Review coding style to match raylib style

Moved AudioError enum inside audio.c

3 files changed, 199 insertions, 207 deletions

diff --git a/src/audio.c b/src/audio.c
index 452459ef..361b8b55 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -35,29 +35,29 @@
     #include "raylib.h"
 #endif
 
-#include "AL/al.h"          // OpenAL basic header
-#include "AL/alc.h"         // OpenAL context header (like OpenGL, OpenAL requires a context to work)
-#include "AL/alext.h"       // OpenAL extensions for other format types
+#include "AL/al.h"              // OpenAL basic header
+#include "AL/alc.h"             // OpenAL context header (like OpenGL, OpenAL requires a context to work)
+#include "AL/alext.h"           // OpenAL extensions for other format types
 
-#include <stdlib.h>         // Required for: malloc(), free()
-#include <string.h>         // Required for: strcmp(), strncmp()
-#include <stdio.h>          // Required for: FILE, fopen(), fclose(), fread()
+#include <stdlib.h>             // Required for: malloc(), free()
+#include <string.h>             // Required for: strcmp(), strncmp()
+#include <stdio.h>              // Required for: FILE, fopen(), fclose(), fread()
 
 #if defined(AUDIO_STANDALONE)
-    #include <stdarg.h>     // Required for: va_list, va_start(), vfprintf(), va_end()
+    #include <stdarg.h>         // Required for: va_list, va_start(), vfprintf(), va_end()
 #else
-    #include "utils.h"      // Required for: DecompressData()
-                            // NOTE: Includes Android fopen() function map
+    #include "utils.h"          // Required for: DecompressData()
+                                // NOTE: Includes Android fopen() function map
 #endif
 
 //#define STB_VORBIS_HEADER_ONLY
-#include "stb_vorbis.h"     // OGG loading functions
+#include "stb_vorbis.h"         // OGG loading functions
 
 #define JAR_XM_IMPLEMENTATION
-#include "jar_xm.h"         // XM loading functions
+#include "jar_xm.h"             // XM loading functions
 
 #define JAR_MOD_IMPLEMENTATION
-#include "jar_mod.h"        // For playing .mod files
+#include "jar_mod.h"            // MOD loading functions
 
 //----------------------------------------------------------------------------------
 // Defines and Macros
@@ -89,25 +89,45 @@ typedef struct MixChannel_t {
     unsigned char mixChannel;            // 0-3 or mixA-mixD, each mix channel can receive up to one dedicated audio stream
     bool floatingPoint;                  // if false then the short datatype is used instead
     bool playing;                        // false if paused
-    ALenum alFormat;                     // openAL format specifier
-    ALuint alSource;                     // openAL source
-    ALuint alBuffer[MAX_STREAM_BUFFERS]; // openAL sample buffer
+    
+    ALenum alFormat;                     // OpenAL format specifier
+    ALuint alSource;                     // OpenAL source
+    ALuint alBuffer[MAX_STREAM_BUFFERS]; // OpenAL sample buffer
 } MixChannel_t;
 
 // Music type (file streaming from memory)
 // NOTE: Anything longer than ~10 seconds should be streamed into a mix channel...
 typedef struct Music {
     stb_vorbis *stream;
-    jar_xm_context_t *xmctx;   // Stores jar_xm mixc, XM chiptune context
-    jar_mod_context_t modctx;  // Stores mod chiptune context
+    jar_xm_context_t *xmctx;   // XM chiptune context
+    jar_mod_context_t modctx;  // MOD chiptune context
     MixChannel_t *mixc;        // mix channel
     
     unsigned int totalSamplesLeft;
     float totalLengthSeconds;
     bool loop;
-    bool chipTune;             // True if chiptune is loaded
+    bool chipTune;             // chiptune is loaded?
 } Music;
 
+// Audio errors registered
+typedef enum {
+    ERROR_RAW_CONTEXT_CREATION = 1,
+    ERROR_XM_CONTEXT_CREATION = 2,
+    ERROR_MOD_CONTEXT_CREATION = 4,
+    ERROR_MIX_CHANNEL_CREATION = 8,
+    ERROR_MUSIC_CHANNEL_CREATION = 16,
+    ERROR_LOADING_XM = 32,
+    ERROR_LOADING_MOD = 64,
+    ERROR_LOADING_WAV = 128,
+    ERROR_LOADING_OGG = 256,
+    ERROR_OUT_OF_MIX_CHANNELS = 512,
+    ERROR_EXTENSION_NOT_RECOGNIZED = 1024,
+    ERROR_UNABLE_TO_OPEN_RRES_FILE = 2048,
+    ERROR_INVALID_RRES_FILE = 4096,
+    ERROR_INVALID_RRES_RESOURCE = 8192,
+    ERROR_UNINITIALIZED_CHANNELS = 16384
+} AudioError;
+
 #if defined(AUDIO_STANDALONE)
 typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType;
 #endif
@@ -119,6 +139,8 @@ static MixChannel_t* mixChannels_g[MAX_MIX_CHANNELS];        // What mix channel
 static bool musicEnabled_g = false;
 static Music musicChannels_g[MAX_MUSIC_STREAMS];             // Current music loaded, up to two can play at the same time
 
