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-rw-r--r--src/raudio.c38
1 files changed, 19 insertions, 19 deletions
diff --git a/src/raudio.c b/src/raudio.c
index 636d15b8..9108a903 100644
--- a/src/raudio.c
+++ b/src/raudio.c
@@ -567,7 +567,7 @@ void SetMasterVolume(float volume)
// Create a new audio buffer. Initially filled with silence
AudioBuffer *CreateAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 bufferSizeInFrames, AudioBufferUsage usage)
{
- AudioBuffer *audioBuffer = (AudioBuffer *)calloc(sizeof(*audioBuffer) + (bufferSizeInFrames*channels*ma_get_bytes_per_sample(format)), 1);
+ AudioBuffer *audioBuffer = (AudioBuffer *)RL_CALLOC(sizeof(*audioBuffer) + (bufferSizeInFrames*channels*ma_get_bytes_per_sample(format)), 1);
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "CreateAudioBuffer() : Failed to allocate memory for audio buffer");
@@ -591,7 +591,7 @@ AudioBuffer *CreateAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 s
if (result != MA_SUCCESS)
{
TraceLog(LOG_ERROR, "CreateAudioBuffer() : Failed to create data conversion pipeline");
- free(audioBuffer);
+ RL_FREE(audioBuffer);
return NULL;
}
@@ -623,7 +623,7 @@ void DeleteAudioBuffer(AudioBuffer *audioBuffer)
}
UntrackAudioBuffer(audioBuffer);
- free(audioBuffer);
+ RL_FREE(audioBuffer);
}
// Check if an audio buffer is playing
@@ -863,7 +863,7 @@ Sound LoadSoundFromWave(Wave wave)
// Unload wave data
void UnloadWave(Wave wave)
{
- if (wave.data != NULL) free(wave.data);
+ if (wave.data != NULL) RL_FREE(wave.data);
TraceLog(LOG_INFO, "Unloaded wave data from RAM");
}
@@ -1017,7 +1017,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
return;
}
- void *data = malloc(frameCount*channels*(sampleSize/8));
+ void *data = RL_MALLOC(frameCount*channels*(sampleSize/8));
frameCount = (ma_uint32)ma_convert_frames(data, formatOut, channels, sampleRate, wave->data, formatIn, wave->channels, wave->sampleRate, frameCountIn);
if (frameCount == 0)
@@ -1030,7 +1030,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
wave->sampleSize = sampleSize;
wave->sampleRate = sampleRate;
wave->channels = channels;
- free(wave->data);
+ RL_FREE(wave->data);
wave->data = data;
}
@@ -1039,7 +1039,7 @@ Wave WaveCopy(Wave wave)
{
Wave newWave = { 0 };
- newWave.data = malloc(wave.sampleCount*wave.sampleSize/8*wave.channels);
+ newWave.data = RL_MALLOC(wave.sampleCount*wave.sampleSize/8*wave.channels);
if (newWave.data != NULL)
{
@@ -1064,11 +1064,11 @@ void WaveCrop(Wave *wave, int initSample, int finalSample)
{
int sampleCount = finalSample - initSample;
- void *data = malloc(sampleCount*wave->sampleSize/8*wave->channels);
+ void *data = RL_MALLOC(sampleCount*wave->sampleSize/8*wave->channels);
memcpy(data, (unsigned char *)wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->channels*wave->sampleSize/8);
- free(wave->data);
+ RL_FREE(wave->data);
wave->data = data;
}
else TraceLog(LOG_WARNING, "Wave crop range out of bounds");
@@ -1078,7 +1078,7 @@ void WaveCrop(Wave *wave, int initSample, int finalSample)
// NOTE: Returned sample values are normalized to range [-1..1]
float *GetWaveData(Wave wave)
{
- float *samples = (float *)malloc(wave.sampleCount*wave.channels*sizeof(float));
+ float *samples = (float *)RL_MALLOC(wave.sampleCount*wave.channels*sizeof(float));
for (unsigned int i = 0; i < wave.sampleCount; i++)
