From 6db44500b75097a581587cba15ce703963d2ced8 Mon Sep 17 00:00:00 2001
From: Joshua Reisenauer <kd7tck@msn.com>
Date: Wed, 11 May 2016 00:37:10 -0700
Subject: [PATCH 01/14] adding multiple music streams

---
 src/audio.c  | 181 +++++++++++++++++++++++----------------------------
 src/audio.h  |   9 +--
 src/raylib.h |   9 +--
 3 files changed, 91 insertions(+), 108 deletions(-)

diff --git a/src/audio.c b/src/audio.c
index fbf53df6..39be4eeb 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -61,6 +61,7 @@
 //----------------------------------------------------------------------------------
 #define MAX_STREAM_BUFFERS          2
 #define MAX_AUDIO_CONTEXTS          4             // Number of open AL sources
+#define MAX_MUSIC_STREAMS           2
 
 #if defined(PLATFORM_RPI) || defined(PLATFORM_ANDROID)
     // NOTE: On RPI and Android should be lower to avoid frame-stalls
@@ -81,13 +82,8 @@
 typedef struct Music {
     stb_vorbis *stream;
     jar_xm_context_t *chipctx; // Stores jar_xm context
-
-    ALuint buffers[MAX_STREAM_BUFFERS];
-    ALuint source;
-    ALenum format;
-
-    int channels;
-    int sampleRate;
+    AudioContext_t *ctx; // audio context
+    
     int totalSamplesLeft;
     float totalLengthSeconds;
     bool loop;
@@ -117,8 +113,8 @@ typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType;
 //----------------------------------------------------------------------------------
 static AudioContext_t* mixChannelsActive_g[MAX_AUDIO_CONTEXTS];      // What mix channels are currently active
 static bool musicEnabled = false;
-static Music currentMusic;          // Current music loaded
-                                    // NOTE: Only one music file playing at a time
+static Music currentMusic[2];                                        // Current music loaded, up to two can play at the same time
+
 //----------------------------------------------------------------------------------
 // Module specific Functions Declaration
 //----------------------------------------------------------------------------------
@@ -769,151 +765,129 @@ void SetSoundPitch(Sound sound, float pitch)
 // Start music playing (open stream)
 void PlayMusicStream(char *fileName)
 {
+    int musicIndex;
+    int mixIndex;
+    for(musicIndex = 0; musicIndex < MAX_MUSIC_STREAMS; musicIndex++) // find empty music slot
+    {
+        if(currentMusic[musicIndex].stream == NULL && !currentMusic[musicIndex].chipTune) break;
+        else if(musicIndex = MAX_MUSIC_STREAMS - 1) return;
+    }
+    for(mixIndex = 0; mixIndex < MAX_AUDIO_CONTEXTS; mixIndex++) // find empty mix channel slot
+    {
+        if(mixChannelsActive_g[mixIndex] == NULL) break;
+        else if(musicIndex = MAX_AUDIO_CONTEXTS - 1) return;
+    }
+    
+    
     if (strcmp(GetExtension(fileName),"ogg") == 0)
     {
-        // Stop current music, clean buffers, unload current stream
-        StopMusicStream();
-
         // Open audio stream
-        currentMusic.stream = stb_vorbis_open_filename(fileName, NULL, NULL);
+        currentMusic[musicIndex].stream = stb_vorbis_open_filename(fileName, NULL, NULL);
 
-        if (currentMusic.stream == NULL)
+        if (currentMusic[musicIndex].stream == NULL)
         {
             TraceLog(WARNING, "[%s] OGG audio file could not be opened", fileName);
         }
         else
         {
             // Get file info
-            stb_vorbis_info info = stb_vorbis_get_info(currentMusic.stream);
-
-            currentMusic.channels = info.channels;
-            currentMusic.sampleRate = info.sample_rate;
+            stb_vorbis_info info = stb_vorbis_get_info(currentMusic[musicIndex].stream);
 
             TraceLog(INFO, "[%s] Ogg sample rate: %i", fileName, info.sample_rate);
             TraceLog(INFO, "[%s] Ogg channels: %i", fileName, info.channels);
             TraceLog(DEBUG, "[%s] Temp memory required: %i", fileName, info.temp_memory_required);
 
-            if (info.channels == 2) currentMusic.format = AL_FORMAT_STEREO16;
-            else currentMusic.format = AL_FORMAT_MONO16;
+            if (info.channels == 2){
+                currentMusic[musicIndex].ctx = InitAudioContext(info.sample_rate, mixIndex, 2, false);
+            }
+            else{
+                currentMusic[musicIndex].ctx = InitAudioContext(info.sample_rate, mixIndex, 1, false);
+            }
 
-            currentMusic.loop = true;                  // We loop by default
+            currentMusic[musicIndex].loop = true;                  // We loop by default
             musicEnabled = true;
 
-            // Create an audio source
-            alGenSources(1, &currentMusic.source);     // Generate pointer to audio source
-
-            alSourcef(currentMusic.source, AL_PITCH, 1);
-            alSourcef(currentMusic.source, AL_GAIN, 1);
-            alSource3f(currentMusic.source, AL_POSITION, 0, 0, 0);
-            alSource3f(currentMusic.source, AL_VELOCITY, 0, 0, 0);
-            //alSourcei(currentMusic.source, AL_LOOPING, AL_TRUE);     // ERROR: Buffers do not queue!
-
-            // Generate two OpenAL buffers
-            alGenBuffers(2, currentMusic.buffers);
-
-            // Fill buffers with music...
-            BufferMusicStream(currentMusic.buffers[0]);
-            BufferMusicStream(currentMusic.buffers[1]);
-
-            // Queue buffers and start playing
-            alSourceQueueBuffers(currentMusic.source, 2, currentMusic.buffers);
-            alSourcePlay(currentMusic.source);
-
-            // NOTE: Regularly, we must check if a buffer has been processed and refill it: UpdateMusicStream()
-
-            currentMusic.totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
-            currentMusic.totalLengthSeconds = stb_vorbis_stream_length_in_seconds(currentMusic.stream);
+            currentMusic[musicIndex].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[musicIndex].stream) * currentMusic[musicIndex].channels;
+            currentMusic[musicIndex].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[musicIndex].stream);
         }
     }
     else if (strcmp(GetExtension(fileName),"xm") == 0)
     {
-        // Stop current music, clean buffers, unload current stream
-        StopMusicStream();
-        
-        // new song settings for xm chiptune
-        currentMusic.chipTune = true;
-        currentMusic.channels = 2;
-        currentMusic.sampleRate = 48000;
-        currentMusic.loop = true;
-        
         // only stereo is supported for xm
-        if(!jar_xm_create_context_from_file(&currentMusic.chipctx, currentMusic.sampleRate, fileName))
+        if(!jar_xm_create_context_from_file(&currentMusic[musicIndex].chipctx, 48000, fileName))
         {
-            currentMusic.format = AL_FORMAT_STEREO16;
-            jar_xm_set_max_loop_count(currentMusic.chipctx, 0); // infinite number of loops
-            currentMusic.totalSamplesLeft =  jar_xm_get_remaining_samples(currentMusic.chipctx);
-            currentMusic.totalLengthSeconds = ((float)currentMusic.totalSamplesLeft) / ((float)currentMusic.sampleRate);
+            currentMusic[musicIndex].chipTune = true;
+            currentMusic[musicIndex].loop = true;
+            jar_xm_set_max_loop_count(currentMusic[musicIndex].chipctx, 0); // infinite number of loops
+            currentMusic[musicIndex].totalSamplesLeft =  jar_xm_get_remaining_samples(currentMusic[musicIndex].chipctx);
+            currentMusic[musicIndex].totalLengthSeconds = ((float)currentMusic[musicIndex].totalSamplesLeft) / ((float)currentMusic[musicIndex].sampleRate);
             musicEnabled = true;
             
-            TraceLog(INFO, "[%s] XM number of samples: %i", fileName, currentMusic.totalSamplesLeft);
-            TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, currentMusic.totalLengthSeconds);
-            
-            // Set up OpenAL
-            alGenSources(1, &currentMusic.source);
-            alSourcef(currentMusic.source, AL_PITCH, 1);
-            alSourcef(currentMusic.source, AL_GAIN, 1);
-            alSource3f(currentMusic.source, AL_POSITION, 0, 0, 0);
-            alSource3f(currentMusic.source, AL_VELOCITY, 0, 0, 0);
-            alGenBuffers(2, currentMusic.buffers);
-            BufferMusicStream(currentMusic.buffers[0]);
-            BufferMusicStream(currentMusic.buffers[1]);
-            alSourceQueueBuffers(currentMusic.source, 2, currentMusic.buffers);
-            alSourcePlay(currentMusic.source);
+            TraceLog(INFO, "[%s] XM number of samples: %i", fileName, currentMusic[musicIndex].totalSamplesLeft);
+            TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, currentMusic[musicIndex].totalLengthSeconds);
             
-            // NOTE: Regularly, we must check if a buffer has been processed and refill it: UpdateMusicStream()
+            currentMusic[musicIndex].ctx = InitAudioContext(48000, mixIndex, 2, true);
         }
         else TraceLog(WARNING, "[%s] XM file could not be opened", fileName);
     }
     else TraceLog(WARNING, "[%s] Music extension not recognized, it can't be loaded", fileName);
 }
 
-// Stop music playing (close stream)
-void StopMusicStream(void)
+// Stop music playing for individual music index of currentMusic array (close stream)
+void StopMusicStream(int index)
 {
-    if (musicEnabled)
+    if (index < MAX_MUSIC_STREAMS && currentMusic[index].ctx)
     {
-        alSourceStop(currentMusic.source);
-        EmptyMusicStream(); // Empty music buffers
-        alDeleteSources(1, &currentMusic.source);
-        alDeleteBuffers(2, currentMusic.buffers);
+        CloseAudioContext(currentMusic[index].ctx);
         
-        if (currentMusic.chipTune)
+        if (currentMusic[index].chipTune)
         {
-            jar_xm_free_context(currentMusic.chipctx);
+            jar_xm_free_context(currentMusic[index].chipctx);
         }
         else
         {
-            stb_vorbis_close(currentMusic.stream);
+            stb_vorbis_close(currentMusic[index].stream);
         }
+        
+        if(!getMusicStreamCount()) musicEnabled = false;
     }
+}
 
-    musicEnabled = false;
+//get number of music channels active at this time, this does not mean they are playing
+int getMusicStreamCount(void)
+{
+    int musicCount;
+    for(int musicIndex = 0; musicIndex < MAX_MUSIC_STREAMS; musicIndex++) // find empty music slot
+        if(currentMusic[musicIndex].stream != NULL || currentMusic[musicIndex].chipTune) musicCount++;
+        
+    return musicCount;
 }
 
 // Pause music playing
-void PauseMusicStream(void)
+void PauseMusicStream(int index)
 {
     // Pause music stream if music available!
     if (musicEnabled)
     {
         TraceLog(INFO, "Pausing music stream");
-        alSourcePause(currentMusic.source);
+        UpdateAudioContext(currentMusic[index].ctx, NULL, 0); // pushing null data auto pauses stream
         musicEnabled = false;
     }
 }
 
 // Resume music playing
-void ResumeMusicStream(void)
+void ResumeMusicStream(int index)
 {
     // Resume music playing... if music available!
     ALenum state;
-    alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state);
-
-    if (state == AL_PAUSED)
-    {
-        TraceLog(INFO, "Resuming music stream");
-        alSourcePlay(currentMusic.source);
-        musicEnabled = true;
+    if(currentMusic[musicIndex].ctx){
+        alGetSourcei(currentMusic[musicIndex].ctx->alSource, AL_SOURCE_STATE, &state);
+        if (state == AL_PAUSED)
+        {
+            TraceLog(INFO, "Resuming music stream");
+            alSourcePlay(currentMusic[musicIndex].ctx->alSource);
+            musicEnabled = true;
+        }
     }
 }
 
@@ -922,17 +896,24 @@ bool IsMusicPlaying(void)
 {
     bool playing = false;
     ALint state;
-
-    alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state);
-    if (state == AL_PLAYING) playing = true;
+    
+    for(int musicIndex = 0; musicIndex < MAX_MUSIC_STREAMS; musicIndex++)
+    {
+        if(currentMusic[musicIndex].ctx){
+            alGetSourcei(currentMusic[musicIndex].ctx->alSource, AL_SOURCE_STATE, &state);
+            if (state == AL_PLAYING) playing = true;
+        }
+    }
 
     return playing;
 }
 
 // Set volume for music
-void SetMusicVolume(float volume)
+void SetMusicVolume(int index, float volume)
 {
-    alSourcef(currentMusic.source, AL_GAIN, volume);
+    if(currentMusic[musicIndex].ctx){
+        alSourcef(currentMusic[musicIndex].ctx->alSource, AL_GAIN, volume);
+    }
 }
 