+static lastAudioError = 0;                          // Registers last audio error
+
 //----------------------------------------------------------------------------------
 // Module specific Functions Declaration
 //----------------------------------------------------------------------------------
@@ -129,10 +151,9 @@ static void UnloadWave(Wave wave);                 // Unload wave data
 static bool BufferMusicStream(int index, int numBuffers); // Fill music buffers with data
 static void EmptyMusicStream(int index);                  // Empty music buffers
 
-
-static MixChannel_t* InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint); // For streaming into mix channels.
-static void CloseMixChannel(MixChannel_t* mixc); // Frees mix channel
-static int BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements); // Pushes more audio data into mixc mix channel, if NULL is passed it pauses
+static MixChannel_t *InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint); // For streaming into mix channels.
+static void CloseMixChannel(MixChannel_t *mixc); // Frees mix channel
+static int BufferMixChannel(MixChannel_t *mixc, void *data, int numberElements); // Pushes more audio data into mixc mix channel, if NULL is passed it pauses
 static int FillAlBufferWithSilence(MixChannel_t *mixc, ALuint buffer); // Fill buffer with zeros, returns number processed
 static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len); // Pass two arrays of the same legnth in
 static void ResampleByteToFloat(char *chars, float *floats, unsigned short len); // Pass two arrays of same length in
@@ -153,13 +174,13 @@ void InitAudioDevice(void)
     // Open and initialize a device with default settings
     ALCdevice *device = alcOpenDevice(NULL);
 
-    if(!device) TraceLog(ERROR, "Audio device could not be opened");
+    if (!device) TraceLog(ERROR, "Audio device could not be opened");
 
     ALCcontext *context = alcCreateContext(device, NULL);
 
-    if(context == NULL || alcMakeContextCurrent(context) == ALC_FALSE)
+    if ((context == NULL) || (alcMakeContextCurrent(context) == ALC_FALSE))
     {
-        if(context != NULL) alcDestroyContext(context);
+        if (context != NULL) alcDestroyContext(context);
 
         alcCloseDevice(device);
 
@@ -177,11 +198,10 @@ void InitAudioDevice(void)
 // Close the audio device for all contexts
 void CloseAudioDevice(void)
 {
-    for(int index=0; index<MAX_MUSIC_STREAMS; index++)
+    for (int index=0; index<MAX_MUSIC_STREAMS; index++)
     {
-        if(musicChannels_g[index].mixc) StopMusicStream(index);      // Stop music streaming and close current stream
+        if (musicChannels_g[index].mixc) StopMusicStream(index);      // Stop music streaming and close current stream
     }
-    
 
     ALCdevice *device;
     ALCcontext *context = alcGetCurrentContext();
@@ -199,9 +219,12 @@ void CloseAudioDevice(void)
 bool IsAudioDeviceReady(void)
 {
     ALCcontext *context = alcGetCurrentContext();
+    
     if (context == NULL) return false;
-    else{
+    else
+    {
         ALCdevice *device = alcGetContextsDevice(context);
+        
         if (device == NULL) return false;
         else return true;
     }
@@ -214,33 +237,30 @@ bool IsAudioDeviceReady(void)
 // For streaming into mix channels.
 // The mixChannel is what audio muxing channel you want to operate on, 0-3 are the ones available. Each mix channel can only be used one at a time.
 // exmple usage is InitMixChannel(48000, 0, 2, true); // mixchannel 1, 48khz, stereo, floating point
-static MixChannel_t* InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint)
+static MixChannel_t *InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint)
 {
-    if(mixChannel >= MAX_MIX_CHANNELS) return NULL;
-    if(!IsAudioDeviceReady()) InitAudioDevice();
+    if (mixChannel >= MAX_MIX_CHANNELS) return NULL;
+    if (!IsAudioDeviceReady()) InitAudioDevice();
     
-    if(!mixChannels_g[mixChannel]){
-        MixChannel_t *mixc = (MixChannel_t*)malloc(sizeof(MixChannel_t));
+    if (!mixChannels_g[mixChannel])
+    {
+        MixChannel_t *mixc = (MixChannel_t *)malloc(sizeof(MixChannel_t));
         mixc->sampleRate = sampleRate;
         mixc->channels = channels;
         mixc->mixChannel = mixChannel;
         mixc->floatingPoint = floatingPoint;
         mixChannels_g[mixChannel] = mixc;
         