{
@@ -1100,7 +1100,7 @@ float *GetWaveData(Wave wave)
// Load music stream from file
Music LoadMusicStream(const char *fileName)
{
- Music music = (MusicData *)malloc(sizeof(MusicData));
+ Music music = (MusicData *)RL_MALLOC(sizeof(MusicData));
bool musicLoaded = true;
#if defined(SUPPORT_FILEFORMAT_OGG)
@@ -1228,7 +1228,7 @@ Music LoadMusicStream(const char *fileName)
if (false) {}
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
- else if (music->ctxType == MUSIC_AUDIO_FLAC) drflac_free(music->ctxFlac);
+ else if (music->ctxType == MUSIC_AUDIO_FLAC) drflac_RL_FREE(music->ctxFlac);
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
else if (music->ctxType == MUSIC_AUDIO_MP3) drmp3_uninit(&music->ctxMp3);
@@ -1240,7 +1240,7 @@ Music LoadMusicStream(const char *fileName)
else if (music->ctxType == MUSIC_MODULE_MOD) jar_mod_unload(&music->ctxMod);
#endif
- free(music);
+ RL_FREE(music);
music = NULL;
TraceLog(LOG_WARNING, "[%s] Music file could not be opened", fileName);
@@ -1262,7 +1262,7 @@ void UnloadMusicStream(Music music)
if (false) {}
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
- else if (music->ctxType == MUSIC_AUDIO_FLAC) drflac_free(music->ctxFlac);
+ else if (music->ctxType == MUSIC_AUDIO_FLAC) drflac_RL_FREE(music->ctxFlac);
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
else if (music->ctxType == MUSIC_AUDIO_MP3) drmp3_uninit(&music->ctxMp3);
@@ -1274,7 +1274,7 @@ void UnloadMusicStream(Music music)
else if (music->ctxType == MUSIC_MODULE_MOD) jar_mod_unload(&music->ctxMod);
#endif
- free(music);
+ RL_FREE(music);
}
// Start music playing (open stream)
@@ -1357,7 +1357,7 @@ void UpdateMusicStream(Music music)
unsigned int subBufferSizeInFrames = ((AudioBuffer *)music->stream.audioBuffer)->bufferSizeInFrames/2;
// NOTE: Using dynamic allocation because it could require more than 16KB
- void *pcm = calloc(subBufferSizeInFrames*music->stream.channels*music->stream.sampleSize/8, 1);
+ void *pcm = RL_CALLOC(subBufferSizeInFrames*music->stream.channels*music->stream.sampleSize/8, 1);
int samplesCount = 0; // Total size of data steamed in L+R samples for xm floats, individual L or R for ogg shorts
@@ -1427,7 +1427,7 @@ void UpdateMusicStream(Music music)
}
// Free allocated pcm data
- free(pcm);
+ RL_FREE(pcm);
// Reset audio stream for looping
if (streamEnding)
@@ -1750,7 +1750,7 @@ static Wave LoadWAV(const char *fileName)
else
{
// Allocate memory for data
- wave.data = malloc(wavData.subChunkSize);
+ wave.data = RL_MALLOC(wavData.subChunkSize);
// Read in the sound data into the soundData variable
fread(wave.data, wavData.subChunkSize, 1, wavFile);
@@ -1891,7 +1891,7 @@ static Wave LoadOGG(const char *fileName)
float totalSeconds = stb_vorbis_stream_length_in_seconds(oggFile);
if (totalSeconds > 10) TraceLog(LOG_WARNING, "[%s] Ogg audio length is larger than 10 seconds (%f), that's a big file in memory, consider music streaming", fileName, totalSeconds);
- wave.data = (short *)malloc(wave.sampleCount*wave.channels*sizeof(short));
+ wave.data = (short *)RL_MALLOC(wave.sampleCount*wave.channels*sizeof(short));
// NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(oggFile, info.channels, (short *)wave.data, wave.sampleCount*wave.channels);
generated by cgit v1.2.3 (git 2.39.1) at 2026-04-17 18:37:32 +0000