 // Get current music time length (in seconds)
diff --git a/src/audio.h b/src/audio.h
index afd881b7..a1602bd9 100644
--- a/src/audio.h
+++ b/src/audio.h
@@ -102,13 +102,14 @@ void SetSoundPitch(Sound sound, float pitch);                   // Set pitch for
 
 void PlayMusicStream(char *fileName);                           // Start music playing (open stream)
 void UpdateMusicStream(void);                                   // Updates buffers for music streaming
-void StopMusicStream(void);                                     // Stop music playing (close stream)
-void PauseMusicStream(void);                                    // Pause music playing
-void ResumeMusicStream(void);                                   // Resume playing paused music
+void StopMusicStream(int index);                                // Stop music playing (close stream)
+void PauseMusicStream(int index);                               // Pause music playing
+void ResumeMusicStream(int index);                              // Resume playing paused music
 bool IsMusicPlaying(void);                                      // Check if music is playing
-void SetMusicVolume(float volume);                              // Set volume for music (1.0 is max level)
+void SetMusicVolume(int index, float volume);                   // Set volume for music (1.0 is max level)
 float GetMusicTimeLength(void);                                 // Get music time length (in seconds)
 float GetMusicTimePlayed(void);                                 // Get current music time played (in seconds)
+int getMusicStreamCount(void);
 
 #ifdef __cplusplus
 }
diff --git a/src/raylib.h b/src/raylib.h
index d464c8e9..0c49a085 100644
--- a/src/raylib.h
+++ b/src/raylib.h
@@ -893,13 +893,14 @@ void SetSoundPitch(Sound sound, float pitch);                   // Set pitch for
 
 void PlayMusicStream(char *fileName);                           // Start music playing (open stream)
 void UpdateMusicStream(void);                                   // Updates buffers for music streaming
-void StopMusicStream(void);                                     // Stop music playing (close stream)
-void PauseMusicStream(void);                                    // Pause music playing
-void ResumeMusicStream(void);                                   // Resume playing paused music
+void StopMusicStream(int index);                                // Stop music playing (close stream)
+void PauseMusicStream(int index);                               // Pause music playing
+void ResumeMusicStream(int index);                              // Resume playing paused music
 bool IsMusicPlaying(void);                                      // Check if music is playing
-void SetMusicVolume(float volume);                              // Set volume for music (1.0 is max level)
+void SetMusicVolume(int index, float volume);                   // Set volume for music (1.0 is max level)
 float GetMusicTimeLength(void);                                 // Get current music time length (in seconds)
 float GetMusicTimePlayed(void);                                 // Get current music time played (in seconds)
+int getMusicStreamCount(void);
 
 #ifdef __cplusplus
 }

From ad3d270c429844462f3fd62fa12257760fac24b5 Mon Sep 17 00:00:00 2001
From: Joshua Reisenauer <kd7tck@msn.com>
Date: Wed, 11 May 2016 18:14:59 -0700
Subject: [PATCH 02/14] added set pitch for music streams

---
 src/audio.c  | 40 ++++++++++++++++++++++------------------
 src/audio.h  |  3 ++-
 src/raylib.h |  3 ++-
 3 files changed, 26 insertions(+), 20 deletions(-)

diff --git a/src/audio.c b/src/audio.c
index 39be4eeb..f89be8bf 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -113,7 +113,7 @@ typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType;
 //----------------------------------------------------------------------------------
 static AudioContext_t* mixChannelsActive_g[MAX_AUDIO_CONTEXTS];      // What mix channels are currently active
 static bool musicEnabled = false;
-static Music currentMusic[2];                                        // Current music loaded, up to two can play at the same time
+static Music currentMusic[MAX_MUSIC_STREAMS];                        // Current music loaded, up to two can play at the same time
 
 //----------------------------------------------------------------------------------
 // Module specific Functions Declaration
@@ -797,18 +797,18 @@ void PlayMusicStream(char *fileName)
             TraceLog(INFO, "[%s] Ogg channels: %i", fileName, info.channels);
             TraceLog(DEBUG, "[%s] Temp memory required: %i", fileName, info.temp_memory_required);
 
+            currentMusic[musicIndex].loop = true;                  // We loop by default
+            musicEnabled = true;
+
+            currentMusic[musicIndex].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[musicIndex].stream) * info.channels;
+            currentMusic[musicIndex].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[musicIndex].stream);
+            
             if (info.channels == 2){
                 currentMusic[musicIndex].ctx = InitAudioContext(info.sample_rate, mixIndex, 2, false);
             }
             else{
                 currentMusic[musicIndex].ctx = InitAudioContext(info.sample_rate, mixIndex, 1, false);
             }
-
-            currentMusic[musicIndex].loop = true;                  // We loop by default
-            musicEnabled = true;
-
-            currentMusic[musicIndex].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[musicIndex].stream) * currentMusic[musicIndex].channels;
-            currentMusic[musicIndex].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[musicIndex].stream);
         }
     }
     else if (strcmp(GetExtension(fileName),"xm") == 0)
@@ -820,7 +820,7 @@ void PlayMusicStream(char *fileName)
             currentMusic[musicIndex].loop = true;
             jar_xm_set_max_loop_count(currentMusic[musicIndex].chipctx, 0); // infinite number of loops
             currentMusic[musicIndex].totalSamplesLeft =  jar_xm_get_remaining_samples(currentMusic[musicIndex].chipctx);
-            currentMusic[musicIndex].totalLengthSeconds = ((float)currentMusic[musicIndex].totalSamplesLeft) / ((float)currentMusic[musicIndex].sampleRate);
+            currentMusic[musicIndex].totalLengthSeconds = ((float)currentMusic[musicIndex].totalSamplesLeft) / 48000.f;
             musicEnabled = true;
             
             TraceLog(INFO, "[%s] XM number of samples: %i", fileName, currentMusic[musicIndex].totalSamplesLeft);
@@ -891,18 +891,15 @@ void ResumeMusicStream(int index)
     }
 }
 
-// Check if music is playing
-bool IsMusicPlaying(void)
+// Check if any music is playing
+bool IsMusicPlaying(int index)
 {
     bool playing = false;
     ALint state;
     
-    for(int musicIndex = 0; musicIndex < MAX_MUSIC_STREAMS; musicIndex++)
-    {
-        if(currentMusic[musicIndex].ctx){
-            alGetSourcei(currentMusic[musicIndex].ctx->alSource, AL_SOURCE_STATE, &state);
-            if (state == AL_PLAYING) playing = true;
-        }
+    if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx){
+        alGetSourcei(currentMusic[index].ctx->alSource, AL_SOURCE_STATE, &state);
+        if (state == AL_PLAYING) playing = true;
     }
 
     return playing;
@@ -911,8 +908,15 @@ bool IsMusicPlaying(void)
 // Set volume for music
 void SetMusicVolume(int index, float volume)
 {
-    if(currentMusic[musicIndex].ctx){
-        alSourcef(currentMusic[musicIndex].ctx->alSource, AL_GAIN, volume);
+    if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx){
+        alSourcef(currentMusic[index].ctx->alSource, AL_GAIN, volume);
+    }
+}
+
+void SetMusicPitch(int index, float pitch)
+{
+    if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx){
+        alSourcef(currentMusic[index].ctx->alSource, AL_PITCH, pitch);
     }
 }
 
diff --git a/src/audio.h b/src/audio.h
index a1602bd9..fef85b4f 100644
--- a/src/audio.h
+++ b/src/audio.h
@@ -105,11 +105,12 @@ void UpdateMusicStream(void);                                   // Updates buffe
 void StopMusicStream(int index);                                // Stop music playing (close stream)
 void PauseMusicStream(int index);                               // Pause music playing
 void ResumeMusicStream(int index);                              // Resume playing paused music
-bool IsMusicPlaying(void);                                      // Check if music is playing
+bool IsMusicPlaying(int index);                                 // Check if music is playing
 void SetMusicVolume(int index, float volume);                   // Set volume for music (1.0 is max level)
 float GetMusicTimeLength(void);                                 // Get music time length (in seconds)
 float GetMusicTimePlayed(void);                                 // Get current music time played (in seconds)
 int getMusicStreamCount(void);
+void SetMusicPitch(int index, float pitch);
 
 #ifdef __cplusplus
 }
diff --git a/src/raylib.h b/src/raylib.h
index 0c49a085..b9390d31 100644
--- a/src/raylib.h
+++ b/src/raylib.h
@@ -896,11 +896,12 @@ void UpdateMusicStream(void);                                   // Updates buffe
 void StopMusicStream(int index);                                // Stop music playing (close stream)
 void PauseMusicStream(int index);                               // Pause music playing
 void ResumeMusicStream(int index);                              // Resume playing paused music
-bool IsMusicPlaying(void);                                      // Check if music is playing
+bool IsMusicPlaying(int index);                                 // Check if music is playing
 void SetMusicVolume(int index, float volume);                   // Set volume for music (1.0 is max level)
 float GetMusicTimeLength(void);                                 // Get current music time length (in seconds)
 float GetMusicTimePlayed(void);                                 // Get current music time played (in seconds)
 int getMusicStreamCount(void);
+void SetMusicPitch(int index, float pitch);
 
 #ifdef __cplusplus
 }

From 9737c58054d5d0cc636fca0c998b31a69d501970 Mon Sep 17 00:00:00 2001
From: Joshua Reisenauer <kd7tck@msn.com>
Date: Wed, 11 May 2016 20:15:37 -0700
Subject: [PATCH 03/14] PlayMusicStream now uses index

---
 src/audio.c  | 29 +++++++++++++++++++----------
 src/audio.h  |  2 +-
 src/raylib.h |  2 +-
 3 files changed, 21 insertions(+), 12 deletions(-)

diff --git a/src/audio.c b/src/audio.c
index f89be8bf..6a8d251e 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -763,19 +763,18 @@ void SetSoundPitch(Sound sound, float pitch)
 //----------------------------------------------------------------------------------
 
 // Start music playing (open stream)
-void PlayMusicStream(char *fileName)
+// returns 0 on success
+int PlayMusicStream(int musicIndex, char *fileName)
 {
-    int musicIndex;
+    int musicIndex = index;
     int mixIndex;
-    for(musicIndex = 0; musicIndex < MAX_MUSIC_STREAMS; musicIndex++) // find empty music slot
-    {
-        if(currentMusic[musicIndex].stream == NULL && !currentMusic[musicIndex].chipTune) break;
-        else if(musicIndex = MAX_MUSIC_STREAMS - 1) return;
-    }
+    
+    if(currentMusic[musicIndex].stream != NULL || currentMusic[musicIndex].chipTune) return 1; // error
+    
     for(mixIndex = 0; mixIndex < MAX_AUDIO_CONTEXTS; mixIndex++) // find empty mix channel slot
     {
         if(mixChannelsActive_g[mixIndex] == NULL) break;
-        else if(musicIndex = MAX_AUDIO_CONTEXTS - 1) return;
+        else if(musicIndex = MAX_AUDIO_CONTEXTS - 1) return 2; // error
     }
     
     
@@ -787,6 +786,7 @@ void PlayMusicStream(char *fileName)
         if (currentMusic[musicIndex].stream == NULL)
         {
             TraceLog(WARNING, "[%s] OGG audio file could not be opened", fileName);
+            return 3; // error
         }
         else
         {
@@ -828,9 +828,18 @@ void PlayMusicStream(char *fileName)
             
             currentMusic[musicIndex].ctx = InitAudioContext(48000, mixIndex, 2, true);
         }
-        else TraceLog(WARNING, "[%s] XM file could not be opened", fileName);
+        else
+        {
+            TraceLog(WARNING, "[%s] XM file could not be opened", fileName);
+            return 4; // error
+        }
+    }
+    else
+    {
+        TraceLog(WARNING, "[%s] Music extension not recognized, it can't be loaded", fileName);
+        return 5; // error
     }
-    else TraceLog(WARNING, "[%s] Music extension not recognized, it can't be loaded", fileName);
+    return 0; // normal return
 }
 
 // Stop music playing for individual music index of currentMusic array (close stream)
diff --git a/src/audio.h b/src/audio.h
index fef85b4f..d09c4acc 100644
--- a/src/audio.h
+++ b/src/audio.h
@@ -100,7 +100,7 @@ bool IsSoundPlaying(Sound sound);                               // Check if a so
 void SetSoundVolume(Sound sound, float volume);                 // Set volume for a sound (1.0 is max level)
 void SetSoundPitch(Sound sound, float pitch);                   // Set pitch for a sound (1.0 is base level)
 
-void PlayMusicStream(char *fileName);                           // Start music playing (open stream)
+int PlayMusicStream(int musicIndex, char *fileName);            // Start music playing (open stream)
 void UpdateMusicStream(void);                                   // Updates buffers for music streaming
 void StopMusicStream(int index);                                // Stop music playing (close stream)
 void PauseMusicStream(int index);                               // Pause music playing
diff --git a/src/raylib.h b/src/raylib.h
index a6507906..cb17aa78 100644
--- a/src/raylib.h
+++ b/src/raylib.h
@@ -894,7 +894,7 @@ bool IsSoundPlaying(Sound sound);                               // Check if a so
 void SetSoundVolume(Sound sound, float volume);                 // Set volume for a sound (1.0 is max level)
 void SetSoundPitch(Sound sound, float pitch);                   // Set pitch for a sound (1.0 is base level)
 
-void PlayMusicStream(char *fileName);                           // Start music playing (open stream)
+int PlayMusicStream(int musicIndex, char *fileName);            // Start music playing (open stream)
 void UpdateMusicStream(void);                                   // Updates buffers for music streaming
 void StopMusicStream(int index);                                // Stop music playing (close stream)
 void PauseMusicStream(int index);                               // Pause music playing

