diff --git a/src/raudio.c b/src/raudio.c
index 2d1c29a93..a67706053 100644
--- a/src/raudio.c
+++ b/src/raudio.c
@@ -383,7 +383,7 @@ void InitAudioDevice(void)
ma_result result = ma_context_init(NULL, 0, &ctxConfig, &AUDIO.System.context);
if (result != MA_SUCCESS)
{
- TRACELOG(LOG_ERROR, "Failed to initialize audio context");
+ TRACELOG(LOG_ERROR, "AUDIO: Failed to initialize context");
return;
}
@@ -403,7 +403,7 @@ void InitAudioDevice(void)
result = ma_device_init(&AUDIO.System.context, &config, &AUDIO.System.device);
if (result != MA_SUCCESS)
{
- TRACELOG(LOG_ERROR, "Failed to initialize audio playback AUDIO.System.device");
+ TRACELOG(LOG_ERROR, "AUDIO: Failed to initialize playback device");
ma_context_uninit(&AUDIO.System.context);
return;
}
@@ -413,7 +413,7 @@ void InitAudioDevice(void)
result = ma_device_start(&AUDIO.System.device);
if (result != MA_SUCCESS)
{
- TRACELOG(LOG_ERROR, "Failed to start audio playback AUDIO.System.device");
+ TRACELOG(LOG_ERROR, "AUDIO: Failed to start playback device");
ma_device_uninit(&AUDIO.System.device);
ma_context_uninit(&AUDIO.System.context);
return;
@@ -423,21 +423,20 @@ void InitAudioDevice(void)
// 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.context, &AUDIO.System.lock) != MA_SUCCESS)
{
- TRACELOG(LOG_ERROR, "Failed to create mutex for audio mixing");
+ TRACELOG(LOG_ERROR, "AUDIO: Failed to create mutex for mixing");
ma_device_uninit(&AUDIO.System.device);
ma_context_uninit(&AUDIO.System.context);
return;
}
- TRACELOG(LOG_INFO, "Audio device initialized successfully");
- TRACELOG(LOG_INFO, "Audio backend: miniaudio / %s", ma_get_backend_name(AUDIO.System.context.backend));
- TRACELOG(LOG_INFO, "Audio format: %s -> %s", ma_get_format_name(AUDIO.System.device.playback.format), ma_get_format_name(AUDIO.System.device.playback.internalFormat));
- TRACELOG(LOG_INFO, "Audio channels: %d -> %d", AUDIO.System.device.playback.channels, AUDIO.System.device.playback.internalChannels);
- TRACELOG(LOG_INFO, "Audio sample rate: %d -> %d", AUDIO.System.device.sampleRate, AUDIO.System.device.playback.internalSampleRate);
- TRACELOG(LOG_INFO, "Audio periods size: %d", AUDIO.System.device.playback.internalPeriodSizeInFrames*AUDIO.System.device.playback.internalPeriods);
+ TRACELOG(LOG_INFO, "AUDIO: Device initialized successfully");
+ TRACELOG(LOG_INFO, " > Backend: miniaudio / %s", ma_get_backend_name(AUDIO.System.context.backend));
+ TRACELOG(LOG_INFO, " > Format: %s -> %s", ma_get_format_name(AUDIO.System.device.playback.format), ma_get_format_name(AUDIO.System.device.playback.internalFormat));
+ TRACELOG(LOG_INFO, " > Channels: %d -> %d", AUDIO.System.device.playback.channels, AUDIO.System.device.playback.internalChannels);
+ TRACELOG(LOG_INFO, " > Sample rate: %d -> %d", AUDIO.System.device.sampleRate, AUDIO.System.device.playback.internalSampleRate);
+ TRACELOG(LOG_INFO, " > Periods size: %d", AUDIO.System.device.playback.internalPeriodSizeInFrames*AUDIO.System.device.playback.internalPeriods);
InitAudioBufferPool();
- TRACELOG(LOG_INFO, "Audio multichannel pool size: %i", MAX_AUDIO_BUFFER_POOL_CHANNELS);
AUDIO.System.isReady = true;
}
@@ -453,9 +452,9 @@ void CloseAudioDevice(void)
CloseAudioBufferPool();
- TRACELOG(LOG_INFO, "Audio device closed successfully");
+ TRACELOG(LOG_INFO, "AUDIO: Device closed successfully");
}
- else TRACELOG(LOG_WARNING, "Could not close audio device because it is not currently initialized");
+ else TRACELOG(LOG_WARNING, "AUDIO: Device could not be closed, not currently initialized");
}
// Check if device has been initialized successfully
