Only resample during mix, store samples a simpler way
Still need a better resampler, I think
This commit is contained in:
Kae
parent
73c5a17746
commit
b4a53e0706
@ -61,54 +61,9 @@ float getAudioLoudness(int16_t* data, size_t samples) {
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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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std::vector<int16_t> samples;
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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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@ -269,14 +224,12 @@ void Voice::readAudioData(uint8_t* stream, int len) {
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}
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void Voice::mix(int16_t* buffer, size_t frameCount, unsigned channels) {
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static std::vector<int16_t> finalBuffer;
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static std::vector<int32_t> voiceBuffer;
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static std::vector<int16_t> resampled;
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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 = voiceMixBuffer.data();
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memset(mixBuf, 0, samples * sizeof(int32_t));
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//read into buffer now
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resampled.resize(samples, 0);
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bool mix = false;
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{
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MutexLocker lock(m_activeSpeakersMutex);
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@ -285,16 +238,17 @@ void Voice::mix(int16_t* buffer, size_t frameCount, unsigned channels) {
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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 (!audio->samples.empty()) {
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std::vector<int16_t> samples = move(audio->samples);
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audioLock.unlock();
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if (!speaker->muted) {
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mix = true;
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if (voiceBuffer.size() < samples.size())
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voiceBuffer.resize(samples.size(), 0);
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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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for (size_t i = 0; i != samples.size(); ++i)
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voiceBuffer[i] += (int32_t)(samples[i]) * channelVolumes[i % 2];
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}
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++it;
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}
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@ -304,15 +258,27 @@ void Voice::mix(int16_t* buffer, size_t frameCount, unsigned channels) {
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}
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}
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}
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static std::unique_ptr<SDL_AudioStream, void(*)(SDL_AudioStream*)> audioStream
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(SDL_NewAudioStream(AUDIO_S16, 2, 48000, AUDIO_S16SYS, 2, 44100), SDL_FreeAudioStream);
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if (mix) {
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int16_t* finBuf = finalMixBuffer.data();
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finalBuffer.resize(voiceBuffer.size(), 0);
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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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for (size_t i = 0; i != voiceBuffer.size(); ++i)
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finalBuffer[i] = (int16_t)clamp<int>(voiceBuffer[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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SDL_AudioStreamPut(audioStream.get(), finalBuffer.data(), finalBuffer.size() * sizeof(int16_t));
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}
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if (size_t available = min<size_t>(samples * sizeof(int16_t), SDL_AudioStreamAvailable(audioStream.get()))) {
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SDL_AudioStreamGet(audioStream.get(), resampled.data(), available);
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SDL_MixAudioFormat((Uint8*)buffer, (Uint8*)resampled.data(), AUDIO_S16, samples * sizeof(int16_t), SDL_MIX_MAXVOLUME);
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}
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resampled.clear();
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voiceBuffer.clear();
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}
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void Voice::update(PositionalAttenuationFunction positionalAttenuationFunction) {
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@ -400,30 +366,22 @@ bool Voice::receive(SpeakerPtr speaker, std::string_view view) {
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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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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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if (decodedSamples < 0)
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throw VoiceException(strf("Decoder error: {}", opus_strerror(samples)), false);
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return true;
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}
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Logger::info("Voice: decoded Opus chunk {} bytes big", opusLength);
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decodedSamples *= channels;
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//Logger::info("Voice: decoded Opus chunk {} bytes -> {} samples", opusLength, decodedSamples);
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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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speaker->decibelLevel = getAudioLoudness(decodeBuffer, decodedSamples);
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{
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MutexLocker lock(speaker->audioStream->mutex);
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auto& samples = speaker->audioStream->samples;
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samples.insert(samples.end(), decodeBuffer, decodeBuffer + decodedSamples);
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}
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playSpeaker(speaker, channels);
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}
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return true;
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@ -489,7 +447,7 @@ void Voice::closeDevice() {
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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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if (speaker->playing || speaker->audioStream->samples.size() < minSamples)
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return false;
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speaker->playing = true;
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@ -542,7 +500,7 @@ void Voice::thread() {
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encoded = ByteArray(VOICE_MAX_PACKET_SIZE, 0);
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}
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Logger::info("Voice: encoded Opus chunk {} bytes big", encodedSize);
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//Logger::info("Voice: encoded Opus chunk {} samples -> {} bytes", frameSamples, encodedSize);
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}
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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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