From f0ada8c40d7887654d05c62f980a0a5473c1d8a7 Mon Sep 17 00:00:00 2001
From: Joshua Reisenauer <kd7tck@msn.com>
Date: Wed, 11 May 2016 22:37:53 -0700
Subject: [PATCH 04/14] apply index to remaining functions

---
 src/audio.c  | 108 ++++++++++++++++++++++++++-------------------------
 src/audio.h  |   6 +--
 src/raylib.h |   6 +--
 3 files changed, 62 insertions(+), 58 deletions(-)

diff --git a/src/audio.c b/src/audio.c
index 6a8d251e..94729566 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -118,12 +118,12 @@ static Music currentMusic[MAX_MUSIC_STREAMS];                        // Current
 //----------------------------------------------------------------------------------
 // Module specific Functions Declaration
 //----------------------------------------------------------------------------------
-static Wave LoadWAV(const char *fileName);          // Load WAV file
-static Wave LoadOGG(char *fileName);                // Load OGG file
-static void UnloadWave(Wave wave);                  // Unload wave data
+static Wave LoadWAV(const char *fileName);                // Load WAV file
+static Wave LoadOGG(char *fileName);                      // Load OGG file
+static void UnloadWave(Wave wave);                        // Unload wave data
 
-static bool BufferMusicStream(ALuint buffer);       // Fill music buffers with data
-static void EmptyMusicStream(void);                 // Empty music buffers
+static bool BufferMusicStream(int index, ALuint buffer);  // Fill music buffers with data
+static void EmptyMusicStream(int index);                  // Empty music buffers
 
 static unsigned short FillAlBufferWithSilence(AudioContext_t *context, 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
@@ -766,7 +766,6 @@ void SetSoundPitch(Sound sound, float pitch)
 // returns 0 on success
 int PlayMusicStream(int musicIndex, char *fileName)
 {
-    int musicIndex = index;
     int mixIndex;
     
     if(currentMusic[musicIndex].stream != NULL || currentMusic[musicIndex].chipTune) return 1; // error
@@ -930,36 +929,36 @@ void SetMusicPitch(int index, float pitch)
 }
 
 // Get current music time length (in seconds)
-float GetMusicTimeLength(void)
+float GetMusicTimeLength(int index)
 {
     float totalSeconds;
-    if (currentMusic.chipTune)
+    if (currentMusic[index].chipTune)
     {
-        totalSeconds = currentMusic.totalLengthSeconds;
+        totalSeconds = currentMusic[index].totalLengthSeconds;
     }
     else
     {
-        totalSeconds = stb_vorbis_stream_length_in_seconds(currentMusic.stream);
+        totalSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[index].stream);
     }
 
     return totalSeconds;
 }
 
 // Get current music time played (in seconds)
-float GetMusicTimePlayed(void)
+float GetMusicTimePlayed(int index)
 {
     float secondsPlayed;
-    if (currentMusic.chipTune)
+    if (currentMusic[index].chipTune)
     {
         uint64_t samples;
-        jar_xm_get_position(currentMusic.chipctx, NULL, NULL, NULL, &samples);
-        secondsPlayed = (float)samples / (currentMusic.sampleRate * currentMusic.channels); // Not sure if this is the correct value
+        jar_xm_get_position(currentMusic[index].chipctx, NULL, NULL, NULL, &samples);
+        secondsPlayed = (float)samples / (48000 * currentMusic[index].ctx->channels); // Not sure if this is the correct value
     }
     else
     {
-        int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
-        int samplesPlayed = totalSamples - currentMusic.totalSamplesLeft;
-        secondsPlayed = (float)samplesPlayed / (currentMusic.sampleRate * currentMusic.channels);
+        int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].ctx->channels;
+        int samplesPlayed = totalSamples - currentMusic[index].totalSamplesLeft;
+        secondsPlayed = (float)samplesPlayed / (currentMusic[index].ctx->sampleRate * currentMusic[index].ctx->channels);
     }
     
 
@@ -971,9 +970,10 @@ float GetMusicTimePlayed(void)
 //----------------------------------------------------------------------------------
 
 // Fill music buffers with new data from music stream
-static bool BufferMusicStream(ALuint buffer)
+static bool BufferMusicStream(int index, ALuint buffer)
 {
     short pcm[MUSIC_BUFFER_SIZE_SHORT];
+    float pcmf[MUSIC_BUFFER_SIZE_FLOAT];
     
     int  size = 0;              // Total size of data steamed (in bytes)
     int  streamedBytes = 0;     // samples of data obtained, channels are not included in calculation
@@ -981,93 +981,97 @@ static bool BufferMusicStream(ALuint buffer)
 
     if (musicEnabled)
     {
-        if (currentMusic.chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
+        if (currentMusic[index].chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
         {
-            int readlen = MUSIC_BUFFER_SIZE_SHORT / 2;
-            jar_xm_generate_samples_16bit(currentMusic.chipctx, pcm, readlen); // reads 2*readlen shorts and moves them to buffer+size memory location
-            size += readlen * currentMusic.channels; // Not sure if this is what it needs
+            int readlen = MUSIC_BUFFER_SIZE_FLOAT / 2;
+            jar_xm_generate_samples(currentMusic[index].chipctx, pcmf, readlen); // reads 2*readlen shorts and moves them to buffer+size memory location
+            size += readlen * currentMusic[index].ctx->channels; // Not sure if this is what it needs
+            
+            alBufferData(buffer, currentMusic[index].ctx->alFormat, pcmf, size*sizeof(float), 48000);
+            currentMusic[index].totalSamplesLeft -= size;
+            if(currentMusic[index].totalSamplesLeft <= 0) active = false;
         }
         else
         {
             while (size < MUSIC_BUFFER_SIZE_SHORT)
             {
-                streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic.stream, currentMusic.channels, pcm + size, MUSIC_BUFFER_SIZE_SHORT - size);
-                if (streamedBytes > 0) size += (streamedBytes*currentMusic.channels);
+                streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic[index].stream, currentMusic[index].ctx->channels, pcm + size, MUSIC_BUFFER_SIZE_SHORT - size);
+                if (streamedBytes > 0) size += (streamedBytes*currentMusic[index].ctx->channels);
                 else break;
             }
+            
+            if (size > 0)
+            {
+                alBufferData(buffer, currentMusic[index].ctx->alFormat, pcm, size*sizeof(short), currentMusic[index].ctx->sampleRate);
+                currentMusic[index].totalSamplesLeft -= size;
+                
+                if(currentMusic[index].totalSamplesLeft <= 0) active = false; // end if no more samples left
+            }
+            else
+            {
+                active = false;
+                TraceLog(WARNING, "No more data obtained from stream");
+            }
         }
         TraceLog(DEBUG, "Streaming music data to buffer. Bytes streamed: %i", size);
     }
 
-    if (size > 0)
-    {
-        alBufferData(buffer, currentMusic.format, pcm, size*sizeof(short), currentMusic.sampleRate);
-        currentMusic.totalSamplesLeft -= size;
-        
-        if(currentMusic.totalSamplesLeft <= 0) active = false; // end if no more samples left
-    }
-    else
-    {
-        active = false;
-        TraceLog(WARNING, "No more data obtained from stream");
-    }
-
     return active;
 }
 
 // Empty music buffers
-static void EmptyMusicStream(void)
+static void EmptyMusicStream(int index)
 {
     ALuint buffer = 0;
     int queued = 0;
 
-    alGetSourcei(currentMusic.source, AL_BUFFERS_QUEUED, &queued);
+    alGetSourcei(currentMusic[index].source, AL_BUFFERS_QUEUED, &queued);
 
     while (queued > 0)
     {
-        alSourceUnqueueBuffers(currentMusic.source, 1, &buffer);
+        alSourceUnqueueBuffers(currentMusic[index].source, 1, &buffer);
 
         queued--;
     }
 }
 
 // Update (re-fill) music buffers if data already processed
-void UpdateMusicStream(void)
+void UpdateMusicStream(int index)
 {
     ALuint buffer = 0;
     ALint processed = 0;
     bool active = true;
 
-    if (musicEnabled)
+    if (index < MAX_MUSIC_STREAMS && musicEnabled)
     {
         // Get the number of already processed buffers (if any)
-        alGetSourcei(currentMusic.source, AL_BUFFERS_PROCESSED, &processed);
+        alGetSourcei(currentMusic[index].source, AL_BUFFERS_PROCESSED, &processed);
 
         while (processed > 0)
         {
             // Recover processed buffer for refill
-            alSourceUnqueueBuffers(currentMusic.source, 1, &buffer);
+            alSourceUnqueueBuffers(currentMusic[index].source, 1, &buffer);
 
             // Refill buffer
             active = BufferMusicStream(buffer);
 
             // If no more data to stream, restart music (if loop)
-            if ((!active) && (currentMusic.loop))
+            if ((!active) && (currentMusic[index].loop))
             {
-                if(currentMusic.chipTune)
+                if(currentMusic[index].chipTune)
                 {
-                    currentMusic.totalSamplesLeft = currentMusic.totalLengthSeconds * currentMusic.sampleRate;
+                    currentMusic[index].totalSamplesLeft = currentMusic[index].totalLengthSeconds * currentMusic[index].ctx->sampleRate;
                 }
                 else
                 {
-                    stb_vorbis_seek_start(currentMusic.stream);
-                    currentMusic.totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic.stream)*currentMusic.channels;
+                    stb_vorbis_seek_start(currentMusic[index].stream);
+                    currentMusic[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[index].stream)*currentMusic[index].ctx->channels;
                 }
                 active = BufferMusicStream(buffer);
             }
 
             // Add refilled buffer to queue again... don't let the music stop!
-            alSourceQueueBuffers(currentMusic.source, 1, &buffer);
+            alSourceQueueBuffers(currentMusic[index].source, 1, &buffer);
 
             if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data...");
 
@@ -1075,9 +1079,9 @@ void UpdateMusicStream(void)
         }
 
         ALenum state;
-        alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state);
+        alGetSourcei(currentMusic[index].source, AL_SOURCE_STATE, &state);
 
-        if ((state != AL_PLAYING) && active) alSourcePlay(currentMusic.source);
+        if ((state != AL_PLAYING) && active) alSourcePlay(currentMusic[index].source);
 
         if (!active) StopMusicStream();
     }
diff --git a/src/audio.h b/src/audio.h
index d09c4acc..63c1c136 100644
--- a/src/audio.h
+++ b/src/audio.h
@@ -101,14 +101,14 @@ void SetSoundVolume(Sound sound, float volume);                 // Set volume fo
 void SetSoundPitch(Sound sound, float pitch);                   // Set pitch for a sound (1.0 is base level)
 
 int PlayMusicStream(int musicIndex, char *fileName);            // Start music playing (open stream)
-void UpdateMusicStream(void);                                   // Updates buffers for music streaming
+void UpdateMusicStream(int index);                              // Updates buffers for music streaming
 void StopMusicStream(int index);                                // Stop music playing (close stream)
 void PauseMusicStream(int index);                               // Pause music playing
 void ResumeMusicStream(int index);                              // Resume playing paused music
 bool IsMusicPlaying(int index);                                 // Check if music is playing
 void SetMusicVolume(int index, float volume);                   // Set volume for music (1.0 is max level)
-float GetMusicTimeLength(void);                                 // Get music time length (in seconds)
-float GetMusicTimePlayed(void);                                 // Get current music time played (in seconds)
+float GetMusicTimeLength(int index);                            // Get music time length (in seconds)
+float GetMusicTimePlayed(int index);                            // Get current music time played (in seconds)
 int getMusicStreamCount(void);
 void SetMusicPitch(int index, float pitch);
 
diff --git a/src/raylib.h b/src/raylib.h
index cb17aa78..05c945f7 100644
--- a/src/raylib.h
+++ b/src/raylib.h
@@ -895,14 +895,14 @@ void SetSoundVolume(Sound sound, float volume);                 // Set volume fo
 void SetSoundPitch(Sound sound, float pitch);                   // Set pitch for a sound (1.0 is base level)
 
 int PlayMusicStream(int musicIndex, char *fileName);            // Start music playing (open stream)
-void UpdateMusicStream(void);                                   // Updates buffers for music streaming
+void UpdateMusicStream(int index);                              // Updates buffers for music streaming
 void StopMusicStream(int index);                                // Stop music playing (close stream)
 void PauseMusicStream(int index);                               // Pause music playing
 void ResumeMusicStream(int index);                              // Resume playing paused music
 bool IsMusicPlaying(int index);                                 // Check if music is playing
 void SetMusicVolume(int index, float volume);                   // Set volume for music (1.0 is max level)
-float GetMusicTimeLength(void);                                 // Get current music time length (in seconds)
-float GetMusicTimePlayed(void);                                 // Get current music time played (in seconds)
+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);
 void SetMusicPitch(int index, float pitch);
 

From 83dbc076507b7025aa0a1315e409ff46f75c1e0b Mon Sep 17 00:00:00 2001
From: Joshua Reisenauer <kd7tck@msn.com>
Date: Thu, 12 May 2016 16:02:23 -0700
Subject: [PATCH 05/14] buffering of music now uses update audio context