@@ -481,7 +480,7 @@ AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam
if (audioBuffer == NULL)
{
- TRACELOG(LOG_ERROR, "LoadAudioBuffer() : Failed to allocate memory for audio buffer");
+ TRACELOG(LOG_ERROR, "AUDIO: Failed to allocate memory for buffer");
return NULL;
}
@@ -495,7 +494,7 @@ AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam
if (result != MA_SUCCESS)
{
- TRACELOG(LOG_ERROR, "LoadAudioBuffer() : Failed to create data conversion pipeline");
+ TRACELOG(LOG_ERROR, "AUDIO: Failed to create data conversion pipeline");
RL_FREE(audioBuffer);
return NULL;
}
@@ -665,7 +664,7 @@ Wave LoadWave(const char *fileName)
#if defined(SUPPORT_FILEFORMAT_MP3)
else if (IsFileExtension(fileName, ".mp3")) wave = LoadMP3(fileName);
#endif
- else TRACELOG(LOG_WARNING, "[%s] Audio fileformat not supported, it can't be loaded", fileName);
+ else TRACELOG(LOG_WARNING, "FILEIO: [%s] File format not supported", fileName);
return wave;
}
@@ -704,13 +703,13 @@ Sound LoadSoundFromWave(Wave wave)
ma_uint32 frameCountIn = wave.sampleCount/wave.channels;
ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, NULL, frameCountIn, formatIn, wave.channels, wave.sampleRate);
- if (frameCount == 0) TRACELOG(LOG_WARNING, "LoadSoundFromWave() : Failed to get frame count for format conversion");
+ if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed to get frame count for format conversion");
AudioBuffer *audioBuffer = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC);
- if (audioBuffer == NULL) TRACELOG(LOG_WARNING, "LoadSoundFromWave() : Failed to create audio buffer");
+ if (audioBuffer == NULL) TRACELOG(LOG_WARNING, "SOUND: Failed to create buffer");
frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, frameCount, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, wave.data, frameCountIn, formatIn, wave.channels, wave.sampleRate);
- if (frameCount == 0) TRACELOG(LOG_WARNING, "LoadSoundFromWave() : Format conversion failed");
+ if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed format conversion");
sound.sampleCount = frameCount*AUDIO_DEVICE_CHANNELS;
sound.stream.sampleRate = AUDIO_DEVICE_SAMPLE_RATE;
@@ -727,7 +726,7 @@ void UnloadWave(Wave wave)
{
if (wave.data != NULL) RL_FREE(wave.data);
- TRACELOG(LOG_INFO, "Unloaded wave data from RAM");
+ TRACELOG(LOG_INFO, "WAVE: Unloaded wave data from RAM");
}
// Unload sound
@@ -735,7 +734,7 @@ void UnloadSound(Sound sound)
{
UnloadAudioBuffer(sound.stream.buffer);
- TRACELOG(LOG_INFO, "Unloaded sound data from RAM");
+ TRACELOG(LOG_INFO, "WAVE: Unloaded sound data from RAM");
}
// Update sound buffer with new data
@@ -767,8 +766,8 @@ void ExportWave(Wave wave, const char *fileName)
success = true;
}
- if (success) TRACELOG(LOG_INFO, "Wave exported successfully: %s", fileName);
- else TRACELOG(LOG_WARNING, "Wave could not be exported.");
+ if (success) TRACELOG(LOG_INFO, "FILEIO: [%s] Wave data exported successfully", fileName);
+ else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to export wave data", fileName);
}
// Export wave sample data to code (.h)
@@ -849,12 +848,12 @@ void PlaySoundMulti(Sound sound)
// If no none playing pool members can be index choose the oldest
if (index == -1)
{
- TRACELOG(LOG_WARNING, "Pool age %i ended a sound early no room in buffer pool", AUDIO.MultiChannel.poolCounter);
+ TRACELOG(LOG_WARNING, "SOUND: Buffer pool is already full, count: %i", AUDIO.MultiChannel.poolCounter);
if (oldIndex == -1)
{
// Shouldn't be able to get here... but just in case something odd happens!