-        // setup openAL format
-        if(channels == 1)
+        // Setup OpenAL format
+        if (channels == 1)
         {
-            if(floatingPoint)
-                mixc->alFormat = AL_FORMAT_MONO_FLOAT32;
-            else
-                mixc->alFormat = AL_FORMAT_MONO16;
+            if (floatingPoint) mixc->alFormat = AL_FORMAT_MONO_FLOAT32;
+            else mixc->alFormat = AL_FORMAT_MONO16;
         }
-        else if(channels == 2)
+        else if (channels == 2)
         {
-            if(floatingPoint)
-                mixc->alFormat = AL_FORMAT_STEREO_FLOAT32;
-            else
-                mixc->alFormat = AL_FORMAT_STEREO16;
+            if (floatingPoint) mixc->alFormat = AL_FORMAT_STEREO_FLOAT32;
+            else mixc->alFormat = AL_FORMAT_STEREO16;
         }
         
         // Create an audio source
@@ -253,10 +273,8 @@ static MixChannel_t* InitMixChannel(unsigned short sampleRate, unsigned char mix
         // Create Buffer
         alGenBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer);
         
-        //fill buffers
-        int x;
-        for(x=0;x<MAX_STREAM_BUFFERS;x++)
-            FillAlBufferWithSilence(mixc, mixc->alBuffer[x]);
+        // Fill buffers
+        for (int i = 0; i < MAX_STREAM_BUFFERS; i++) FillAlBufferWithSilence(mixc, mixc->alBuffer[i]);
         
         alSourceQueueBuffers(mixc->alSource, MAX_STREAM_BUFFERS, mixc->alBuffer);
         mixc->playing = true;
@@ -264,27 +282,30 @@ static MixChannel_t* InitMixChannel(unsigned short sampleRate, unsigned char mix
         
         return mixc;
     }
+    
     return NULL;
 }
 
 // Frees buffer in mix channel
-static void CloseMixChannel(MixChannel_t* mixc)
+static void CloseMixChannel(MixChannel_t *mixc)
 {
-    if(mixc){
+    if (mixc)
+    {
         alSourceStop(mixc->alSource);
         mixc->playing = false;
         
-        //flush out all queued buffers
+        // Flush out all queued buffers
         ALuint buffer = 0;
         int queued = 0;
         alGetSourcei(mixc->alSource, AL_BUFFERS_QUEUED, &queued);
+        
         while (queued > 0)
         {
             alSourceUnqueueBuffers(mixc->alSource, 1, &buffer);
             queued--;
         }
         
-        //delete source and buffers
+        // Delete source and buffers
         alDeleteSources(1, &mixc->alSource);
         alDeleteBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer);
         mixChannels_g[mixc->mixChannel] = NULL;
@@ -296,39 +317,46 @@ static void CloseMixChannel(MixChannel_t* mixc)
 // Pushes more audio data into mixc mix channel, only one buffer per call
 // Call "BufferMixChannel(mixc, NULL, 0)" if you want to pause the audio.
 // @Returns number of samples that where processed.
-static int BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements)
+static int BufferMixChannel(MixChannel_t *mixc, void *data, int numberElements)
 {
-    if(!mixc || mixChannels_g[mixc->mixChannel] != mixc) return 0; // when there is two channels there must be an even number of samples
+    if (!mixc || (mixChannels_g[mixc->mixChannel] != mixc)) return 0; // When there is two channels there must be an even number of samples
     
-    if (!data || !numberElements)
-    { // pauses audio until data is given
-        if(mixc->playing){
+    if (!data || !numberElements)   
+    { 
+        // Pauses audio until data is given
+        if (mixc->playing)
+        {
             alSourcePause(mixc->alSource);
             mixc->playing = false;
         }
+        
         return 0;
     }
-    else if(!mixc->playing)
-    { // restart audio otherwise
+    else if (!mixc->playing)
+    { 
+        // Restart audio otherwise
         alSourcePlay(mixc->alSource);
         mixc->playing = true;
     }
-    
-   
+
     ALuint buffer = 0;
     
     alSourceUnqueueBuffers(mixc->alSource, 1, &buffer);
-    if(!buffer) return 0;
-    if(mixc->floatingPoint) // process float buffers
+    if (!buffer) return 0;
+    
+    if (mixc->floatingPoint)
     {
-        float *ptr = (float*)data;
+        // Process float buffers
+        float *ptr = (float *)data;
         alBufferData(buffer, mixc->alFormat, ptr, numberElements*sizeof(float), mixc->sampleRate);
     }
-    else // process short buffers
+    else
     {
-        short *ptr = (short*)data;
+        // Process short buffers
+        short *ptr = (short *)data;
         alBufferData(buffer, mixc->alFormat, ptr, numberElements*sizeof(short), mixc->sampleRate);
     }
+    
     alSourceQueueBuffers(mixc->alSource, 1, &buffer);
     