---
 src/audio.c   | 94 ++++++++++++++++++---------------------------------
 src/easings.h | 10 +++---
 2 files changed, 37 insertions(+), 67 deletions(-)

diff --git a/src/audio.c b/src/audio.c
index 94729566..cc964cd1 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -118,12 +118,12 @@ static Music currentMusic[MAX_MUSIC_STREAMS];                        // Current
 //----------------------------------------------------------------------------------
 // Module specific Functions Declaration
 //----------------------------------------------------------------------------------
-static Wave LoadWAV(const char *fileName);                // Load WAV file
-static Wave LoadOGG(char *fileName);                      // Load OGG file
-static void UnloadWave(Wave wave);                        // Unload wave data
+static Wave LoadWAV(const char *fileName);         // Load WAV file
+static Wave LoadOGG(char *fileName);               // Load OGG file
+static void UnloadWave(Wave wave);                 // Unload wave data
 
-static bool BufferMusicStream(int index, ALuint buffer);  // Fill music buffers with data
-static void EmptyMusicStream(int index);                  // Empty music buffers
+static bool BufferMusicStream(int index);          // Fill music buffers with data
+static void EmptyMusicStream(int index);           // Empty music buffers
 
 static unsigned short FillAlBufferWithSilence(AudioContext_t *context, 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
@@ -970,7 +970,7 @@ float GetMusicTimePlayed(int index)
 //----------------------------------------------------------------------------------
 
 // Fill music buffers with new data from music stream
-static bool BufferMusicStream(int index, ALuint buffer)
+static bool BufferMusicStream(int index)
 {
     short pcm[MUSIC_BUFFER_SIZE_SHORT];
     float pcmf[MUSIC_BUFFER_SIZE_FLOAT];
@@ -985,33 +985,17 @@ static bool BufferMusicStream(int index, ALuint buffer)
         {
             int readlen = MUSIC_BUFFER_SIZE_FLOAT / 2;
             jar_xm_generate_samples(currentMusic[index].chipctx, pcmf, readlen); // reads 2*readlen shorts and moves them to buffer+size memory location
+            UpdateAudioContext(currentMusic[index].ctx, pcmf, MUSIC_BUFFER_SIZE_FLOAT);
             size += readlen * currentMusic[index].ctx->channels; // Not sure if this is what it needs
-            
-            alBufferData(buffer, currentMusic[index].ctx->alFormat, pcmf, size*sizeof(float), 48000);
             currentMusic[index].totalSamplesLeft -= size;
             if(currentMusic[index].totalSamplesLeft <= 0) active = false;
         }
         else
         {
-            while (size < MUSIC_BUFFER_SIZE_SHORT)
-            {
-                streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic[index].stream, currentMusic[index].ctx->channels, pcm + size, MUSIC_BUFFER_SIZE_SHORT - size);
-                if (streamedBytes > 0) size += (streamedBytes*currentMusic[index].ctx->channels);
-                else break;
-            }
-            
-            if (size > 0)
-            {
-                alBufferData(buffer, currentMusic[index].ctx->alFormat, pcm, size*sizeof(short), currentMusic[index].ctx->sampleRate);
-                currentMusic[index].totalSamplesLeft -= size;
-                
-                if(currentMusic[index].totalSamplesLeft <= 0) active = false; // end if no more samples left
-            }
-            else
-            {
-                active = false;
-                TraceLog(WARNING, "No more data obtained from stream");
-            }
+            streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic[index].stream, currentMusic[index].ctx->channels, pcm, MUSIC_BUFFER_SIZE_SHORT);
+            UpdateAudioContext(currentMusic[index].ctx, pcm, MUSIC_BUFFER_SIZE_SHORT);
+            currentMusic[index].totalSamplesLeft -= MUSIC_BUFFER_SIZE_SHORT;
+            if(currentMusic[index].totalSamplesLeft <= 0) active = false;
         }
         TraceLog(DEBUG, "Streaming music data to buffer. Bytes streamed: %i", size);
     }
@@ -1038,53 +1022,39 @@ static void EmptyMusicStream(int index)
 // Update (re-fill) music buffers if data already processed
 void UpdateMusicStream(int index)
 {
-    ALuint buffer = 0;
-    ALint processed = 0;
+    ALenum state;
     bool active = true;
 
     if (index < MAX_MUSIC_STREAMS && musicEnabled)
     {
-        // Get the number of already processed buffers (if any)
-        alGetSourcei(currentMusic[index].source, AL_BUFFERS_PROCESSED, &processed);
-
-        while (processed > 0)
+        active = BufferMusicStream(index);
+        
+        if ((!active) && (currentMusic[index].loop))
         {
-            // Recover processed buffer for refill
-            alSourceUnqueueBuffers(currentMusic[index].source, 1, &buffer);
-
-            // Refill buffer
-            active = BufferMusicStream(buffer);
-
-            // If no more data to stream, restart music (if loop)
-            if ((!active) && (currentMusic[index].loop))
+            if(currentMusic[index].chipTune)
             {
-                if(currentMusic[index].chipTune)
-                {
-                    currentMusic[index].totalSamplesLeft = currentMusic[index].totalLengthSeconds * currentMusic[index].ctx->sampleRate;
-                }
-                else
-                {
-                    stb_vorbis_seek_start(currentMusic[index].stream);
-                    currentMusic[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[index].stream)*currentMusic[index].ctx->channels;
-                }
-                active = BufferMusicStream(buffer);
+                currentMusic[index].totalSamplesLeft = currentMusic[index].totalLengthSeconds * currentMusic[index].ctx->sampleRate;
             }
+            else
+            {
+                stb_vorbis_seek_start(currentMusic[index].stream);
+                currentMusic[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[index].stream)*currentMusic[index].ctx->channels;
+            }
+            active = BufferMusicStream(index);
+        }
+        
 
-            // Add refilled buffer to queue again... don't let the music stop!
-            alSourceQueueBuffers(currentMusic[index].source, 1, &buffer);
-
-            if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data...");
+        if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data...");
 
-            processed--;
-        }
+        processed--;
+    }
 
-        ALenum state;
-        alGetSourcei(currentMusic[index].source, AL_SOURCE_STATE, &state);
+    
+    alGetSourcei(currentMusic[index].source, AL_SOURCE_STATE, &state);
 
-        if ((state != AL_PLAYING) && active) alSourcePlay(currentMusic[index].source);
+    if ((state != AL_PLAYING) && active) alSourcePlay(currentMusic[index].source);
 
-        if (!active) StopMusicStream();
-    }
+    if (!active) StopMusicStream();
 }
 
 // Load WAV file into Wave structure
diff --git a/src/easings.h b/src/easings.h
index e1e5465a..a8178f4a 100644
--- a/src/easings.h
+++ b/src/easings.h
@@ -18,11 +18,11 @@
 *   float speed = 1.f;
 *   float currentTime = 0.f;
 *   float currentPos[2] = {0,0};
-*   float newPos[2] = {1,1};
-*   float tempPosition[2] = currentPos;//x,y positions
-*   while(currentPos[0] < newPos[0])
-*       currentPos[0] = EaseSineIn(currentTime, tempPosition[0], tempPosition[0]-newPos[0], speed);
-*       currentPos[1] = EaseSineIn(currentTime, tempPosition[1], tempPosition[1]-newPos[0], speed);
+*   float finalPos[2] = {1,1};
+*   float startPosition[2] = currentPos;//x,y positions
+*   while(currentPos[0] < finalPos[0])
+*       currentPos[0] = EaseSineIn(currentTime, startPosition[0], startPosition[0]-finalPos[0], speed);
+*       currentPos[1] = EaseSineIn(currentTime, startPosition[1], startPosition[1]-finalPos[0], speed);
 *       currentTime += diffTime();
 *
 *   A port of Robert Penner's easing equations to C (http://robertpenner.com/easing/)

From 5107a2dc40330623f61205ee70f067d24afb0af8 Mon Sep 17 00:00:00 2001
From: Joshua Reisenauer <kd7tck@msn.com>
Date: Thu, 12 May 2016 21:14:02 -0700
Subject: [PATCH 06/14] bug fixes

---
 src/audio.c | 69 +++++++++++++++++++++++++++--------------------------
 1 file changed, 35 insertions(+), 34 deletions(-)

diff --git a/src/audio.c b/src/audio.c
index cc964cd1..d19d3306 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -77,19 +77,6 @@
 // Types and Structures Definition
 //----------------------------------------------------------------------------------
 
-// Music type (file streaming from memory)
-// NOTE: Anything longer than ~10 seconds should be streamed...
-typedef struct Music {
-    stb_vorbis *stream;
-    jar_xm_context_t *chipctx; // Stores jar_xm context
-    AudioContext_t *ctx; // audio context
-    
-    int totalSamplesLeft;
-    float totalLengthSeconds;
-    bool loop;
-    bool chipTune; // True if chiptune is loaded
-} Music;
-
 // Audio Context, used to create custom audio streams that are not bound to a sound file. There can be
 // no more than 4 concurrent audio contexts in use. This is due to each active context being tied to
 // a dedicated mix channel. All audio is 32bit floating point in stereo.
@@ -104,6 +91,19 @@ typedef struct AudioContext_t {
     ALuint alBuffer[MAX_STREAM_BUFFERS]; // openAL sample buffer
 } AudioContext_t;
 
+// Music type (file streaming from memory)
+// NOTE: Anything longer than ~10 seconds should be streamed...
+typedef struct Music {
+    stb_vorbis *stream;
+    jar_xm_context_t *chipctx; // Stores jar_xm context
+    AudioContext_t *ctx; // audio context
+    
+    int totalSamplesLeft;
+    float totalLengthSeconds;
+    bool loop;
+    bool chipTune; // True if chiptune is loaded
+} Music;
+
 #if defined(AUDIO_STANDALONE)
 typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType;
 #endif
@@ -168,7 +168,10 @@ void InitAudioDevice(void)
 // Close the audio device for the current context, and destroys the context
 void CloseAudioDevice(void)
 {
-    StopMusicStream();      // Stop music streaming and close current stream
+    for(int index=0; index<MAX_MUSIC_STREAMS; index++)
+    {
+        if(currentMusic[index].ctx) StopMusicStream(index);      // Stop music streaming and close current stream
+    }
 
     ALCdevice *device;
     ALCcontext *context = alcGetCurrentContext();
@@ -205,7 +208,6 @@ AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChanne
 {
     if(mixChannel >= MAX_AUDIO_CONTEXTS) return NULL;
     if(!IsAudioDeviceReady()) InitAudioDevice();
-    else StopMusicStream();
     
     if(!mixChannelsActive_g[mixChannel]){
         AudioContext_t *ac = (AudioContext_t*)malloc(sizeof(AudioContext_t));
@@ -776,7 +778,6 @@ int PlayMusicStream(int musicIndex, char *fileName)
         else if(musicIndex = MAX_AUDIO_CONTEXTS - 1) return 2; // error
     }
     
-    
     if (strcmp(GetExtension(fileName),"ogg") == 0)
     {
         // Open audio stream
@@ -888,12 +889,12 @@ void ResumeMusicStream(int index)
 {
     // Resume music playing... if music available!
     ALenum state;
-    if(currentMusic[musicIndex].ctx){
-        alGetSourcei(currentMusic[musicIndex].ctx->alSource, AL_SOURCE_STATE, &state);
+    if(currentMusic[index].ctx){
+        alGetSourcei(currentMusic[index].ctx->alSource, AL_SOURCE_STATE, &state);
         if (state == AL_PAUSED)
         {
             TraceLog(INFO, "Resuming music stream");
-            alSourcePlay(currentMusic[musicIndex].ctx->alSource);
+            alSourcePlay(currentMusic[index].ctx->alSource);
             musicEnabled = true;
         }
     }
@@ -976,9 +977,8 @@ static bool BufferMusicStream(int index)
     float pcmf[MUSIC_BUFFER_SIZE_FLOAT];
     
     int  size = 0;              // Total size of data steamed (in bytes)
-    int  streamedBytes = 0;     // samples of data obtained, channels are not included in calculation
     bool active = true;         // We can get more data from stream (not finished)
-
+        
     if (musicEnabled)
     {
         if (currentMusic[index].chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
@@ -992,12 +992,12 @@ static bool BufferMusicStream(int index)
         }
         else
         {
-            streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic[index].stream, currentMusic[index].ctx->channels, pcm, MUSIC_BUFFER_SIZE_SHORT);
-            UpdateAudioContext(currentMusic[index].ctx, pcm, MUSIC_BUFFER_SIZE_SHORT);
-            currentMusic[index].totalSamplesLeft -= MUSIC_BUFFER_SIZE_SHORT;
+            int streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic[index].stream, currentMusic[index].ctx->channels, pcm, MUSIC_BUFFER_SIZE_SHORT);
+            UpdateAudioContext(currentMusic[index].ctx, pcm, streamedBytes);
+            currentMusic[index].totalSamplesLeft -= streamedBytes*currentMusic[index].ctx->channels;
             if(currentMusic[index].totalSamplesLeft <= 0) active = false;
         }
-        TraceLog(DEBUG, "Streaming music data to buffer. Bytes streamed: %i", size);
+        // TraceLog(DEBUG, "Streaming music data to buffer. Bytes streamed: %i", size);
     }
 
     return active;
@@ -1009,11 +1009,11 @@ static void EmptyMusicStream(int index)
     ALuint buffer = 0;
     int queued = 0;
 