- TRACELOG(LOG_ERROR, "Sound buffer pool could not determine oldest buffer not playing sound");
+ TRACELOG(LOG_WARNING, "SOUND: Buffer pool could not determine oldest buffer not playing sound");
return;
}
@@ -949,7 +948,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, formatOut, channels, sampleRate, NULL, frameCountIn, formatIn, wave->channels, wave->sampleRate);
if (frameCount == 0)
{
- TRACELOG(LOG_WARNING, "WaveFormat() : Failed to get frame count for format conversion.");
+ TRACELOG(LOG_WARNING, "WAVE: Failed to get frame count for format conversion");
return;
}
@@ -958,7 +957,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
frameCount = (ma_uint32)ma_convert_frames(data, frameCount, formatOut, channels, sampleRate, wave->data, frameCountIn, formatIn, wave->channels, wave->sampleRate);
if (frameCount == 0)
{
- TRACELOG(LOG_WARNING, "WaveFormat() : Format conversion failed.");
+ TRACELOG(LOG_WARNING, "WAVE: Failed format conversion");
return;
}
@@ -1007,7 +1006,7 @@ void WaveCrop(Wave *wave, int initSample, int finalSample)
RL_FREE(wave->data);
wave->data = data;
}
- else TRACELOG(LOG_WARNING, "Wave crop range out of bounds");
+ else TRACELOG(LOG_WARNING, "WAVE: Crop range out of bounds");
}
// Get samples data from wave as a floats array
@@ -1160,16 +1159,16 @@ Music LoadMusicStream(const char *fileName)
else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); }
#endif
- TRACELOG(LOG_WARNING, "[%s] Music file could not be opened", fileName);
+ TRACELOG(LOG_WARNING, "FILEIO: [%s] Music file could not be opened", fileName);
}
else
{
// Show some music stream info
- TRACELOG(LOG_INFO, "[%s] Music file successfully loaded:", fileName);
- TRACELOG(LOG_INFO, " Total samples: %i", music.sampleCount);
- TRACELOG(LOG_INFO, " Sample rate: %i Hz", music.stream.sampleRate);
- TRACELOG(LOG_INFO, " Sample size: %i bits", music.stream.sampleSize);
- TRACELOG(LOG_INFO, " Channels: %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi");
+ TRACELOG(LOG_INFO, "FILEIO: [%s] Music file successfully loaded:", fileName);
+ TRACELOG(LOG_INFO, " > Total samples: %i", music.sampleCount);
+ TRACELOG(LOG_INFO, " > Sample rate: %i Hz", music.stream.sampleRate);
+ TRACELOG(LOG_INFO, " > Sample size: %i bits", music.stream.sampleSize);
+ TRACELOG(LOG_INFO, " > Channels: %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi");
}
return music;
@@ -1430,9 +1429,9 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un
if (stream.buffer != NULL)
{
stream.buffer->looping = true; // Always loop for streaming buffers
- TRACELOG(LOG_INFO, "Audio stream initialized successfully (%i Hz, %i bit, %s)", stream.sampleRate, stream.sampleSize, (stream.channels == 1)? "Mono" : "Stereo");
+ TRACELOG(LOG_INFO, "STREAM: Initialized successfully (%i Hz, %i bit, %s)", stream.sampleRate, stream.sampleSize, (stream.channels == 1)? "Mono" : "Stereo");
}
- else TRACELOG(LOG_ERROR, "Audio stream could not be created: Failed to load audio buffer");
+ else TRACELOG(LOG_WARNING, "STREAM: Failed to load audio buffer, stream could not be created");
return stream;
}
@@ -1442,7 +1441,7 @@ void CloseAudioStream(AudioStream stream)
{
UnloadAudioBuffer(stream.buffer);
- TRACELOG(LOG_INFO, "Unloaded audio stream data");
+ TRACELOG(LOG_INFO, "STREAM: Unloaded audio stream data from RAM");
}
// Update audio stream buffers with data
@@ -1493,9 +1492,9 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
stream.buffer->isSubBufferProcessed[subBufferToUpdate] = false;
}
- else TRACELOG(LOG_ERROR, "UpdateAudioStream() : Attempting to write too many frames to buffer");
+ else TRACELOG(LOG_WARNING, "STREAM: Attempting to write too many frames to buffer");
}
- else TRACELOG(LOG_ERROR, "UpdateAudioStream() : Audio buffer not available for updating");
+ else TRACELOG(LOG_WARNING, "STREAM: Buffer not available for updating");
}
}
@@ -1565,7 +1564,7 @@ static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel,
(void)pContext;
(void)pDevice;
- TRACELOG(LOG_ERROR, message); // All log messages from miniaudio are errors
+ TRACELOG(LOG_ERROR, "miniaudio: %s", message); // All log messages from miniaudio are errors
}
// Reads audio data from an AudioBuffer object in internal format.