     return numberElements;
@@ -337,15 +365,18 @@ static int BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements)
 // fill buffer with zeros, returns number processed
 static int FillAlBufferWithSilence(MixChannel_t *mixc, ALuint buffer)
 {
-    if(mixc->floatingPoint){
-        float pcm[MUSIC_BUFFER_SIZE_FLOAT] = {0.f};
+    if (mixc->floatingPoint)
+    {
+        float pcm[MUSIC_BUFFER_SIZE_FLOAT] = { 0.0f };
         alBufferData(buffer, mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_FLOAT*sizeof(float), mixc->sampleRate);
+        
         return MUSIC_BUFFER_SIZE_FLOAT;
     }
     else
     {
-        short pcm[MUSIC_BUFFER_SIZE_SHORT] = {0};
+        short pcm[MUSIC_BUFFER_SIZE_SHORT] = { 0 };
         alBufferData(buffer, mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_SHORT*sizeof(short), mixc->sampleRate);
+        
         return MUSIC_BUFFER_SIZE_SHORT;
     }
 }
@@ -355,13 +386,10 @@ static int FillAlBufferWithSilence(MixChannel_t *mixc, ALuint buffer)
 // ResampleShortToFloat(sh,fl,3);
 static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len)
 {
-    int x;
-    for(x=0;x<len;x++)
+    for (int i = 0; i < len; i++)
     {
-        if(shorts[x] < 0)
-            floats[x] = (float)shorts[x] / 32766.f;
-        else
-            floats[x] = (float)shorts[x] / 32767.f;
+        if (shorts[x] < 0) floats[x] = (float)shorts[x]/32766.0f;
+        else floats[x] = (float)shorts[x]/32767.0f;
     }
 }
 
@@ -370,13 +398,10 @@ static void ResampleShortToFloat(short *shorts, float *floats, unsigned short le
 // ResampleByteToFloat(ch,fl,3);
 static void ResampleByteToFloat(char *chars, float *floats, unsigned short len)
 {
-    int x;
-    for(x=0;x<len;x++)
+    for (int i = 0; i < len; i++)
     {
-        if(chars[x] < 0)
-            floats[x] = (float)chars[x] / 127.f;
-        else
-            floats[x] = (float)chars[x] / 128.f;
+        if (chars[x] < 0) floats[x] = (float)chars[x]/127.0f;
+        else floats[x] = (float)chars[x]/128.0f;
     }
 }
 
@@ -385,22 +410,19 @@ static void ResampleByteToFloat(char *chars, float *floats, unsigned short len)
 RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint)
 {
     int mixIndex;
-    for(mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot
+    for (mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot
     {
-        if(mixChannels_g[mixIndex] == NULL) break;
-        else if(mixIndex == MAX_MIX_CHANNELS - 1) return ERROR_OUT_OF_MIX_CHANNELS; // error
+        if (mixChannels_g[mixIndex] == NULL) break;
+        else if (mixIndex == (MAX_MIX_CHANNELS - 1)) return ERROR_OUT_OF_MIX_CHANNELS; // error
     }
     
-    if(InitMixChannel(sampleRate, mixIndex, channels, floatingPoint))
-        return mixIndex;
-    else
-        return ERROR_RAW_CONTEXT_CREATION; // error
+    if (InitMixChannel(sampleRate, mixIndex, channels, floatingPoint)) return mixIndex;
+    else return ERROR_RAW_CONTEXT_CREATION; // error
 }
 
 void CloseRawAudioContext(RawAudioContext ctx)
 {
-    if(mixChannels_g[ctx])
-        CloseMixChannel(mixChannels_g[ctx]);
+    if (mixChannels_g[ctx]) CloseMixChannel(mixChannels_g[ctx]);
 }
 
 // if 0 is returned, the buffers are still full and you need to keep trying with the same data until a + number is returned.
@@ -409,18 +431,16 @@ void CloseRawAudioContext(RawAudioContext ctx)
 int BufferRawAudioContext(RawAudioContext ctx, void *data, unsigned short numberElements)
 {
     int numBuffered = 0;
-    if(ctx >= 0)
+    
+    if (ctx >= 0)
     {
         MixChannel_t* mixc = mixChannels_g[ctx];
         numBuffered = BufferMixChannel(mixc, data, numberElements);
     }
+    
     return numBuffered;
 }
 
-
-
-
-
 //----------------------------------------------------------------------------------
 // Module Functions Definition - Sounds loading and playing (.WAV)
 //----------------------------------------------------------------------------------
@@ -438,9 +458,12 @@ Sound LoadSound(char *fileName)
 
     if (strcmp(GetExtension(fileName),"wav") == 0) wave = LoadWAV(fileName);
     else if (strcmp(GetExtension(fileName),"ogg") == 0) wave = LoadOGG(fileName);
-    else{
+    else
+    {
         TraceLog(WARNING, "[%s] Sound extension not recognized, it can't be loaded", fileName);
-        sound.error = ERROR_EXTENSION_NOT_RECOGNIZED; //error
+        
+        // TODO: Find a better way to register errors (similar to glGetError())
+        lastAudioError = ERROR_EXTENSION_NOT_RECOGNIZED;
     }
 