-    alGetSourcei(currentMusic[index].source, AL_BUFFERS_QUEUED, &queued);
+    alGetSourcei(currentMusic[index].ctx->alSource, AL_BUFFERS_QUEUED, &queued);
 
     while (queued > 0)
     {
-        alSourceUnqueueBuffers(currentMusic[index].source, 1, &buffer);
+        alSourceUnqueueBuffers(currentMusic[index].ctx->alSource, 1, &buffer);
 
         queued--;
     }
@@ -1045,16 +1045,17 @@ void UpdateMusicStream(int index)
         
 
         if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data...");
+        
+        alGetSourcei(currentMusic[index].ctx->alSource, AL_SOURCE_STATE, &state);
 
-        processed--;
-    }
-
-    
-    alGetSourcei(currentMusic[index].source, AL_SOURCE_STATE, &state);
+        if ((state != AL_PLAYING) && active) alSourcePlay(currentMusic[index].ctx->alSource);
 
-    if ((state != AL_PLAYING) && active) alSourcePlay(currentMusic[index].source);
+        if (!active) StopMusicStream(index);
+        
+    }
+    else
+        return;
 
-    if (!active) StopMusicStream();
 }
 
 // Load WAV file into Wave structure

From ea4b5552c2b4ff8a907dd1fefc084317187af168 Mon Sep 17 00:00:00 2001
From: Joshua Reisenauer <kd7tck@msn.com>
Date: Sat, 14 May 2016 00:25:40 -0700
Subject: [PATCH 07/14] corrected typos

---
 src/audio.c | 42 +++++++++++++++++++++++++++---------------
 1 file changed, 27 insertions(+), 15 deletions(-)

diff --git a/src/audio.c b/src/audio.c
index d19d3306..c53d6318 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -85,7 +85,7 @@ typedef struct AudioContext_t {
     unsigned char channels;              // 1=mono,2=stereo
     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;
+    bool playing;                        // false if paused
     ALenum alFormat;                     // openAL format specifier
     ALuint alSource;                     // openAL source
     ALuint alBuffer[MAX_STREAM_BUFFERS]; // openAL sample buffer
@@ -165,13 +165,14 @@ void InitAudioDevice(void)
     alListener3f(AL_ORIENTATION, 0, 0, -1);
 }
 
-// Close the audio device for the current context, and destroys the context
+// Close the audio device for all contexts
 void CloseAudioDevice(void)
 {
     for(int index=0; index<MAX_MUSIC_STREAMS; index++)
     {
         if(currentMusic[index].ctx) StopMusicStream(index);      // Stop music streaming and close current stream
     }
+    
 
     ALCdevice *device;
     ALCcontext *context = alcGetCurrentContext();
@@ -770,12 +771,12 @@ int PlayMusicStream(int musicIndex, char *fileName)
 {
     int mixIndex;
     
-    if(currentMusic[musicIndex].stream != NULL || currentMusic[musicIndex].chipTune) return 1; // error
+    if(currentMusic[musicIndex].stream || currentMusic[musicIndex].chipctx) return 1; // error
     
     for(mixIndex = 0; mixIndex < MAX_AUDIO_CONTEXTS; mixIndex++) // find empty mix channel slot
     {
         if(mixChannelsActive_g[mixIndex] == NULL) break;
-        else if(musicIndex = MAX_AUDIO_CONTEXTS - 1) return 2; // error
+        else if(mixIndex = MAX_AUDIO_CONTEXTS - 1) return 2; // error
     }
     
     if (strcmp(GetExtension(fileName),"ogg") == 0)
@@ -799,6 +800,7 @@ int PlayMusicStream(int musicIndex, char *fileName)
 
             currentMusic[musicIndex].loop = true;                  // We loop by default
             musicEnabled = true;
+            currentMusic[musicIndex].ctx->playing = true;
 
             currentMusic[musicIndex].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[musicIndex].stream) * info.channels;
             currentMusic[musicIndex].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[musicIndex].stream);
@@ -822,6 +824,7 @@ int PlayMusicStream(int musicIndex, char *fileName)
             currentMusic[musicIndex].totalSamplesLeft =  jar_xm_get_remaining_samples(currentMusic[musicIndex].chipctx);
             currentMusic[musicIndex].totalLengthSeconds = ((float)currentMusic[musicIndex].totalSamplesLeft) / 48000.f;
             musicEnabled = true;
+            currentMusic[musicIndex].ctx->playing = true;
             
             TraceLog(INFO, "[%s] XM number of samples: %i", fileName, currentMusic[musicIndex].totalSamplesLeft);
             TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, currentMusic[musicIndex].totalLengthSeconds);
@@ -859,16 +862,21 @@ void StopMusicStream(int index)
         }
         
         if(!getMusicStreamCount()) musicEnabled = false;
+        if(currentMusic[index].stream || currentMusic[index].chipctx)
+        {
+            currentMusic[index].stream = NULL;
+            currentMusic[index].chipctx = NULL;
+        }
     }
 }
 
 //get number of music channels active at this time, this does not mean they are playing
 int getMusicStreamCount(void)
 {
-    int musicCount;
+    int musicCount = 0;
     for(int musicIndex = 0; musicIndex < MAX_MUSIC_STREAMS; musicIndex++) // find empty music slot
         if(currentMusic[musicIndex].stream != NULL || currentMusic[musicIndex].chipTune) musicCount++;
-        
+    
     return musicCount;
 }
 
@@ -876,11 +884,11 @@ int getMusicStreamCount(void)
 void PauseMusicStream(int index)
 {
     // Pause music stream if music available!
-    if (musicEnabled)
+    if (currentMusic[index].ctx && musicEnabled)
     {
         TraceLog(INFO, "Pausing music stream");
-        UpdateAudioContext(currentMusic[index].ctx, NULL, 0); // pushing null data auto pauses stream
-        musicEnabled = false;
+        alSourcePause(currentMusic[index].ctx->alSource);
+        currentMusic[index].ctx->playing = false;
     }
 }
 
@@ -895,7 +903,7 @@ void ResumeMusicStream(int index)
         {
             TraceLog(INFO, "Resuming music stream");
             alSourcePlay(currentMusic[index].ctx->alSource);
-            musicEnabled = true;
+            currentMusic[index].ctx->playing = true;
         }
     }
 }
@@ -981,13 +989,17 @@ static bool BufferMusicStream(int index)
         
     if (musicEnabled)
     {
+        if(!currentMusic[index].ctx->playing)
+        {
+            UpdateAudioContext(currentMusic[index].ctx, NULL, 0);
+            return false;
+        }
+        
         if (currentMusic[index].chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
         {
-            int readlen = MUSIC_BUFFER_SIZE_FLOAT / 2;
-            jar_xm_generate_samples(currentMusic[index].chipctx, pcmf, readlen); // reads 2*readlen shorts and moves them to buffer+size memory location
+            jar_xm_generate_samples(currentMusic[index].chipctx, pcmf, MUSIC_BUFFER_SIZE_FLOAT / 2); // reads 2*readlen shorts and moves them to buffer+size memory location
             UpdateAudioContext(currentMusic[index].ctx, pcmf, MUSIC_BUFFER_SIZE_FLOAT);
-            size += readlen * currentMusic[index].ctx->channels; // Not sure if this is what it needs
-            currentMusic[index].totalSamplesLeft -= size;
+            currentMusic[index].totalSamplesLeft -= MUSIC_BUFFER_SIZE_FLOAT;
             if(currentMusic[index].totalSamplesLeft <= 0) active = false;
         }
         else
@@ -997,7 +1009,7 @@ static bool BufferMusicStream(int index)
             currentMusic[index].totalSamplesLeft -= streamedBytes*currentMusic[index].ctx->channels;
             if(currentMusic[index].totalSamplesLeft <= 0) active = false;
         }
-        // TraceLog(DEBUG, "Streaming music data to buffer. Bytes streamed: %i", size);
+        TraceLog(DEBUG, "Buffering index:%i, chiptune:%i", index, (int)currentMusic[index].chipTune);
     }
 
     return active;

From 8c5d403dda7a7070dc58204bb10539b3646f183b Mon Sep 17 00:00:00 2001
From: Joshua Reisenauer <kd7tck@msn.com>
Date: Sat, 14 May 2016 15:26:17 -0700
Subject: [PATCH 08/14] new function to check if music stream is ready

_g naming convention for globals, new error exit numbers.
---
 src/audio.c | 43 ++++++++++++++++++++++++++++++-------------
 1 file changed, 30 insertions(+), 13 deletions(-)

diff --git a/src/audio.c b/src/audio.c
index c53d6318..0fd1bb13 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -112,7 +112,7 @@ typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType;
 // Global Variables Definition
 //----------------------------------------------------------------------------------
 static AudioContext_t* mixChannelsActive_g[MAX_AUDIO_CONTEXTS];      // What mix channels are currently active
-static bool musicEnabled = false;
+static bool musicEnabled_g = false;
 static Music currentMusic[MAX_MUSIC_STREAMS];                        // Current music loaded, up to two can play at the same time
 
 //----------------------------------------------------------------------------------
@@ -126,8 +126,9 @@ static bool BufferMusicStream(int index);          // Fill music buffers with da
 static void EmptyMusicStream(int index);           // Empty music buffers
 
 static unsigned short FillAlBufferWithSilence(AudioContext_t *context, 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
+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
+static bool isMusicStreamReady(int index);                                            // Checks if music buffer is ready to be refilled
 
 #if defined(AUDIO_STANDALONE)
 const char *GetExtension(const char *fileName);     // Get the extension for a filename
@@ -799,18 +800,21 @@ int PlayMusicStream(int musicIndex, char *fileName)
             TraceLog(DEBUG, "[%s] Temp memory required: %i", fileName, info.temp_memory_required);
 
             currentMusic[musicIndex].loop = true;                  // We loop by default
-            musicEnabled = true;
-            currentMusic[musicIndex].ctx->playing = true;
+            musicEnabled_g = true;
+            
 
             currentMusic[musicIndex].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[musicIndex].stream) * info.channels;
             currentMusic[musicIndex].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[musicIndex].stream);
             
             if (info.channels == 2){
                 currentMusic[musicIndex].ctx = InitAudioContext(info.sample_rate, mixIndex, 2, false);
+                currentMusic[musicIndex].ctx->playing = true;
             }
             else{
                 currentMusic[musicIndex].ctx = InitAudioContext(info.sample_rate, mixIndex, 1, false);
+                currentMusic[musicIndex].ctx->playing = true;
             }
+            if(!currentMusic[musicIndex].ctx) return 4; // error
         }
     }
     else if (strcmp(GetExtension(fileName),"xm") == 0)
@@ -823,24 +827,25 @@ int PlayMusicStream(int musicIndex, char *fileName)
             jar_xm_set_max_loop_count(currentMusic[musicIndex].chipctx, 0); // infinite number of loops
             currentMusic[musicIndex].totalSamplesLeft =  jar_xm_get_remaining_samples(currentMusic[musicIndex].chipctx);
             currentMusic[musicIndex].totalLengthSeconds = ((float)currentMusic[musicIndex].totalSamplesLeft) / 48000.f;
-            musicEnabled = true;
-            currentMusic[musicIndex].ctx->playing = true;
+            musicEnabled_g = true;
             
             TraceLog(INFO, "[%s] XM number of samples: %i", fileName, currentMusic[musicIndex].totalSamplesLeft);
             TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, currentMusic[musicIndex].totalLengthSeconds);
             
             currentMusic[musicIndex].ctx = InitAudioContext(48000, mixIndex, 2, true);
+            if(!currentMusic[musicIndex].ctx) return 5; // error
+            currentMusic[musicIndex].ctx->playing = true;
         }
         else
         {
             TraceLog(WARNING, "[%s] XM file could not be opened", fileName);
-            return 4; // error
+            return 6; // error
         }
     }
     else
     {
         TraceLog(WARNING, "[%s] Music extension not recognized, it can't be loaded", fileName);
-        return 5; // error
+        return 7; // error
     }
     return 0; // normal return
 }
@@ -861,7 +866,7 @@ void StopMusicStream(int index)
             stb_vorbis_close(currentMusic[index].stream);
         }
         
-        if(!getMusicStreamCount()) musicEnabled = false;
+        if(!getMusicStreamCount()) musicEnabled_g = false;
         if(currentMusic[index].stream || currentMusic[index].chipctx)
         {
             currentMusic[index].stream = NULL;
@@ -884,7 +889,7 @@ int getMusicStreamCount(void)
 void PauseMusicStream(int index)
 {
     // Pause music stream if music available!
-    if (currentMusic[index].ctx && musicEnabled)
+    if (currentMusic[index].ctx && musicEnabled_g)
     {
         TraceLog(INFO, "Pausing music stream");
         alSourcePause(currentMusic[index].ctx->alSource);
@@ -987,7 +992,7 @@ static bool BufferMusicStream(int index)
     int  size = 0;              // Total size of data steamed (in bytes)
     bool active = true;         // We can get more data from stream (not finished)
         
-    if (musicEnabled)
+    if (musicEnabled_g)
     {
         if(!currentMusic[index].ctx->playing)
         {
@@ -1031,13 +1036,25 @@ static void EmptyMusicStream(int index)
     }
 }
 
