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Formating review, using imperative mode in comments

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Ray
2026-03-03 22:40:34 +01:00
parent bf830c3f7b
commit b4746469d4
18 changed files with 164 additions and 161 deletions
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33
src/raudio.c
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@ -495,8 +495,9 @@ void InitAudioDevice(void)
return;
}
// Mixing happens on a separate thread which means we need to synchronize. I'm using a mutex here to make things simple, but may
// want to look at something a bit smarter later on to keep everything real-time, if that's necessary
// Mixing happens on a separate thread which means synchronization is needed
// A mutex is used here to make things simple, but may want to look at something
// a bit smarter later on to keep everything real-time, if that's necessary
if (ma_mutex_init(&AUDIO.System.lock) != MA_SUCCESS)
{
TRACELOG(LOG_WARNING, "AUDIO: Failed to create mutex for mixing");
@ -822,7 +823,7 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
wave.channels = wav.channels;
wave.data = (short *)RL_MALLOC((size_t)wave.frameCount*wave.channels*sizeof(short));
// NOTE: We are forcing conversion to 16bit sample size on reading
// NOTE: Forcing conversion to 16bit sample size on reading
drwav_read_pcm_frames_s16(&wav, wave.frameCount, (drwav_int16 *)wave.data);
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load WAV data");
@ -845,7 +846,7 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
wave.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggData); // NOTE: It returns frames!
wave.data = (short *)RL_MALLOC(wave.frameCount*wave.channels*sizeof(short));
// NOTE: Get the number of samples to process (be careful! we ask for number of shorts, not bytes!)
// NOTE: Get the number of samples to process (be careful! asking for number of shorts, not bytes!)
stb_vorbis_get_samples_short_interleaved(oggData, info.channels, (short *)wave.data, wave.frameCount*wave.channels);
stb_vorbis_close(oggData);
}
@ -858,7 +859,7 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
drmp3_config config = { 0 };
unsigned long long int totalFrameCount = 0;
// NOTE: We are forcing conversion to 32bit float sample size on reading
// NOTE: Forcing conversion to 32bit float sample size on reading
wave.data = drmp3_open_memory_and_read_pcm_frames_f32(fileData, dataSize, &config, &totalFrameCount, NULL);
wave.sampleSize = 32;
@ -896,7 +897,7 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
{
unsigned long long int totalFrameCount = 0;
// NOTE: We are forcing conversion to 16bit sample size on reading
// NOTE: Forcing conversion to 16bit sample size on reading
wave.data = drflac_open_memory_and_read_pcm_frames_s16(fileData, dataSize, &wave.channels, &wave.sampleRate, &totalFrameCount, NULL);
wave.sampleSize = 16;
@ -933,7 +934,7 @@ Sound LoadSound(const char *fileName)
Sound sound = LoadSoundFromWave(wave);
UnloadWave(wave); // Sound is loaded, we can unload wave
UnloadWave(wave); // Sound is loaded, wave can be unloaded
return sound;
}
@ -2475,20 +2476,20 @@ static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer,
// Reads audio data from an AudioBuffer object in device format, returned data will be in a format appropriate for mixing
static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, float *framesOut, ma_uint32 frameCount)
{
// NOTE: Continuously converting data from the AudioBuffer's internal format to the mixing format,
// which should be defined by the output format of the data converter.
// This is done until frameCount frames have been output.
// NOTE: Continuously converting data from the AudioBuffer's internal format to the mixing format,
// which should be defined by the output format of the data converter.
// This is done until frameCount frames have been output.
ma_uint32 bpf = ma_get_bytes_per_frame(audioBuffer->converter.formatIn, audioBuffer->converter.channelsIn);
ma_uint8 inputBuffer[4096] = { 0 };
ma_uint32 inputBufferFrameCap = sizeof(inputBuffer)/bpf;
ma_uint32 totalOutputFramesProcessed = 0;
while (totalOutputFramesProcessed < frameCount)
{
float *runningFramesOut = framesOut + (totalOutputFramesProcessed*audioBuffer->converter.channelsOut);
ma_uint64 outputFramesToProcessThisIteration = frameCount - totalOutputFramesProcessed;
//ma_uint64 inputFramesToProcessThisIteration = 0;
// Process any residual input frames from the previous read first.
if (audioBuffer->converterResidualCount > 0)
{
@ -2656,7 +2657,7 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 fr
const float localVolume = buffer->volume;
const ma_uint32 channels = AUDIO.System.device.playback.channels;
if (channels == 2) // We consider panning
if (channels == 2) // Consider panning
{
const float right = (buffer->pan + 1.0f)/2.0f; // Normalize: [-1..1] -> [0..1]
const float left = 1.0f - right;
@ -2676,7 +2677,7 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 fr
frameIn += 2;
}
}
else // We do not consider panning
else // Do not consider panning
{
for (ma_uint32 frame = 0; frame < frameCount; frame++)
{
@ -2831,7 +2832,7 @@ static const char *GetFileNameWithoutExt(const char *filePath)
{
if (fileName[i] == '.')
{
// NOTE: We break on first '.' found
// NOTE: Break on first '.' found
fileName[i] = '\0';
break;
}
@ -2862,7 +2863,7 @@ static unsigned char *LoadFileData(const char *fileName, int *dataSize)
{
data = (unsigned char *)RL_MALLOC(size*sizeof(unsigned char));
// NOTE: fread() returns number of read elements instead of bytes, so we read [1 byte, size elements]
// NOTE: fread() returns number of read elements instead of bytes, so reading [1 byte, size elements]
unsigned int count = (unsigned int)fread(data, sizeof(unsigned char), size, file);
*dataSize = count;