     if (wave.data != NULL)
@@ -568,7 +591,7 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
     if (rresFile == NULL)
     {
         TraceLog(WARNING, "[%s] rRES raylib resource file could not be opened", rresName);
-        sound.error = ERROR_UNABLE_TO_OPEN_RRES_FILE; //error
+        lastAudioError = ERROR_UNABLE_TO_OPEN_RRES_FILE;
     }
     else
     {
@@ -583,7 +606,7 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
         if ((id[0] != 'r') && (id[1] != 'R') && (id[2] != 'E') &&(id[3] != 'S'))
         {
             TraceLog(WARNING, "[%s] This is not a valid raylib resource file", rresName);
-            sound.error = ERROR_INVALID_RRES_FILE;
+            lastAudioError = ERROR_INVALID_RRES_FILE;
         }
         else
         {
@@ -674,7 +697,7 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
                     else
                     {
                         TraceLog(WARNING, "[%s] Required resource do not seem to be a valid SOUND resource", rresName);
-                        sound.error = ERROR_INVALID_RRES_RESOURCE;
+                        lastAudioError = ERROR_INVALID_RRES_RESOURCE;
                     }
                 }
                 else
@@ -780,12 +803,12 @@ int PlayMusicStream(int musicIndex, char *fileName)
 {
     int mixIndex;
     
-    if(musicChannels_g[musicIndex].stream || musicChannels_g[musicIndex].xmctx) return ERROR_UNINITIALIZED_CHANNELS; // error
+    if (musicChannels_g[musicIndex].stream || musicChannels_g[musicIndex].xmctx) return ERROR_UNINITIALIZED_CHANNELS; // error
     
-    for(mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot
+    for (mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot
     {
-        if(mixChannels_g[mixIndex] == NULL) break;
-        else if(mixIndex == MAX_MIX_CHANNELS - 1) return ERROR_OUT_OF_MIX_CHANNELS; // error
+        if (mixChannels_g[mixIndex] == NULL) break;
+        else if (mixIndex == (MAX_MIX_CHANNELS - 1)) return ERROR_OUT_OF_MIX_CHANNELS; // error
     }
     
     if (strcmp(GetExtension(fileName),"ogg") == 0)
@@ -814,21 +837,24 @@ int PlayMusicStream(int musicIndex, char *fileName)
             musicChannels_g[musicIndex].totalSamplesLeft = (unsigned int)stb_vorbis_stream_length_in_samples(musicChannels_g[musicIndex].stream) * info.channels;
             musicChannels_g[musicIndex].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(musicChannels_g[musicIndex].stream);
             
-            if (info.channels == 2){
+            if (info.channels == 2)
+            {
                 musicChannels_g[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 2, false);
                 musicChannels_g[musicIndex].mixc->playing = true;
             }
-            else{
+            else
+            {
                 musicChannels_g[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 1, false);
                 musicChannels_g[musicIndex].mixc->playing = true;
             }
-            if(!musicChannels_g[musicIndex].mixc) return ERROR_LOADING_OGG; // error
+            
+            if (!musicChannels_g[musicIndex].mixc) return ERROR_LOADING_OGG; // error
         }
     }
     else if (strcmp(GetExtension(fileName),"xm") == 0)
     {
         // only stereo is supported for xm
-        if(!jar_xm_create_context_from_file(&musicChannels_g[musicIndex].xmctx, 48000, fileName))
+        if (!jar_xm_create_context_from_file(&musicChannels_g[musicIndex].xmctx, 48000, fileName))
         {
             musicChannels_g[musicIndex].chipTune = true;
             musicChannels_g[musicIndex].loop = true;
@@ -841,7 +867,9 @@ int PlayMusicStream(int musicIndex, char *fileName)
             TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, musicChannels_g[musicIndex].totalLengthSeconds);
             
             musicChannels_g[musicIndex].mixc = InitMixChannel(48000, mixIndex, 2, true);
-            if(!musicChannels_g[musicIndex].mixc) return ERROR_XM_CONTEXT_CREATION; // error
+            
+            if (!musicChannels_g[musicIndex].mixc) return ERROR_XM_CONTEXT_CREATION; // error
+            
             musicChannels_g[musicIndex].mixc->playing = true;
         }
         else
@@ -853,7 +881,8 @@ int PlayMusicStream(int musicIndex, char *fileName)
     else if (strcmp(GetExtension(fileName),"mod") == 0)
     {
         jar_mod_init(&musicChannels_g[musicIndex].modctx);
-        if(jar_mod_load_file(&musicChannels_g[musicIndex].modctx, fileName))
+        
+        if (jar_mod_load_file(&musicChannels_g[musicIndex].modctx, fileName))
         {
             musicChannels_g[musicIndex].chipTune = true;
             musicChannels_g[musicIndex].loop = true;
@@ -865,7 +894,9 @@ int PlayMusicStream(int musicIndex, char *fileName)
             TraceLog(INFO, "[%s] MOD track length: %11.6f sec", fileName, musicChannels_g[musicIndex].totalLengthSeconds);
             