+//determine if a music stream is ready to be written to
+static bool isMusicStreamReady(int index)
+{
+    ALint processed = 0;
+    alGetSourcei(currentMusic[index].ctx->alSource, AL_BUFFERS_PROCESSED, &processed);
+    
+    if(processed)
+        return true;
+    
+    return false;
+}
+
 // Update (re-fill) music buffers if data already processed
 void UpdateMusicStream(int index)
 {
     ALenum state;
     bool active = true;
 
-    if (index < MAX_MUSIC_STREAMS && musicEnabled)
+    if (index < MAX_MUSIC_STREAMS && musicEnabled_g && isMusicStreamReady(index))
     {
         active = BufferMusicStream(index);
         

From d38d7a1bedadf7c4824c9bb1d4a9ecd3f0935b9c Mon Sep 17 00:00:00 2001
From: Joshua Reisenauer <kd7tck@msn.com>
Date: Sat, 14 May 2016 16:30:32 -0700
Subject: [PATCH 09/14] clean up on buffering and preconditions

---
 src/audio.c | 89 ++++++++++++++++++++++++++++++-----------------------
 1 file changed, 50 insertions(+), 39 deletions(-)

diff --git a/src/audio.c b/src/audio.c
index 0fd1bb13..30e2343f 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -889,7 +889,7 @@ int getMusicStreamCount(void)
 void PauseMusicStream(int index)
 {
     // Pause music stream if music available!
-    if (currentMusic[index].ctx && musicEnabled_g)
+    if (index < MAX_MUSIC_STREAMS && currentMusic[index].ctx && musicEnabled_g)
     {
         TraceLog(INFO, "Pausing music stream");
         alSourcePause(currentMusic[index].ctx->alSource);
@@ -902,7 +902,7 @@ void ResumeMusicStream(int index)
 {
     // Resume music playing... if music available!
     ALenum state;
-    if(currentMusic[index].ctx){
+    if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx){
         alGetSourcei(currentMusic[index].ctx->alSource, AL_SOURCE_STATE, &state);
         if (state == AL_PAUSED)
         {
@@ -962,17 +962,20 @@ float GetMusicTimeLength(int index)
 float GetMusicTimePlayed(int index)
 {
     float secondsPlayed;
-    if (currentMusic[index].chipTune)
-    {
-        uint64_t samples;
-        jar_xm_get_position(currentMusic[index].chipctx, NULL, NULL, NULL, &samples);
-        secondsPlayed = (float)samples / (48000 * currentMusic[index].ctx->channels); // Not sure if this is the correct value
-    }
-    else
+    if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx)
     {
-        int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].ctx->channels;
-        int samplesPlayed = totalSamples - currentMusic[index].totalSamplesLeft;
-        secondsPlayed = (float)samplesPlayed / (currentMusic[index].ctx->sampleRate * currentMusic[index].ctx->channels);
+        if (currentMusic[index].chipTune)
+        {
+            uint64_t samples;
+            jar_xm_get_position(currentMusic[index].chipctx, NULL, NULL, NULL, &samples);
+            secondsPlayed = (float)samples / (48000 * currentMusic[index].ctx->channels); // Not sure if this is the correct value
+        }
+        else
+        {
+            int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].ctx->channels;
+            int samplesPlayed = totalSamples - currentMusic[index].totalSamplesLeft;
+            secondsPlayed = (float)samplesPlayed / (currentMusic[index].ctx->sampleRate * currentMusic[index].ctx->channels);
+        }
     }
     
 
@@ -989,33 +992,42 @@ static bool BufferMusicStream(int index)
     short pcm[MUSIC_BUFFER_SIZE_SHORT];
     float pcmf[MUSIC_BUFFER_SIZE_FLOAT];
     
-    int  size = 0;              // Total size of data steamed (in bytes)
+    int  size = 0;              // Total size of data steamed in L+R samples
     bool active = true;         // We can get more data from stream (not finished)
         
-    if (musicEnabled_g)
+    
+    if(!currentMusic[index].ctx->playing && currentMusic[index].totalSamplesLeft > 0)
     {
-        if(!currentMusic[index].ctx->playing)
-        {
-            UpdateAudioContext(currentMusic[index].ctx, NULL, 0);
-            return false;
-        }
-        
-        if (currentMusic[index].chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
-        {
-            jar_xm_generate_samples(currentMusic[index].chipctx, pcmf, MUSIC_BUFFER_SIZE_FLOAT / 2); // reads 2*readlen shorts and moves them to buffer+size memory location
-            UpdateAudioContext(currentMusic[index].ctx, pcmf, MUSIC_BUFFER_SIZE_FLOAT);
-            currentMusic[index].totalSamplesLeft -= MUSIC_BUFFER_SIZE_FLOAT;
-            if(currentMusic[index].totalSamplesLeft <= 0) active = false;
-        }
+        UpdateAudioContext(currentMusic[index].ctx, NULL, 0);
+        return true; // it is still active, only it is paused
+    }
+    
+    
+    if (currentMusic[index].chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
+    {
+        if(currentMusic[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_FLOAT / 2)
+            size = MUSIC_BUFFER_SIZE_FLOAT / 2;
         else
-        {
-            int streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic[index].stream, currentMusic[index].ctx->channels, pcm, MUSIC_BUFFER_SIZE_SHORT);
-            UpdateAudioContext(currentMusic[index].ctx, pcm, streamedBytes);
-            currentMusic[index].totalSamplesLeft -= streamedBytes*currentMusic[index].ctx->channels;
-            if(currentMusic[index].totalSamplesLeft <= 0) active = false;
-        }
-        TraceLog(DEBUG, "Buffering index:%i, chiptune:%i", index, (int)currentMusic[index].chipTune);
+            size = currentMusic[index].totalSamplesLeft / 2;
+    
+        jar_xm_generate_samples(currentMusic[index].chipctx, pcmf, size); // reads 2*readlen shorts and moves them to buffer+size memory location
+        UpdateAudioContext(currentMusic[index].ctx, pcmf, size * 2);
+        currentMusic[index].totalSamplesLeft -= size * 2;
     }
+    else
+    {
+        if(currentMusic[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT)
+            size = MUSIC_BUFFER_SIZE_SHORT;
+        else
+            size = currentMusic[index].totalSamplesLeft;
+        
+        int streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic[index].stream, currentMusic[index].ctx->channels, pcm, size);
+        UpdateAudioContext(currentMusic[index].ctx, pcm, streamedBytes * currentMusic[index].ctx->channels);
+        currentMusic[index].totalSamplesLeft -= streamedBytes * currentMusic[index].ctx->channels;
+    }
+    
+    TraceLog(DEBUG, "Buffering index:%i, chiptune:%i", index, (int)currentMusic[index].chipTune);
+    if(currentMusic[index].totalSamplesLeft <= 0) active = false;
 
     return active;
 }
@@ -1042,8 +1054,7 @@ static bool isMusicStreamReady(int index)
     ALint processed = 0;
     alGetSourcei(currentMusic[index].ctx->alSource, AL_BUFFERS_PROCESSED, &processed);
     
-    if(processed)
-        return true;
+    if(processed) return true;
     
     return false;
 }
@@ -1054,11 +1065,11 @@ void UpdateMusicStream(int index)
     ALenum state;
     bool active = true;
 
-    if (index < MAX_MUSIC_STREAMS && musicEnabled_g && isMusicStreamReady(index))
+    if (index < MAX_MUSIC_STREAMS && musicEnabled_g && currentMusic[index].ctx && isMusicStreamReady(index))
     {
         active = BufferMusicStream(index);
         
-        if ((!active) && (currentMusic[index].loop))
+        if (!active && currentMusic[index].loop && currentMusic[index].ctx->playing)
         {
             if(currentMusic[index].chipTune)
             {
@@ -1067,7 +1078,7 @@ void UpdateMusicStream(int index)
             else
             {
                 stb_vorbis_seek_start(currentMusic[index].stream);
-                currentMusic[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[index].stream)*currentMusic[index].ctx->channels;
+                currentMusic[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].ctx->channels;
             }
             active = BufferMusicStream(index);
         }

From 86fbf4fd8f0cd5301259b2115125c45c9872c02a Mon Sep 17 00:00:00 2001
From: Joshua Reisenauer <kd7tck@msn.com>
Date: Sun, 15 May 2016 02:09:57 -0700
Subject: [PATCH 10/14] logic bug fix

---
 src/audio.c | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

diff --git a/src/audio.c b/src/audio.c
index 30e2343f..cc5ca1a2 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -996,10 +996,10 @@ static bool BufferMusicStream(int index)
     bool active = true;         // We can get more data from stream (not finished)
         
     
-    if(!currentMusic[index].ctx->playing && currentMusic[index].totalSamplesLeft > 0)
+    if (!currentMusic[index].ctx->playing && currentMusic[index].totalSamplesLeft > 0)
     {
         UpdateAudioContext(currentMusic[index].ctx, NULL, 0);
-        return true; // it is still active, only it is paused
+        return true; // it is still active but it is paused
     }
     
     
@@ -1071,7 +1071,7 @@ void UpdateMusicStream(int index)
         
         if (!active && currentMusic[index].loop && currentMusic[index].ctx->playing)
         {
-            if(currentMusic[index].chipTune)
+            if (currentMusic[index].chipTune)
             {
                 currentMusic[index].totalSamplesLeft = currentMusic[index].totalLengthSeconds * currentMusic[index].ctx->sampleRate;
             }
@@ -1080,7 +1080,7 @@ void UpdateMusicStream(int index)
                 stb_vorbis_seek_start(currentMusic[index].stream);
                 currentMusic[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].ctx->channels;
             }
-            active = BufferMusicStream(index);
+            active = true;
         }
         
 
@@ -1088,7 +1088,7 @@ void UpdateMusicStream(int index)
         
         alGetSourcei(currentMusic[index].ctx->alSource, AL_SOURCE_STATE, &state);
 
-        if ((state != AL_PLAYING) && active) alSourcePlay(currentMusic[index].ctx->alSource);
+        if (state != AL_PLAYING && active && currentMusic[index].ctx->playing) alSourcePlay(currentMusic[index].ctx->alSource);
 
         if (!active) StopMusicStream(index);
         

From 76ff4d220ee735b8b86bd4dae776665cf68e4fb4 Mon Sep 17 00:00:00 2001
From: Joshua Reisenauer <kd7tck@msn.com>
Date: Sun, 15 May 2016 19:37:15 -0700
Subject: [PATCH 11/14] renamed everything so it is obvious what it does

---
 src/audio.c  | 380 ++++++++++++++++++++++++---------------------------
 src/audio.h  |  15 +-
 src/raylib.h |  15 +-
 3 files changed, 190 insertions(+), 220 deletions(-)

diff --git a/src/audio.c b/src/audio.c
index cc5ca1a2..584d3ad1 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -77,10 +77,10 @@
 // Types and Structures Definition
 //----------------------------------------------------------------------------------
 
-// Audio Context, used to create custom audio streams that are not bound to a sound file. There can be
-// no more than 4 concurrent audio contexts in use. This is due to each active context being tied to
-// a dedicated mix channel. All audio is 32bit floating point in stereo.
-typedef struct AudioContext_t {
+// Used to create custom audio streams that are not bound to a specific file. There can be
+// no more than 4 concurrent mixchannels in use. This is due to each active mixc being tied to
+// a dedicated mix channel.
+typedef struct MixChannel_t {
     unsigned short sampleRate;           // default is 48000
     unsigned char channels;              // 1=mono,2=stereo
     unsigned char mixChannel;            // 0-3 or mixA-mixD, each mix channel can receive up to one dedicated audio stream
@@ -89,14 +89,14 @@ typedef struct AudioContext_t {
     ALenum alFormat;                     // openAL format specifier
     ALuint alSource;                     // openAL source
     ALuint alBuffer[MAX_STREAM_BUFFERS]; // openAL sample buffer
-} AudioContext_t;
+} MixChannel_t;
 
 // Music type (file streaming from memory)
-// NOTE: Anything longer than ~10 seconds should be streamed...
+// NOTE: Anything longer than ~10 seconds should be streamed into a mix channel...
 typedef struct Music {
     stb_vorbis *stream;
-    jar_xm_context_t *chipctx; // Stores jar_xm context
-    AudioContext_t *ctx; // audio context
+    jar_xm_context_t *chipctx; // Stores jar_xm mixc
+    MixChannel_t *mixc; // mix channel
     
     int totalSamplesLeft;
     float totalLengthSeconds;
@@ -111,9 +111,9 @@ typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType;
 //----------------------------------------------------------------------------------
 // Global Variables Definition
 //----------------------------------------------------------------------------------
-static AudioContext_t* mixChannelsActive_g[MAX_AUDIO_CONTEXTS];      // What mix channels are currently active
+static MixChannel_t* mixChannelsActive_g[MAX_AUDIO_CONTEXTS];      // What mix channels are currently active
 static bool musicEnabled_g = false;
-static Music currentMusic[MAX_MUSIC_STREAMS];                        // Current music loaded, up to two can play at the same time
+static Music currentMusic[MAX_MUSIC_STREAMS];                      // Current music loaded, up to two can play at the same time
 
 //----------------------------------------------------------------------------------
 // Module specific Functions Declaration
@@ -122,13 +122,17 @@ static Wave LoadWAV(const char *fileName);         // Load WAV file
 static Wave LoadOGG(char *fileName);               // Load OGG file
 static void UnloadWave(Wave wave);                 // Unload wave data
 