@@ -1807,6 +1806,9 @@ static void InitAudioBufferPool(void)
{
AUDIO.MultiChannel.pool[i] = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, 0, AUDIO_BUFFER_USAGE_STATIC);
}
+
+ // TODO: Verification required for log
+ TRACELOG(LOG_INFO, "AUDIO: Multichannel pool size: %i", MAX_AUDIO_BUFFER_POOL_CHANNELS);
}
// Close the audio buffers pool
@@ -1857,7 +1859,7 @@ static Wave LoadWAV(const char *fileName)
if (wavFile == NULL)
{
- TRACELOG(LOG_WARNING, "[%s] WAV file could not be opened", fileName);
+ TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open WAV file", fileName);
wave.data = NULL;
}
else
@@ -1875,7 +1877,7 @@ static Wave LoadWAV(const char *fileName)
(wavRiffHeader.format[2] != 'V') ||
(wavRiffHeader.format[3] != 'E'))
{
- TRACELOG(LOG_WARNING, "[%s] Invalid RIFF or WAVE Header", fileName);
+ TRACELOG(LOG_WARNING, "WAVE: [%s] Invalid RIFF or WAVE header", fileName);
}
else
{
@@ -1886,7 +1888,7 @@ static Wave LoadWAV(const char *fileName)
if ((wavFormat.subChunkID[0] != 'f') || (wavFormat.subChunkID[1] != 'm') ||
(wavFormat.subChunkID[2] != 't') || (wavFormat.subChunkID[3] != ' '))
{
- TRACELOG(LOG_WARNING, "[%s] Invalid Wave format", fileName);
+ TRACELOG(LOG_WARNING, "WAVE: [%s] Invalid Wave format", fileName);
}
else
{
@@ -1900,7 +1902,7 @@ static Wave LoadWAV(const char *fileName)
if ((wavData.subChunkID[0] != 'd') || (wavData.subChunkID[1] != 'a') ||
(wavData.subChunkID[2] != 't') || (wavData.subChunkID[3] != 'a'))
{
- TRACELOG(LOG_WARNING, "[%s] Invalid data header", fileName);
+ TRACELOG(LOG_WARNING, "WAVE: [%s] Invalid data header", fileName);
}
else
{
@@ -1918,7 +1920,7 @@ static Wave LoadWAV(const char *fileName)
// NOTE: Only support 8 bit, 16 bit and 32 bit sample sizes
if ((wave.sampleSize != 8) && (wave.sampleSize != 16) && (wave.sampleSize != 32))
{
- TRACELOG(LOG_WARNING, "[%s] WAV sample size (%ibit) not supported, converted to 16bit", fileName, wave.sampleSize);
+ TRACELOG(LOG_WARNING, "WAVE: [%s] Sample size (%ibit) not supported, converted to 16bit", fileName, wave.sampleSize);
WaveFormat(&wave, wave.sampleRate, 16, wave.channels);
}
@@ -1926,13 +1928,13 @@ static Wave LoadWAV(const char *fileName)
if (wave.channels > 2)
{
WaveFormat(&wave, wave.sampleRate, wave.sampleSize, 2);
- TRACELOG(LOG_WARNING, "[%s] WAV channels number (%i) not supported, converted to 2 channels", fileName, wave.channels);
+ TRACELOG(LOG_WARNING, "WAVE: [%s] Channels number (%i) not supported, converted to 2 channels", fileName, wave.channels);
}
// NOTE: subChunkSize comes in bytes, we need to translate it to number of samples
wave.sampleCount = (wavData.subChunkSize/(wave.sampleSize/8))/wave.channels;
- TRACELOG(LOG_INFO, "[%s] WAV file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
+ TRACELOG(LOG_INFO, "WAVE: [%s] File loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
}
}
}
@@ -1974,7 +1976,7 @@ static int SaveWAV(Wave wave, const char *fileName)
FILE *wavFile = fopen(fileName, "wb");
- if (wavFile == NULL) TRACELOG(LOG_WARNING, "[%s] WAV audio file could not be created", fileName);
+ if (wavFile == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open audio file", fileName);
else
{
RiffHeader riffHeader;
@@ -2033,7 +2035,7 @@ static Wave LoadOGG(const char *fileName)
stb_vorbis *oggFile = stb_vorbis_open_filename(fileName, NULL, NULL);
- if (oggFile == NULL) TRACELOG(LOG_WARNING, "[%s] OGG file could not be opened", fileName);
+ if (oggFile == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open OGG file", fileName);
else
{
stb_vorbis_info info = stb_vorbis_get_info(oggFile);
@@ -2044,13 +2046,13 @@ static Wave LoadOGG(const char *fileName)
wave.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggFile)*info.channels; // Independent by channel
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);
+ if (totalSeconds > 10) TRACELOG(LOG_WARNING, "WAVE: [%s] Ogg audio length larger than 10 seconds (%f), that's a big file in memory, consider music streaming", fileName, totalSeconds);
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!)