             musicChannels_g[musicIndex].mixc = InitMixChannel(48000, mixIndex, 2, false);
-            if(!musicChannels_g[musicIndex].mixc) return ERROR_MOD_CONTEXT_CREATION; // error
+            
+            if (!musicChannels_g[musicIndex].mixc) return ERROR_MOD_CONTEXT_CREATION; // error
+            
             musicChannels_g[musicIndex].mixc->playing = true;
         }
         else
@@ -879,6 +910,7 @@ int PlayMusicStream(int musicIndex, char *fileName)
         TraceLog(WARNING, "[%s] Music extension not recognized, it can't be loaded", fileName);
         return ERROR_EXTENSION_NOT_RECOGNIZED; // error
     }
+    
     return 0; // normal return
 }
 
@@ -894,17 +926,12 @@ void StopMusicStream(int index)
             jar_xm_free_context(musicChannels_g[index].xmctx);
             musicChannels_g[index].xmctx = 0;
         }
-        else if(musicChannels_g[index].chipTune && musicChannels_g[index].modctx.mod_loaded)
-        {
-            jar_mod_unload(&musicChannels_g[index].modctx);
-        }
-        else
-        {
-            stb_vorbis_close(musicChannels_g[index].stream);
-        }
+        else if (musicChannels_g[index].chipTune && musicChannels_g[index].modctx.mod_loaded) jar_mod_unload(&musicChannels_g[index].modctx);
+        else stb_vorbis_close(musicChannels_g[index].stream);
         
-        if(!getMusicStreamCount()) musicEnabled_g = false;
-        if(musicChannels_g[index].stream || musicChannels_g[index].xmctx)
+        if (!GetMusicStreamCount()) musicEnabled_g = false;
+        
+        if (musicChannels_g[index].stream || musicChannels_g[index].xmctx)
         {
             musicChannels_g[index].stream = NULL;
             musicChannels_g[index].xmctx = NULL;
@@ -913,11 +940,15 @@ void StopMusicStream(int index)
 }
 
 //get number of music channels active at this time, this does not mean they are playing
-int getMusicStreamCount(void)
+int GetMusicStreamCount(void)
 {
     int musicCount = 0;
-    for(int musicIndex = 0; musicIndex < MAX_MUSIC_STREAMS; musicIndex++) // find empty music slot
+    
+    // Find empty music slot
+    for (int musicIndex = 0; musicIndex < MAX_MUSIC_STREAMS; musicIndex++)
+    {
         if(musicChannels_g[musicIndex].stream != NULL || musicChannels_g[musicIndex].chipTune) musicCount++;
+    }
     
     return musicCount;
 }
@@ -939,8 +970,11 @@ void ResumeMusicStream(int index)
 {
     // Resume music playing... if music available!
     ALenum state;
-    if(index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc){
+    
+    if (index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc)
+    {
         alGetSourcei(musicChannels_g[index].mixc->alSource, AL_SOURCE_STATE, &state);
+        
         if (state == AL_PAUSED)
         {
             TraceLog(INFO, "Resuming music stream");
@@ -956,8 +990,10 @@ bool IsMusicPlaying(int index)
     bool playing = false;
     ALint state;
     
-    if(index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc){
+    if (index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc)
+    {
         alGetSourcei(musicChannels_g[index].mixc->alSource, AL_SOURCE_STATE, &state);
+        
         if (state == AL_PLAYING) playing = true;
     }
 
@@ -967,14 +1003,17 @@ bool IsMusicPlaying(int index)
 // Set volume for music
 void SetMusicVolume(int index, float volume)
 {
-    if(index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc){
+    if (index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc)
+    {
         alSourcef(musicChannels_g[index].mixc->alSource, AL_GAIN, volume);
     }
 }
 
+// Set pitch for music
 void SetMusicPitch(int index, float pitch)
 {
-    if(index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc){
+    if (index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc)
+    {
         alSourcef(musicChannels_g[index].mixc->alSource, AL_PITCH, pitch);
     }
 }
@@ -983,14 +1022,9 @@ void SetMusicPitch(int index, float pitch)
 float GetMusicTimeLength(int index)
 {
     float totalSeconds;
-    if (musicChannels_g[index].chipTune)
-    {
-        totalSeconds = (float)musicChannels_g[index].totalLengthSeconds;
-    }
-    else
-    {
-        totalSeconds = stb_vorbis_stream_length_in_seconds(musicChannels_g[index].stream);
-    }
+    
+    if (musicChannels_g[index].chipTune) totalSeconds = (float)musicChannels_g[index].totalLengthSeconds;
+    else totalSeconds = stb_vorbis_stream_length_in_seconds(musicChannels_g[index].stream);
 
     return totalSeconds;
 }
@@ -999,7 +1033,8 @@ float GetMusicTimeLength(int index)
 float GetMusicTimePlayed(int index)
 {
     float secondsPlayed = 0.0f;
-    if(index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc)
+    
+    if (index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc)
     {
         if (musicChannels_g[index].chipTune && musicChannels_g[index].xmctx)
         {
@@ -1033,33 +1068,33 @@ static bool BufferMusicStream(int index, int numBuffers)
     short pcm[MUSIC_BUFFER_SIZE_SHORT];
     float pcmf[MUSIC_BUFFER_SIZE_FLOAT];
     