-static bool BufferMusicStream(int index);          // Fill music buffers with data
-static void EmptyMusicStream(int index);           // Empty music buffers
+static bool BufferMusicStream(int index, int numBuffers); // Fill music buffers with data
+static void EmptyMusicStream(int index); // Empty music buffers
 
-static unsigned short FillAlBufferWithSilence(AudioContext_t *context, 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
-static bool isMusicStreamReady(int index);                                            // Checks if music buffer is ready to be refilled
+
+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 unsigned short BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements); // Pushes more audio data into mixc mix channel, if NULL is passed it pauses
+static unsigned short 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
+static int IsMusicStreamReadyForBuffering(int index); // Checks if music buffer is ready to be refilled
 
 #if defined(AUDIO_STANDALONE)
 const char *GetExtension(const char *fileName);     // Get the extension for a filename
@@ -139,7 +143,7 @@ void TraceLog(int msgType, const char *text, ...);  // Outputs a trace log messa
 // Module Functions Definition - Audio Device initialization and Closing
 //----------------------------------------------------------------------------------
 
-// Initialize audio device and context
+// Initialize audio device and mixc
 void InitAudioDevice(void)
 {
     // Open and initialize a device with default settings
@@ -155,7 +159,7 @@ void InitAudioDevice(void)
 
         alcCloseDevice(device);
 
-        TraceLog(ERROR, "Could not setup audio context");
+        TraceLog(ERROR, "Could not setup mix channel");
     }
 
     TraceLog(INFO, "Audio device and context initialized successfully: %s", alcGetString(device, ALC_DEVICE_SPECIFIER));
@@ -171,14 +175,14 @@ void CloseAudioDevice(void)
 {
     for(int index=0; index<MAX_MUSIC_STREAMS; index++)
     {
-        if(currentMusic[index].ctx) StopMusicStream(index);      // Stop music streaming and close current stream
+        if(currentMusic[index].mixc) StopMusicStream(index);      // Stop music streaming and close current stream
     }
     
 
     ALCdevice *device;
     ALCcontext *context = alcGetCurrentContext();
 
-    if (context == NULL) TraceLog(WARNING, "Could not get current audio context for closing");
+    if (context == NULL) TraceLog(WARNING, "Could not get current mix channel for closing");
 
     device = alcGetContextsDevice(context);
 
@@ -203,186 +207,141 @@ bool IsAudioDeviceReady(void)
 // Module Functions Definition - Custom audio output
 //----------------------------------------------------------------------------------
 
-// Audio contexts are for outputing custom audio waveforms, This will shut down any other sound sources currently playing
-// The mixChannel is what mix 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 InitAudioContext(48000, 0, 2, true); // mixchannel 1, 48khz, stereo, floating point
-AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint)
+// 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)
 {
     if(mixChannel >= MAX_AUDIO_CONTEXTS) return NULL;
     if(!IsAudioDeviceReady()) InitAudioDevice();
     
     if(!mixChannelsActive_g[mixChannel]){
-        AudioContext_t *ac = (AudioContext_t*)malloc(sizeof(AudioContext_t));
-        ac->sampleRate = sampleRate;
-        ac->channels = channels;
-        ac->mixChannel = mixChannel;
-        ac->floatingPoint = floatingPoint;
-        mixChannelsActive_g[mixChannel] = ac;
+        MixChannel_t *mixc = (MixChannel_t*)malloc(sizeof(MixChannel_t));
+        mixc->sampleRate = sampleRate;
+        mixc->channels = channels;
+        mixc->mixChannel = mixChannel;
+        mixc->floatingPoint = floatingPoint;
+        mixChannelsActive_g[mixChannel] = mixc;
         
         // setup openAL format
         if(channels == 1)
         {
             if(floatingPoint)
-                ac->alFormat = AL_FORMAT_MONO_FLOAT32;
+                mixc->alFormat = AL_FORMAT_MONO_FLOAT32;
             else
-                ac->alFormat = AL_FORMAT_MONO16;
+                mixc->alFormat = AL_FORMAT_MONO16;
         }
         else if(channels == 2)
         {
             if(floatingPoint)
-                ac->alFormat = AL_FORMAT_STEREO_FLOAT32;
+                mixc->alFormat = AL_FORMAT_STEREO_FLOAT32;
             else
-                ac->alFormat = AL_FORMAT_STEREO16;
+                mixc->alFormat = AL_FORMAT_STEREO16;
         }
         
         // Create an audio source
-        alGenSources(1, &ac->alSource);
-        alSourcef(ac->alSource, AL_PITCH, 1);
-        alSourcef(ac->alSource, AL_GAIN, 1);
-        alSource3f(ac->alSource, AL_POSITION, 0, 0, 0);
-        alSource3f(ac->alSource, AL_VELOCITY, 0, 0, 0);
+        alGenSources(1, &mixc->alSource);
+        alSourcef(mixc->alSource, AL_PITCH, 1);
+        alSourcef(mixc->alSource, AL_GAIN, 1);
+        alSource3f(mixc->alSource, AL_POSITION, 0, 0, 0);
+        alSource3f(mixc->alSource, AL_VELOCITY, 0, 0, 0);
         
         // Create Buffer
-        alGenBuffers(MAX_STREAM_BUFFERS, ac->alBuffer);
+        alGenBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer);
         
         //fill buffers
         int x;
         for(x=0;x<MAX_STREAM_BUFFERS;x++)
-            FillAlBufferWithSilence(ac, ac->alBuffer[x]);
+            FillAlBufferWithSilence(mixc, mixc->alBuffer[x]);
         
-        alSourceQueueBuffers(ac->alSource, MAX_STREAM_BUFFERS, ac->alBuffer);
-        alSourcePlay(ac->alSource);
-        ac->playing = true;
+        alSourceQueueBuffers(mixc->alSource, MAX_STREAM_BUFFERS, mixc->alBuffer);
+        mixc->playing = true;
+        alSourcePlay(mixc->alSource);
         
-        return ac;
+        return mixc;
     }
     return NULL;
 }
 
-// Frees buffer in audio context
-void CloseAudioContext(AudioContext ctx)
+// Frees buffer in mix channel
+static void CloseMixChannel(MixChannel_t* mixc)
 {
-    AudioContext_t *context = (AudioContext_t*)ctx;
-    if(context){
-        alSourceStop(context->alSource);
-        context->playing = false;
+    if(mixc){
+        alSourceStop(mixc->alSource);
+        mixc->playing = false;
         
         //flush out all queued buffers
         ALuint buffer = 0;
         int queued = 0;
-        alGetSourcei(context->alSource, AL_BUFFERS_QUEUED, &queued);
+        alGetSourcei(mixc->alSource, AL_BUFFERS_QUEUED, &queued);
         while (queued > 0)
         {
-            alSourceUnqueueBuffers(context->alSource, 1, &buffer);
+            alSourceUnqueueBuffers(mixc->alSource, 1, &buffer);
             queued--;
         }
         
         //delete source and buffers
-        alDeleteSources(1, &context->alSource);
-        alDeleteBuffers(MAX_STREAM_BUFFERS, context->alBuffer);
-        mixChannelsActive_g[context->mixChannel] = NULL;
-        free(context);
-        ctx = NULL;
+        alDeleteSources(1, &mixc->alSource);
+        alDeleteBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer);
+        mixChannelsActive_g[mixc->mixChannel] = NULL;
+        free(mixc);
+        mixc = NULL;
     }
 }
 
-// Pushes more audio data into context mix channel, if none are ever pushed then zeros are fed in.
-// Call "UpdateAudioContext(ctx, NULL, 0)" if you want to pause the audio.
+// 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.
-unsigned short UpdateAudioContext(AudioContext ctx, void *data, unsigned short numberElements)
+static unsigned short BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements)
 {
-    AudioContext_t *context = (AudioContext_t*)ctx;
-    
-    if(!context || (context->channels == 2 && numberElements % 2 != 0)) return 0; // when there is two channels there must be an even number of samples
+    if(!mixc || mixChannelsActive_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
-        alSourcePause(context->alSource);
-        context->playing = false;
+        if(mixc->playing){
+            alSourcePause(mixc->alSource);
+            mixc->playing = false;
+        }
         return 0;
     }
-    else
+    else if(!mixc->playing)
     { // restart audio otherwise
-        ALint state;
-        alGetSourcei(context->alSource, AL_SOURCE_STATE, &state);
-        if (state != AL_PLAYING){
-            alSourcePlay(context->alSource);
-            context->playing = true;
-        }
+        alSourcePlay(mixc->alSource);
+        mixc->playing = true;
     }
     
-    if (context && context->playing && mixChannelsActive_g[context->mixChannel] == context)
+   
+    ALuint buffer = 0;
+    
+    alSourceUnqueueBuffers(mixc->alSource, 1, &buffer);
+    if(!buffer) return 0;
+    if(mixc->floatingPoint) // process float buffers
     {
-        ALint processed = 0;
-        ALuint buffer = 0;
-        unsigned short numberProcessed = 0;
-        unsigned short numberRemaining = numberElements;
-        
-        
-        alGetSourcei(context->alSource, AL_BUFFERS_PROCESSED, &processed); // Get the number of already processed buffers (if any)
-        if(!processed) return 0; // nothing to process, queue is still full
-        
-        
-        while (processed > 0)
-        {
-            if(context->floatingPoint) // process float buffers
-            {
-                float *ptr = (float*)data;
-                alSourceUnqueueBuffers(context->alSource, 1, &buffer);
-                if(numberRemaining >= MUSIC_BUFFER_SIZE_FLOAT)
-                {
-                    alBufferData(buffer, context->alFormat, &ptr[numberProcessed], MUSIC_BUFFER_SIZE_FLOAT*sizeof(float), context->sampleRate);
-                    numberProcessed+=MUSIC_BUFFER_SIZE_FLOAT;
-                    numberRemaining-=MUSIC_BUFFER_SIZE_FLOAT;
-                }
-                else
-                {
-                    alBufferData(buffer, context->alFormat, &ptr[numberProcessed], numberRemaining*sizeof(float), context->sampleRate);
-                    numberProcessed+=numberRemaining;
-                    numberRemaining=0;
-                }
-                alSourceQueueBuffers(context->alSource, 1, &buffer);
-                processed--;
-            }
-            else if(!context->floatingPoint) // process short buffers
-            {
-                short *ptr = (short*)data;
-                alSourceUnqueueBuffers(context->alSource, 1, &buffer);
-                if(numberRemaining >= MUSIC_BUFFER_SIZE_SHORT)
-                {
-                    alBufferData(buffer, context->alFormat, &ptr[numberProcessed], MUSIC_BUFFER_SIZE_FLOAT*sizeof(short), context->sampleRate);
-                    numberProcessed+=MUSIC_BUFFER_SIZE_SHORT;
-                    numberRemaining-=MUSIC_BUFFER_SIZE_SHORT;
-                }
-                else
-                {
-                    alBufferData(buffer, context->alFormat, &ptr[numberProcessed], numberRemaining*sizeof(short), context->sampleRate);
-                    numberProcessed+=numberRemaining;
-                    numberRemaining=0;
-                }
-                alSourceQueueBuffers(context->alSource, 1, &buffer);
-                processed--;
-            }
-            else
-                break;
-        }
-        return numberProcessed;
+        float *ptr = (float*)data;
+        alBufferData(buffer, mixc->alFormat, ptr, numberElements*sizeof(float), mixc->sampleRate);
+    }
+    else // process short buffers
+    {
+        short *ptr = (short*)data;
+        alBufferData(buffer, mixc->alFormat, ptr, numberElements*sizeof(short), mixc->sampleRate);
     }
-    return 0;
+    alSourceQueueBuffers(mixc->alSource, 1, &buffer);
+    
+    return numberElements;
 }
 
 // fill buffer with zeros, returns number processed
-static unsigned short FillAlBufferWithSilence(AudioContext_t *context, ALuint buffer)
+static unsigned short FillAlBufferWithSilence(MixChannel_t *mixc, ALuint buffer)
 {
-    if(context->floatingPoint){
+    if(mixc->floatingPoint){
         float pcm[MUSIC_BUFFER_SIZE_FLOAT] = {0.f};
-        alBufferData(buffer, context->alFormat, pcm, MUSIC_BUFFER_SIZE_FLOAT*sizeof(float), context->sampleRate);
+        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};
-        alBufferData(buffer, context->alFormat, pcm, MUSIC_BUFFER_SIZE_SHORT*sizeof(short), context->sampleRate);
+        alBufferData(buffer, mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_SHORT*sizeof(short), mixc->sampleRate);
         return MUSIC_BUFFER_SIZE_SHORT;
     }
 }
@@ -417,6 +376,28 @@ static void ResampleByteToFloat(char *chars, float *floats, unsigned short len)
     }
 }
 
+// used to output raw audio streams, returns negative numbers on error
+RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint)
+{
+    int mixIndex;
+    for(mixIndex = 0; mixIndex < MAX_AUDIO_CONTEXTS; mixIndex++) // find empty mix channel slot
+    {
+        if(mixChannelsActive_g[mixIndex] == NULL) break;
+        else if(mixIndex = MAX_AUDIO_CONTEXTS - 1) return -1; // error
+    }
+    
+    if(InitMixChannel(sampleRate, mixIndex, channels, floatingPoint))
+        return mixIndex;
+    else
+        return -2; // error
+}
+
+void CloseRawAudioContext(RawAudioContext ctx)
+{
+    if(mixChannelsActive_g[ctx])
+        CloseMixChannel(mixChannelsActive_g[ctx]);
+}
+
 
