Fix a few bugs and style fixes

* We can't realloc converters, that'll break internal pointers
* inverted condition in an assert
* less magic numbers
* can't loop streams

src/modules/audio/audio.c

@@ -62,7 +62,6 @@ static bool mix(Source* source, float* output, uint32_t count) {
         ma_data_converter_get_required_input_frame_count(source->converter, count));
         // ^^^ Note need to min `count` with 'capacity of aux buffer' and 'capacity of mix buffer'
         // could skip min-ing with one of the buffers if you can guarantee that one is bigger/equal to the other (you can because their formats are known)
-
     ma_uint64 framesIn = source->sound->read(source->sound, source->offset, chunk, raw);
     ma_uint64 framesOut = sizeof(aux) / (sizeof(float) * outputChannelCountForSource(source));
 
@@ -71,7 +70,7 @@ static bool mix(Source* source, float* output, uint32_t count) {
     if (source->spatial) {
       state.spatializer->apply(source, source->transform, aux, mix, framesOut);
     } else {
-      memcpy(mix, aux, framesOut * OUTPUT_CHANNELS * sizeof(float));
+      memcpy(mix, aux, framesOut * bytesPerAudioFrame(OUTPUT_CHANNELS, SAMPLE_F32));
     }
 
     for (uint32_t i = 0; i < framesOut * OUTPUT_CHANNELS; i++) {
@@ -180,6 +179,7 @@ bool lovrAudioInit(AudioConfig config[2]) {
   lovrAssert(state.spatializer != NULL, "Must have at least one spatializer");
 
   arr_init(&state.converters);
+  arr_reserve(&state.converters, 16);
 
   return state.initialized = true;
 }
@@ -287,7 +287,8 @@ static void _lovrSourceAssignConverter(Source *source) {
     config.channelsOut = outputChannelCountForSource(source);
     config.sampleRateIn = source->sound->sampleRate;
     config.sampleRateOut = LOVR_AUDIO_SAMPLE_RATE;
-    arr_expand(&state.converters, 1);
+    // can't use arr_expand because that will destroy the internal pointers inside the converter
+    lovrAssert(state.converters.length+1 < state.converters.capacity, "Out of space for converters");
     ma_data_converter* converter = &state.converters.data[state.converters.length++];
     lovrAssert(!ma_data_converter_init(&config, converter), "Problem creating Source data converter");
     source->converter = converter;
@@ -344,6 +345,7 @@ bool lovrSourceIsLooping(Source* source) {
 }
 
 void lovrSourceSetLooping(Source* source, bool loop) {
+  lovrAssert(loop == false || lovrSoundDataIsStream(source->sound) == false, "Can't loop streams");
   source->looping = loop;
 }
 

src/modules/data/soundData.c

@@ -60,10 +60,9 @@ static uint32_t lovrSoundDataReadRing(SoundData* soundData, uint32_t offset, uin
     lovrAssert(acquire_status == MA_SUCCESS, "Failed to acquire ring buffer for read: %d\n", acquire_status);
     memcpy(data, store, availableFramesInRing * bytesPerFrame);
     ma_result commit_status = ma_pcm_rb_commit_read(soundData->ring, availableFramesInRing, store);
-    lovrAssert(commit_status != MA_SUCCESS, "Failed to commit ring buffer for read: %d\n", acquire_status);
+    lovrAssert(commit_status == MA_SUCCESS, "Failed to commit ring buffer for read: %d\n", acquire_status);
 
-    if (availableFramesInRing == 0 && count > 0) {
-      memset(data, 0, count * bytesPerFrame);
+    if (availableFramesInRing == 0) {
       return totalRead;
     }
 
@@ -74,6 +73,7 @@ static uint32_t lovrSoundDataReadRing(SoundData* soundData, uint32_t offset, uin
   return totalRead;
 }
 
+
 SoundData* lovrSoundDataCreateRaw(uint32_t frameCount, uint32_t channelCount, uint32_t sampleRate, SampleFormat format, struct Blob* blob) {
   SoundData* soundData = lovrAlloc(SoundData);
   soundData->format = format;
@@ -95,8 +95,7 @@ SoundData* lovrSoundDataCreateRaw(uint32_t frameCount, uint32_t channelCount, ui
   return soundData;
 }
 
-SoundData* lovrSoundDataCreateStream(uint32_t bufferSizeInFrames, uint32_t channels, uint32_t sampleRate, SampleFormat format)
-{
+SoundData* lovrSoundDataCreateStream(uint32_t bufferSizeInFrames, uint32_t channels, uint32_t sampleRate, SampleFormat format) {
   SoundData* soundData = lovrAlloc(SoundData);
   soundData->format = format;
   soundData->sampleRate = sampleRate;
@@ -176,12 +175,14 @@ size_t lovrSoundDataStreamAppendSound(SoundData *dest, SoundData *src) {
   return lovrSoundDataStreamAppendBlob(dest, src->blob);
 }
 
+bool lovrSoundDataIsStream(SoundData *soundData) {
+  return soundData->read == lovrSoundDataReadRing;
+}
+
 void lovrSoundDataDestroy(void* ref) {
   SoundData* soundData = (SoundData*) ref;
   stb_vorbis_close(soundData->decoder);
   lovrRelease(Blob, soundData->blob);
-  if (soundData->ring) {
-    ma_pcm_rb_uninit(soundData->ring);
-    free(soundData->ring);
-  }
+  ma_pcm_rb_uninit(soundData->ring);
+  free(soundData->ring);
 }

src/modules/data/soundData.h

@@ -34,4 +34,6 @@ SoundData* lovrSoundDataCreateFromFile(struct Blob* blob, bool decode);
 // returns the number of frames successfully appended (if it's less than the size of blob, the internal ring buffer is full)
 size_t lovrSoundDataStreamAppendBlob(SoundData *dest, struct Blob* blob);
 size_t lovrSoundDataStreamAppendSound(SoundData *dest, SoundData *src);
+
+bool lovrSoundDataIsStream(SoundData *soundData);
 void lovrSoundDataDestroy(void* ref);