-    int  size = 0;              // Total size of data steamed in L+R samples for xm floats, individual L or R for ogg shorts
-    bool active = true;         // We can get more data from stream (not finished)
+    int size = 0;              // Total size of data steamed in L+R samples for xm floats, individual L or R for ogg shorts
+    bool active = true;        // We can get more data from stream (not finished)
     
     if (musicChannels_g[index].chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
     {
-        for(int x=0; x<numBuffers; x++)
+        for (int i = 0; i < numBuffers; i++)
         {
-            if(musicChannels_g[index].modctx.mod_loaded){
-                if(musicChannels_g[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT)
-                    size = MUSIC_BUFFER_SIZE_SHORT / 2;
-                else
-                    size = musicChannels_g[index].totalSamplesLeft / 2;
+            if (musicChannels_g[index].modctx.mod_loaded)
+            {
+                if (musicChannels_g[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT) size = MUSIC_BUFFER_SIZE_SHORT/2;
+                else size = musicChannels_g[index].totalSamplesLeft/2;
+                
                 jar_mod_fillbuffer(&musicChannels_g[index].modctx, pcm, size, 0 );
-                BufferMixChannel(musicChannels_g[index].mixc, pcm, size * 2);
+                BufferMixChannel(musicChannels_g[index].mixc, pcm, size*2);
             }
-            else if(musicChannels_g[index].xmctx){
-                if(musicChannels_g[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_FLOAT)
-                    size = MUSIC_BUFFER_SIZE_FLOAT / 2;
-                else
-                    size = musicChannels_g[index].totalSamplesLeft / 2;
+            else if (musicChannels_g[index].xmctx)
+            {
+                if (musicChannels_g[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_FLOAT) size = MUSIC_BUFFER_SIZE_FLOAT/2;
+                else size = musicChannels_g[index].totalSamplesLeft/2;
+                
                 jar_xm_generate_samples(musicChannels_g[index].xmctx, pcmf, size); // reads 2*readlen shorts and moves them to buffer+size memory location
-                BufferMixChannel(musicChannels_g[index].mixc, pcmf, size * 2);
+                BufferMixChannel(musicChannels_g[index].mixc, pcmf, size*2);
             }
-            
 
             musicChannels_g[index].totalSamplesLeft -= size;
-            if(musicChannels_g[index].totalSamplesLeft <= 0)
+            
+            if (musicChannels_g[index].totalSamplesLeft <= 0)
             {
                 active = false;
                 break;
@@ -1068,17 +1103,16 @@ static bool BufferMusicStream(int index, int numBuffers)
     }
     else
     {
-        if(musicChannels_g[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT)
-            size = MUSIC_BUFFER_SIZE_SHORT;
-        else
-            size = musicChannels_g[index].totalSamplesLeft;
+        if (musicChannels_g[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT) size = MUSIC_BUFFER_SIZE_SHORT;
+        else size = musicChannels_g[index].totalSamplesLeft;
         
-        for(int x=0; x<numBuffers; x++)
+        for (int i = 0; i < numBuffers; i++)
         {
             int streamedBytes = stb_vorbis_get_samples_short_interleaved(musicChannels_g[index].stream, musicChannels_g[index].mixc->channels, pcm, size);
             BufferMixChannel(musicChannels_g[index].mixc, pcm, streamedBytes * musicChannels_g[index].mixc->channels);
             musicChannels_g[index].totalSamplesLeft -= streamedBytes * musicChannels_g[index].mixc->channels;
-            if(musicChannels_g[index].totalSamplesLeft <= 0)
+            
+            if (musicChannels_g[index].totalSamplesLeft <= 0)
             {
                 active = false;
                 break;
@@ -1105,7 +1139,7 @@ static void EmptyMusicStream(int index)
     }
 }
 
-//determine if a music stream is ready to be written to
+// Determine if a music stream is ready to be written
 static int IsMusicStreamReadyForBuffering(int index)
 {
     ALint processed = 0;
@@ -1120,7 +1154,7 @@ void UpdateMusicStream(int index)
     bool active = true;
     int numBuffers = IsMusicStreamReadyForBuffering(index);
     
-    if (musicChannels_g[index].mixc->playing && index < MAX_MUSIC_STREAMS && musicEnabled_g && musicChannels_g[index].mixc && numBuffers)
+    if (musicChannels_g[index].mixc->playing && (index < MAX_MUSIC_STREAMS) && musicEnabled_g && musicChannels_g[index].mixc && numBuffers)
     {
         active = BufferMusicStream(index, numBuffers);
         
