mirror of https://github.com/bjornbytes/lovr.git
Use fixed-size audio callbacks;
WebAudio may need...attention.
This commit is contained in:
bjorn
parent
20caeca69a
commit
a1d0b2fa11
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@ -47,9 +47,6 @@ static struct {
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float position[4];
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float orientation[4];
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Spatializer* spatializer;
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uint32_t leftoverOffset;
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uint32_t leftoverFrames;
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float leftovers[BUFFER_SIZE * 2];
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float absorption[3];
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ma_data_converter playbackConverter;
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uint32_t sampleRate;
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@ -71,138 +68,103 @@ static float linearToDb(float linear) {
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// Device callbacks
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static void onPlayback(ma_device* device, void* out, const void* in, uint32_t count) {
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lovrAssert(count == BUFFER_SIZE, "Unreachable");
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float raw[BUFFER_SIZE * 2];
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float aux[BUFFER_SIZE * 2];
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float mix[BUFFER_SIZE * 2];
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uint32_t total = count;
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float* output = out;
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// Consume any leftovers from the previous callback
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if (state.leftoverFrames > 0) {
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uint32_t leftoverFrames = MIN(count, state.leftoverFrames);
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memcpy(output, state.leftovers + state.leftoverOffset * OUTPUT_CHANNELS, leftoverFrames * OUTPUT_CHANNELS * sizeof(float));
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state.leftoverOffset += leftoverFrames;
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state.leftoverFrames -= leftoverFrames;
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output += leftoverFrames * OUTPUT_CHANNELS;
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count -= leftoverFrames;
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if (count == 0) {
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goto sink;
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}
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}
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float* dst = out;
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float* buf = NULL; // The "current" buffer (used for fast paths)
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ma_mutex_lock(&state.lock);
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do {
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float* dst = count >= BUFFER_SIZE ? output : state.leftovers;
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float* buf = NULL; // The "current" buffer (used for fast paths)
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if (dst == state.leftovers) {
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memset(dst, 0, sizeof(state.leftovers));
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Source* source;
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FOREACH_SOURCE(source) {
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if (!source->playing) {
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state.sources[source->index] = NULL;
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state.sourceMask &= ~(1ull << source->index);
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state.spatializer->sourceDestroy(source);
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source->index = ~0u;
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lovrRelease(source, lovrSourceDestroy);
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continue;
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}
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Source* source;
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FOREACH_SOURCE(source) {
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if (!source->playing) {
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state.sources[source->index] = NULL;
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state.sourceMask &= ~(1ull << source->index);
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state.spatializer->sourceDestroy(source);
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source->index = ~0u;
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lovrRelease(source, lovrSourceDestroy);
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continue;
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// Read and convert raw frames until there's BUFFER_SIZE converted frames
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// - No converter: just read frames into raw (it has enough space for BUFFER_SIZE frames).
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// - Converter: keep reading as many frames as possible/needed into raw and convert into aux.
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// - If EOF is reached, rewind and continue for looping sources, otherwise pad end with zero.
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buf = source->converter ? aux : raw;
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float* cursor = buf; // Edge of processed frames
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uint32_t channelsOut = lovrSourceUsesSpatializer(source) ? 1 : 2; // If spatializer isn't converting to stereo, converter must do it
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uint32_t framesRemaining = BUFFER_SIZE;
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while (framesRemaining > 0) {
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uint32_t framesRead;
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if (source->converter) {
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uint32_t channelsIn = lovrSoundGetChannelCount(source->sound);
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uint32_t capacity = sizeof(raw) / (channelsIn * sizeof(float));
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ma_uint64 chunk;
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ma_data_converter_get_required_input_frame_count(source->converter, framesRemaining, &chunk);
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framesRead = lovrSoundRead(source->sound, source->offset, MIN(chunk, capacity), raw);
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} else {
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framesRead = lovrSoundRead(source->sound, source->offset, framesRemaining, cursor);
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}
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// Read and convert raw frames until there's BUFFER_SIZE converted frames
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// - No converter: just read frames into raw (it has enough space for BUFFER_SIZE frames).
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// - Converter: keep reading as many frames as possible/needed into raw and convert into aux.
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// - If EOF is reached, rewind and continue for looping sources, otherwise pad end with zero.