stb_vorbis_get_samples_short_interleaved(oggFile, info.channels, (short *)wave.data, wave.sampleCount*wave.channels);
- TRACELOG(LOG_INFO, "[%s] OGG file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
+ TRACELOG(LOG_INFO, "WAVE: [%s] OGG file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
stb_vorbis_close(oggFile);
}
@@ -2064,21 +2066,24 @@ static Wave LoadOGG(const char *fileName)
// NOTE: Using dr_flac library
static Wave LoadFLAC(const char *fileName)
{
- Wave wave;
+ Wave wave = { 0 };
// Decode an entire FLAC file in one go
- unsigned long long int totalSampleCount;
+ unsigned long long int totalSampleCount = 0;
wave.data = drflac_open_file_and_read_pcm_frames_s16(fileName, &wave.channels, &wave.sampleRate, &totalSampleCount);
- wave.sampleCount = (unsigned int)totalSampleCount;
- wave.sampleSize = 16;
-
- // NOTE: Only support up to 2 channels (mono, stereo)
- if (wave.channels > 2) TRACELOG(LOG_WARNING, "[%s] FLAC channels number (%i) not supported", fileName, wave.channels);
+ if (wave.data == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to load FLAC data", fileName);
+ else
+ {
+ wave.sampleCount = (unsigned int)totalSampleCount;
+ wave.sampleSize = 16;
- if (wave.data == NULL) TRACELOG(LOG_WARNING, "[%s] FLAC data could not be loaded", fileName);
- else TRACELOG(LOG_INFO, "[%s] FLAC file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
+ // NOTE: Only support up to 2 channels (mono, stereo)
+ if (wave.channels > 2) TRACELOG(LOG_WARNING, "WAVE: [%s] FLAC channels number (%i) not supported", fileName, wave.channels);
+ TRACELOG(LOG_INFO, "WAVE: [%s] FLAC file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
+ }
+
return wave;
}
#endif
@@ -2095,17 +2100,20 @@ static Wave LoadMP3(const char *fileName)
drmp3_config config = { 0 };
wave.data = drmp3_open_file_and_read_f32(fileName, &config, &totalFrameCount);
- wave.channels = config.outputChannels;
- wave.sampleRate = config.outputSampleRate;
- wave.sampleCount = (int)totalFrameCount*wave.channels;
- wave.sampleSize = 32;
-
- // NOTE: Only support up to 2 channels (mono, stereo)
- if (wave.channels > 2) TRACELOG(LOG_WARNING, "[%s] MP3 channels number (%i) not supported", fileName, wave.channels);
+ if (wave.data == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to load MP3 data", fileName);
+ else
+ {
+ wave.channels = config.outputChannels;
+ wave.sampleRate = config.outputSampleRate;
+ wave.sampleCount = (int)totalFrameCount*wave.channels;
+ wave.sampleSize = 32;
- if (wave.data == NULL) TRACELOG(LOG_WARNING, "[%s] MP3 data could not be loaded", fileName);
- else TRACELOG(LOG_INFO, "[%s] MP3 file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
+ // NOTE: Only support up to 2 channels (mono, stereo)
+ if (wave.channels > 2) TRACELOG(LOG_WARNING, "WAVE: [%s] MP3 channels number (%i) not supported", fileName, wave.channels);
+ TRACELOG(LOG_INFO, "WAVE: [%s] MP3 file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
+ }
+
return wave;
}
#endif