEncoderPtr m_encoder;
|
||||
@ -111,9 +160,14 @@ private:
|
||||
float m_inputVolume = 1.0f;
|
||||
float m_threshold = -50.0f;
|
||||
|
||||
int64_t m_lastSentTime = 0;
|
||||
int64_t m_lastInputTime = 0;
|
||||
int64_t m_lastThresholdTime = 0;
|
||||
int64_t m_nextSaveTime = 0;
|
||||
bool m_enabled = true;
|
||||
bool m_inputEnabled = true;
|
||||
|
||||
int m_deviceChannels = 1;
|
||||
bool m_deviceOpen = false;
|
||||
Maybe<String> m_deviceName;
|
||||
VoiceInputMode m_inputMode;
|
||||
@ -121,7 +175,23 @@ private:
|
||||
|
||||
ApplicationControllerPtr m_applicationController;
|
||||
|
||||
double nextSaveTime = 0.0f;
|
||||
struct EncodedChunk {
|
||||
std::unique_ptr<unsigned char[]> data;
|
||||
size_t size;
|
||||
|
||||
EncodedChunk(unsigned char* _data, size_t len) {
|
||||
data.reset(_data);
|
||||
size = len;
|
||||
}
|
||||
};
|
||||
|
||||
std::vector<ByteArray> m_encodedChunks;
|
||||
size_t m_encodedChunksLength = 0;
|
||||
|
||||
std::queue<VoiceAudioChunk> m_capturedChunks;
|
||||
size_t m_capturedChunksFrames = 0;
|
||||
|
||||
Mutex m_captureMutex;
|
||||
};
|
||||
|
||||
}
|
||||
|
Loading…
Reference in New Issue
Block a user