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buf = source->converter ? aux : raw;
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float* cursor = buf; // Edge of processed frames
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uint32_t channelsOut = lovrSourceUsesSpatializer(source) ? 1 : 2; // If spatializer isn't converting to stereo, converter must do it
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uint32_t framesRemaining = BUFFER_SIZE;
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while (framesRemaining > 0) {
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uint32_t framesRead;
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if (source->converter) {
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uint32_t channelsIn = lovrSoundGetChannelCount(source->sound);
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uint32_t capacity = sizeof(raw) / (channelsIn * sizeof(float));
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ma_uint64 chunk;
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ma_data_converter_get_required_input_frame_count(source->converter, framesRemaining, &chunk);
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framesRead = lovrSoundRead(source->sound, source->offset, MIN(chunk, capacity), raw);
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if (framesRead == 0) {
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if (source->looping) {
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source->offset = 0;
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continue;
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} else {
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framesRead = lovrSoundRead(source->sound, source->offset, framesRemaining, cursor);
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}
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if (framesRead == 0) {
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if (source->looping) {
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source->offset = 0;
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continue;
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} else {
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source->offset = 0;
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source->playing = false;
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memset(cursor, 0, framesRemaining * channelsOut * sizeof(float));
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break;
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}
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} else {
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source->offset += framesRead;
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}
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if (source->converter) {
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ma_uint64 framesIn = framesRead;
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ma_uint64 framesOut = framesRemaining;
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ma_data_converter_process_pcm_frames(source->converter, raw, &framesIn, cursor, &framesOut);
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cursor += framesOut * channelsOut;
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framesRemaining -= framesOut;
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} else {
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cursor += framesRead * channelsOut;
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framesRemaining -= framesRead;
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source->offset = 0;
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source->playing = false;
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memset(cursor, 0, framesRemaining * channelsOut * sizeof(float));
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break;
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}
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} else {
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source->offset += framesRead;
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}
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// Spatialize
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if (lovrSourceUsesSpatializer(source)) {
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state.spatializer->apply(source, buf, mix, BUFFER_SIZE, BUFFER_SIZE);
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buf = mix;
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}
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// Mix
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float volume = source->volume;
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for (uint32_t i = 0; i < OUTPUT_CHANNELS * BUFFER_SIZE; i++) {
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dst[i] += buf[i] * volume;
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if (source->converter) {
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ma_uint64 framesIn = framesRead;
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ma_uint64 framesOut = framesRemaining;
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ma_data_converter_process_pcm_frames(source->converter, raw, &framesIn, cursor, &framesOut);
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cursor += framesOut * channelsOut;
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framesRemaining -= framesOut;
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} else {
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cursor += framesRead * channelsOut;
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framesRemaining -= framesRead;
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}
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}
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// Tail
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uint32_t tailCount = state.spatializer->tail(aux, mix, BUFFER_SIZE);
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for (uint32_t i = 0; i < tailCount * OUTPUT_CHANNELS; i++) {
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dst[i] += mix[i];
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// Spatialize
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if (lovrSourceUsesSpatializer(source)) {
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state.spatializer->apply(source, buf, mix, BUFFER_SIZE, BUFFER_SIZE);
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buf = mix;
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}
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// Copy some leftovers to output
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if (dst == state.leftovers) {
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memcpy(output, state.leftovers, count * OUTPUT_CHANNELS * sizeof(float));
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state.leftoverFrames = BUFFER_SIZE - count;
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state.leftoverOffset = count;
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// Mix
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float volume = source->volume;
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for (uint32_t i = 0; i < OUTPUT_CHANNELS * BUFFER_SIZE; i++) {
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dst[i] += buf[i] * volume;
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}
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}
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output += BUFFER_SIZE * OUTPUT_CHANNELS;
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count -= MIN(count, BUFFER_SIZE);
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} while (count > 0);
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// Tail
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uint32_t tailCount = state.spatializer->tail(aux, mix, BUFFER_SIZE);
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for (uint32_t i = 0; i < tailCount * OUTPUT_CHANNELS; i++) {
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dst[i] += mix[i];
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}
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ma_mutex_unlock(&state.lock);
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sink:
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if (state.sinks[AUDIO_PLAYBACK]) {
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uint64_t remaining = total;
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uint64_t capacity = sizeof(aux) / lovrSoundGetChannelCount(state.sinks[AUDIO_PLAYBACK]) / sizeof(float);
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output = out;
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while (remaining > 0) {
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ma_uint64 framesConsumed = remaining;
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while (count > 0) {
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ma_uint64 framesConsumed = count;
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ma_uint64 framesWritten = capacity;
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ma_data_converter_process_pcm_frames(&state.playbackConverter, output, &framesConsumed, aux, &framesWritten);
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ma_data_converter_process_pcm_frames(&state.playbackConverter, dst, &framesConsumed, aux, &framesWritten);
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lovrSoundWrite(state.sinks[AUDIO_PLAYBACK], 0, framesWritten, aux);
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output += framesConsumed * OUTPUT_CHANNELS;
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remaining -= framesConsumed;
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dst += framesConsumed * OUTPUT_CHANNELS;
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count -= framesConsumed;
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}
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}
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}
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@ -350,9 +312,7 @@ bool lovrAudioSetDevice(AudioType type, void* id, size_t size, Sound* sink, Audi
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config.sampleRate = lovrSoundGetSampleRate(sink);
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}
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#ifndef EMSCRIPTEN // Web needs to use the default bigger buffer size to prevent stutters
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config.periodSizeInFrames = BUFFER_SIZE;
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#endif
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config.dataCallback = callbacks[type];
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ma_result result = ma_device_init(&state.context, &config, &state.devices[type]);
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