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Add loopback, fix mono and leak

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This commit is contained in:
Kae 2023-07-19 18:59:35 +10:00
parent 3cdbf8bf01
commit 35b1c36b17
4 changed files with 43 additions and 27 deletions
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2
assets/opensb/scripts/opensb/universeclient/voicemanager.lua
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@ -19,7 +19,7 @@ local LINE_COLOR = {50, 210, 255, 255}
local FONT_DIRECTIVES = "?border=1;333;3337?border=1;333;3330"
local NAME_PREFIX = "^noshadow,white,set;"
local function dbToLoudness(db) return 2 ^ (db / 6) end
local function dbToLoudness(db) return 2 ^ (db / 8) end
local canvas

1
source/client/StarClientApplication.cpp
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@ -892,6 +892,7 @@ void ClientApplication::updateRunning() {
}
if (auto mainPlayer = m_universeClient->mainPlayer()) {
auto localSpeaker = m_voice->localSpeaker();
localSpeaker->position = mainPlayer->position();
localSpeaker->entityId = mainPlayer->entityId();
localSpeaker->name = mainPlayer->name();
}

63
source/frontend/StarVoice.cpp
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@ -63,11 +63,17 @@ float getAudioLoudness(int16_t* data, size_t samples, float volume = 1.0f) {
struct VoiceAudioStream {
// TODO: This should really be a ring buffer instead.
std::queue<int16_t> samples;
SDL_AudioStream* sdlAudioStream;
SDL_AudioStream* sdlAudioStreamMono;
SDL_AudioStream* sdlAudioStreamStereo;
Mutex mutex;
VoiceAudioStream() : sdlAudioStream(SDL_NewAudioStream(AUDIO_S16, 2, 48000, AUDIO_S16SYS, 2, 44100)) {};
~VoiceAudioStream() { SDL_FreeAudioStream(sdlAudioStream); }
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;
@ -78,11 +84,12 @@ struct VoiceAudioStream {
return sample;
}
size_t resample(int16_t* in, size_t inSamples, std::vector<int16_t>& out) {
SDL_AudioStreamPut(sdlAudioStream, in, inSamples * sizeof(int16_t));
if (int available = SDL_AudioStreamAvailable(sdlAudioStream)) {
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(sdlAudioStream, out.data(), available);
SDL_AudioStreamGet(stream, out.data(), available);
return available;
}
return 0;
@ -171,6 +178,9 @@ void Voice::loadJson(Json const& config) {
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)))
m_clientSpeaker->playing = false;
if (auto inputMode = config.optString("inputMode")) {
if (change(m_inputMode, VoiceInputModeNames.getLeft(*inputMode)))
@ -273,10 +283,14 @@ void Voice::readAudioData(uint8_t* stream, int len) {
}
}
if (active && !m_clientSpeaker->playing)
m_clientSpeaker->lastPlayTime = now;
if (!m_loopBack) {
if (active && !m_clientSpeaker->playing)
m_clientSpeaker->lastPlayTime = now;
if (!(m_clientSpeaker->playing = active))
m_clientSpeaker->playing = active;
}
if (!active)
return;
MutexLocker captureLock(m_captureMutex);
@ -311,7 +325,6 @@ void Voice::mix(int16_t* buffer, size_t frameCount, unsigned channels) {
VoiceAudioStream* audio = speaker->audioStream.get();
MutexLocker audioLock(audio->mutex);
if (!audio->samples.empty()) {
SDL_AudioStream* sdlStream = audio->sdlAudioStream;
if (!speaker->muted) {
mix = true;
for (size_t i = 0; i != samples; ++i)
@ -440,6 +453,8 @@ int Voice::send(DataStreamBuffer& out, size_t budget) {
}
m_lastSentTime = Time::monotonicMilliseconds();
if (m_loopBack)
receive(m_clientSpeaker, { out.ptr(), out.size() });
return 1;
}
@ -472,30 +487,26 @@ bool Voice::receive(SpeakerPtr speaker, std::string_view view) {
if (samples < 0)
throw VoiceException(strf("Decoder error: {}", opus_strerror(samples)), false);
size_t decodeBufferSize = samples * sizeof(opus_int16) * (size_t)channels;
opus_int16* decodeBuffer = (opus_int16*)malloc(decodeBufferSize);
m_decodeBuffer.resize(samples * (size_t)channels);
int decodedSamples = opus_decode(decoder, opusData, opusLength, decodeBuffer, decodeBufferSize, 0);
int decodedSamples = opus_decode(decoder, opusData, opusLength, m_decodeBuffer.data(), m_decodeBuffer.size() * sizeof(int16_t), 0);
if (decodedSamples <= 0) {
free(decodeBuffer);
if (decodedSamples < 0)
throw VoiceException(strf("Decoder error: {}", opus_strerror(samples)), false);
return true;
}
decodedSamples *= channels;
//Logger::info("Voice: decoded Opus chunk {} bytes -> {} samples", opusLength, decodedSamples);
//Logger::info("Voice: decoded Opus chunk {} bytes -> {} samples", opusLength, decodedSamples * channels);
speaker->audioStream->resample(m_decodeBuffer.data(), (size_t)decodedSamples * channels, m_resampleBuffer, mono);
{
std::vector<int16_t> resamBuffer(decodedSamples, 0);
speaker->audioStream->resample(decodeBuffer, decodedSamples, resamBuffer);
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) * (size_t)channels;
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();
@ -507,13 +518,13 @@ bool Voice::receive(SpeakerPtr speaker, std::string_view view) {
speaker->lastReceiveTime = now;
if (mono) {
for (int16_t sample : resamBuffer) {
for (int16_t sample : m_resampleBuffer) {
samples.push(sample);
samples.push(sample);
}
}
else {
for (int16_t sample : resamBuffer)
for (int16_t sample : m_resampleBuffer)
samples.push(sample);
}
}
@ -589,8 +600,7 @@ void Voice::closeDevice() {
}
bool Voice::playSpeaker(SpeakerPtr const& speaker, int channels) {
unsigned int minSamples = speaker->minimumPlaySamples * channels;
if (speaker->playing || speaker->audioStream->samples.size() < minSamples)
if (speaker->playing || speaker->audioStream->samples.size() < speaker->minimumPlaySamples)
return false;
if (!speaker->playing) {
@ -632,7 +642,8 @@ void Voice::thread() {
samples[i] *= m_inputVolume;
}
m_clientSpeaker->decibelLevel = getAudioLoudness(samples.data(), samples.size());
if (!m_loopBack)
m_clientSpeaker->decibelLevel = getAudioLoudness(samples.data(), samples.size());
if (int encodedSize = opus_encode(m_encoder.get(), samples.data(), VOICE_FRAME_SIZE, (unsigned char*)encoded.ptr(), encoded.size())) {
if (encodedSize == 1)

4
source/frontend/StarVoice.hpp
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@ -180,6 +180,7 @@ private:
int64_t m_nextSaveTime = 0;
bool m_enabled = true;
bool m_inputEnabled = false;
bool m_loopBack = false;
int m_deviceChannels = 1;
bool m_deviceOpen = false;
@ -192,6 +193,9 @@ private:
ConditionVariable m_threadCond;
atomic<bool> m_stopThread;
std::vector<int16_t> m_decodeBuffer;
std::vector<int16_t> m_resampleBuffer;
ApplicationControllerPtr m_applicationController;
struct EncodedChunk {