@@ -1136,9 +1170,9 @@ void UpdateMusicStream(int index)
                 stb_vorbis_seek_start(musicChannels_g[index].stream);
                 musicChannels_g[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(musicChannels_g[index].stream) * musicChannels_g[index].mixc->channels;
             }
+            
             active = true;
         }
-        
 
         if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data...");
         
@@ -1149,9 +1183,6 @@ void UpdateMusicStream(int index)
         if (!active) StopMusicStream(index);
         
     }
-    else
-        return;
-
 }
 
 // Load WAV file into Wave structure
diff --git a/src/audio.h b/src/audio.h
index a7475566..e30087ba 100644
--- a/src/audio.h
+++ b/src/audio.h
@@ -47,24 +47,6 @@
     #endif
 #endif
 
-typedef enum {
-    ERROR_RAW_CONTEXT_CREATION = 1,
-    ERROR_XM_CONTEXT_CREATION = 2,
-    ERROR_MOD_CONTEXT_CREATION = 4,
-    ERROR_MIX_CHANNEL_CREATION = 8,
-    ERROR_MUSIC_CHANNEL_CREATION = 16,
-    ERROR_LOADING_XM = 32,
-    ERROR_LOADING_MOD = 64,
-    ERROR_LOADING_WAV = 128,
-    ERROR_LOADING_OGG = 256,
-    ERROR_OUT_OF_MIX_CHANNELS = 512,
-    ERROR_EXTENSION_NOT_RECOGNIZED = 1024,
-    ERROR_UNABLE_TO_OPEN_RRES_FILE = 2048,
-    ERROR_INVALID_RRES_FILE = 4096,
-    ERROR_INVALID_RRES_RESOURCE = 8192,
-    ERROR_UNINITIALIZED_CHANNELS = 16384
-} AudioError;
-
 // Sound source type
 typedef struct Sound {
     unsigned int source;
@@ -120,7 +102,7 @@ bool IsMusicPlaying(int index);                                 // Check if musi
 void SetMusicVolume(int index, float volume);                   // Set volume for music (1.0 is max level)
 float GetMusicTimeLength(int index);                            // Get music time length (in seconds)
 float GetMusicTimePlayed(int index);                            // Get current music time played (in seconds)
-int getMusicStreamCount(void);
+int GetMusicStreamCount(void);
 void SetMusicPitch(int index, float pitch);
 
 // used to output raw audio streams, returns negative numbers on error
diff --git a/src/raylib.h b/src/raylib.h
index 21892d68..97f4a2e6 100644
--- a/src/raylib.h
+++ b/src/raylib.h
@@ -452,29 +452,10 @@ typedef struct Ray {
     Vector3 direction;
 } Ray;
 
-typedef enum { // allows errors to be & together
-    ERROR_RAW_CONTEXT_CREATION = 1,
-    ERROR_XM_CONTEXT_CREATION = 2,
-    ERROR_MOD_CONTEXT_CREATION = 4,
-    ERROR_MIX_CHANNEL_CREATION = 8,
-    ERROR_MUSIC_CHANNEL_CREATION = 16,
-    ERROR_LOADING_XM = 32,
-    ERROR_LOADING_MOD = 64,
-    ERROR_LOADING_WAV = 128,
-    ERROR_LOADING_OGG = 256,
-    ERROR_OUT_OF_MIX_CHANNELS = 512,
-    ERROR_EXTENSION_NOT_RECOGNIZED = 1024,
-    ERROR_UNABLE_TO_OPEN_RRES_FILE = 2048,
-    ERROR_INVALID_RRES_FILE = 4096,
-    ERROR_INVALID_RRES_RESOURCE = 8192,
-    ERROR_UNINITIALIZED_CHANNELS = 16384
-} AudioError;
-
 // Sound source type
 typedef struct Sound {
     unsigned int source;
     unsigned int buffer;
-    AudioError error; // if there was any error during the creation or use of this Sound
 } Sound;
 
 // Wave type, defines audio wave data
@@ -488,8 +469,6 @@ typedef struct Wave {
 
 typedef int RawAudioContext;
 
-
-
 // Texture formats
 // NOTE: Support depends on OpenGL version and platform
 typedef enum {
@@ -940,7 +919,7 @@ bool IsMusicPlaying(int index);                                 // Check if musi
 void SetMusicVolume(int index, float volume);                   // Set volume for music (1.0 is max level)
 float GetMusicTimeLength(int index);                            // Get current music time length (in seconds)
 float GetMusicTimePlayed(int index);                            // Get current music time played (in seconds)
-int getMusicStreamCount(void);
+int GetMusicStreamCount(void);
 void SetMusicPitch(int index, float pitch);
 
 // used to output raw audio streams, returns negative numbers on error