 
 //----------------------------------------------------------------------------------
@@ -807,14 +788,14 @@ int PlayMusicStream(int musicIndex, char *fileName)
             currentMusic[musicIndex].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[musicIndex].stream);
             
             if (info.channels == 2){
-                currentMusic[musicIndex].ctx = InitAudioContext(info.sample_rate, mixIndex, 2, false);
-                currentMusic[musicIndex].ctx->playing = true;
+                currentMusic[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 2, false);
+                currentMusic[musicIndex].mixc->playing = true;
             }
             else{
-                currentMusic[musicIndex].ctx = InitAudioContext(info.sample_rate, mixIndex, 1, false);
-                currentMusic[musicIndex].ctx->playing = true;
+                currentMusic[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 1, false);
+                currentMusic[musicIndex].mixc->playing = true;
             }
-            if(!currentMusic[musicIndex].ctx) return 4; // error
+            if(!currentMusic[musicIndex].mixc) return 4; // error
         }
     }
     else if (strcmp(GetExtension(fileName),"xm") == 0)
@@ -832,9 +813,9 @@ int PlayMusicStream(int musicIndex, char *fileName)
             TraceLog(INFO, "[%s] XM number of samples: %i", fileName, currentMusic[musicIndex].totalSamplesLeft);
             TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, currentMusic[musicIndex].totalLengthSeconds);
             
-            currentMusic[musicIndex].ctx = InitAudioContext(48000, mixIndex, 2, true);
-            if(!currentMusic[musicIndex].ctx) return 5; // error
-            currentMusic[musicIndex].ctx->playing = true;
+            currentMusic[musicIndex].mixc = InitMixChannel(48000, mixIndex, 2, false);
+            if(!currentMusic[musicIndex].mixc) return 5; // error
+            currentMusic[musicIndex].mixc->playing = true;
         }
         else
         {
@@ -853,9 +834,9 @@ int PlayMusicStream(int musicIndex, char *fileName)
 // Stop music playing for individual music index of currentMusic array (close stream)
 void StopMusicStream(int index)
 {
-    if (index < MAX_MUSIC_STREAMS && currentMusic[index].ctx)
+    if (index < MAX_MUSIC_STREAMS && currentMusic[index].mixc)
     {
-        CloseAudioContext(currentMusic[index].ctx);
+        CloseMixChannel(currentMusic[index].mixc);
         
         if (currentMusic[index].chipTune)
         {
@@ -889,11 +870,11 @@ int getMusicStreamCount(void)
 void PauseMusicStream(int index)
 {
     // Pause music stream if music available!
-    if (index < MAX_MUSIC_STREAMS && currentMusic[index].ctx && musicEnabled_g)
+    if (index < MAX_MUSIC_STREAMS && currentMusic[index].mixc && musicEnabled_g)
     {
         TraceLog(INFO, "Pausing music stream");
-        alSourcePause(currentMusic[index].ctx->alSource);
-        currentMusic[index].ctx->playing = false;
+        alSourcePause(currentMusic[index].mixc->alSource);
+        currentMusic[index].mixc->playing = false;
     }
 }
 
@@ -902,13 +883,13 @@ void ResumeMusicStream(int index)
 {
     // Resume music playing... if music available!
     ALenum state;
-    if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx){
-        alGetSourcei(currentMusic[index].ctx->alSource, AL_SOURCE_STATE, &state);
+    if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){
+        alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state);
         if (state == AL_PAUSED)
         {
             TraceLog(INFO, "Resuming music stream");
-            alSourcePlay(currentMusic[index].ctx->alSource);
-            currentMusic[index].ctx->playing = true;
+            alSourcePlay(currentMusic[index].mixc->alSource);
+            currentMusic[index].mixc->playing = true;
         }
     }
 }
@@ -919,8 +900,8 @@ bool IsMusicPlaying(int index)
     bool playing = false;
     ALint state;
     
-    if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx){
-        alGetSourcei(currentMusic[index].ctx->alSource, AL_SOURCE_STATE, &state);
+    if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){
+        alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state);
         if (state == AL_PLAYING) playing = true;
     }
 
@@ -930,15 +911,15 @@ bool IsMusicPlaying(int index)
 // Set volume for music
 void SetMusicVolume(int index, float volume)
 {
-    if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx){
-        alSourcef(currentMusic[index].ctx->alSource, AL_GAIN, volume);
+    if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){
+        alSourcef(currentMusic[index].mixc->alSource, AL_GAIN, volume);
     }
 }
 
 void SetMusicPitch(int index, float pitch)
 {
-    if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx){
-        alSourcef(currentMusic[index].ctx->alSource, AL_PITCH, pitch);
+    if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){
+        alSourcef(currentMusic[index].mixc->alSource, AL_PITCH, pitch);
     }
 }
 
@@ -962,19 +943,19 @@ float GetMusicTimeLength(int index)
 float GetMusicTimePlayed(int index)
 {
     float secondsPlayed;
-    if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx)
+    if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc)
     {
         if (currentMusic[index].chipTune)
         {
             uint64_t samples;
             jar_xm_get_position(currentMusic[index].chipctx, NULL, NULL, NULL, &samples);
-            secondsPlayed = (float)samples / (48000 * currentMusic[index].ctx->channels); // Not sure if this is the correct value
+            secondsPlayed = (float)samples / (48000 * currentMusic[index].mixc->channels); // Not sure if this is the correct value
         }
         else
         {
-            int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].ctx->channels;
+            int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].mixc->channels;
             int samplesPlayed = totalSamples - currentMusic[index].totalSamplesLeft;
-            secondsPlayed = (float)samplesPlayed / (currentMusic[index].ctx->sampleRate * currentMusic[index].ctx->channels);
+            secondsPlayed = (float)samplesPlayed / (currentMusic[index].mixc->sampleRate * currentMusic[index].mixc->channels);
         }
     }
     
@@ -987,32 +968,32 @@ float GetMusicTimePlayed(int index)
 //----------------------------------------------------------------------------------
 
 // Fill music buffers with new data from music stream
-static bool BufferMusicStream(int index)
+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
+    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 (!currentMusic[index].ctx->playing && currentMusic[index].totalSamplesLeft > 0)
-    {
-        UpdateAudioContext(currentMusic[index].ctx, NULL, 0);
-        return true; // it is still active but it is paused
-    }
-    
     
     if (currentMusic[index].chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
     {
-        if(currentMusic[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_FLOAT / 2)
-            size = MUSIC_BUFFER_SIZE_FLOAT / 2;
+        if(currentMusic[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT)
+            size = MUSIC_BUFFER_SIZE_SHORT / 2;
         else
             size = currentMusic[index].totalSamplesLeft / 2;
-    
-        jar_xm_generate_samples(currentMusic[index].chipctx, pcmf, size); // reads 2*readlen shorts and moves them to buffer+size memory location
-        UpdateAudioContext(currentMusic[index].ctx, pcmf, size * 2);
-        currentMusic[index].totalSamplesLeft -= size * 2;
+        
+        for(int x=0; x<numBuffers; x++)
+        {
+            jar_xm_generate_samples_16bit(currentMusic[index].chipctx, pcm, size); // reads 2*readlen shorts and moves them to buffer+size memory location
+            BufferMixChannel(currentMusic[index].mixc, pcm, size * 2);
+            currentMusic[index].totalSamplesLeft -= size * 2;
+            if(currentMusic[index].totalSamplesLeft <= 0)
+            {
+                active = false;
+                break;
+            }
+        }
     }
     else
     {
@@ -1021,13 +1002,18 @@ static bool BufferMusicStream(int index)
         else
             size = currentMusic[index].totalSamplesLeft;
         
-        int streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic[index].stream, currentMusic[index].ctx->channels, pcm, size);
-        UpdateAudioContext(currentMusic[index].ctx, pcm, streamedBytes * currentMusic[index].ctx->channels);
-        currentMusic[index].totalSamplesLeft -= streamedBytes * currentMusic[index].ctx->channels;
+        for(int x=0; x<numBuffers; x++)
+        {
+            int streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic[index].stream, currentMusic[index].mixc->channels, pcm, size);
+            BufferMixChannel(currentMusic[index].mixc, pcm, streamedBytes * currentMusic[index].mixc->channels);
+            currentMusic[index].totalSamplesLeft -= streamedBytes * currentMusic[index].mixc->channels;
+            if(currentMusic[index].totalSamplesLeft <= 0)
+            {
+                active = false;
+                break;
+            }
+        }
     }
-    
-    TraceLog(DEBUG, "Buffering index:%i, chiptune:%i", index, (int)currentMusic[index].chipTune);
-    if(currentMusic[index].totalSamplesLeft <= 0) active = false;
 
     return active;
 }
@@ -1038,25 +1024,22 @@ static void EmptyMusicStream(int index)
     ALuint buffer = 0;
     int queued = 0;
 
-    alGetSourcei(currentMusic[index].ctx->alSource, AL_BUFFERS_QUEUED, &queued);
+    alGetSourcei(currentMusic[index].mixc->alSource, AL_BUFFERS_QUEUED, &queued);
 
     while (queued > 0)
     {
-        alSourceUnqueueBuffers(currentMusic[index].ctx->alSource, 1, &buffer);
+        alSourceUnqueueBuffers(currentMusic[index].mixc->alSource, 1, &buffer);
 
         queued--;
     }
 }
 
 //determine if a music stream is ready to be written to
-static bool isMusicStreamReady(int index)
+static int IsMusicStreamReadyForBuffering(int index)
 {
     ALint processed = 0;
-    alGetSourcei(currentMusic[index].ctx->alSource, AL_BUFFERS_PROCESSED, &processed);
-    
-    if(processed) return true;
-    
-    return false;
+    alGetSourcei(currentMusic[index].mixc->alSource, AL_BUFFERS_PROCESSED, &processed);
+    return processed;
 }
 
 // Update (re-fill) music buffers if data already processed
@@ -1064,21 +1047,22 @@ void UpdateMusicStream(int index)
 {
     ALenum state;
     bool active = true;
-
-    if (index < MAX_MUSIC_STREAMS && musicEnabled_g && currentMusic[index].ctx && isMusicStreamReady(index))
+    int numBuffers = IsMusicStreamReadyForBuffering(index);
+    
+    if (currentMusic[index].mixc->playing && index < MAX_MUSIC_STREAMS && musicEnabled_g && currentMusic[index].mixc && numBuffers)
     {
-        active = BufferMusicStream(index);
+        active = BufferMusicStream(index, numBuffers);
         
-        if (!active && currentMusic[index].loop && currentMusic[index].ctx->playing)
+        if (!active && currentMusic[index].loop)
         {
             if (currentMusic[index].chipTune)
             {
-                currentMusic[index].totalSamplesLeft = currentMusic[index].totalLengthSeconds * currentMusic[index].ctx->sampleRate;
+                currentMusic[index].totalSamplesLeft = currentMusic[index].totalLengthSeconds * 48000;
             }
             else
             {
                 stb_vorbis_seek_start(currentMusic[index].stream);
-                currentMusic[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].ctx->channels;
+                currentMusic[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].mixc->channels;
             }
             active = true;
         }
@@ -1086,9 +1070,9 @@ void UpdateMusicStream(int index)
 
         if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data...");
         
-        alGetSourcei(currentMusic[index].ctx->alSource, AL_SOURCE_STATE, &state);
+        alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state);
 
-        if (state != AL_PLAYING && active && currentMusic[index].ctx->playing) alSourcePlay(currentMusic[index].ctx->alSource);
+        if (state != AL_PLAYING && active) alSourcePlay(currentMusic[index].mixc->alSource);
 
         if (!active) StopMusicStream(index);
         
diff --git a/src/audio.h b/src/audio.h
index 63c1c136..d3276bf6 100644
--- a/src/audio.h
+++ b/src/audio.h
@@ -61,10 +61,7 @@ typedef struct Wave {
     short channels;
 } Wave;
 
-// Audio Context, used to create custom audio streams that are not bound to a sound file. There can be
-// no more than 4 concurrent audio contexts in use. This is due to each active context being tied to
-// a dedicated mix channel.
-typedef void* AudioContext;
+typedef int RawAudioContext;
 
 #ifdef __cplusplus
 extern "C" {            // Prevents name mangling of functions
@@ -82,13 +79,6 @@ void InitAudioDevice(void);                                     // Initialize au
 void CloseAudioDevice(void);                                    // Close the audio device and context (and music stream)
 bool IsAudioDeviceReady(void);                                  // True if call to InitAudioDevice() was successful and CloseAudioDevice() has not been called yet
 
-// Audio contexts are for outputing custom audio waveforms, This will shut down any other sound sources currently playing
-// The mixChannel is what mix 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 InitAudioContext(48000, 0, 2, true); // mixchannel 1, 48khz, stereo, floating point
-AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint);
-void CloseAudioContext(AudioContext ctx);                       // Frees audio context
-unsigned short UpdateAudioContext(AudioContext ctx, void *data, unsigned short numberElements); // Pushes more audio data into context mix channel, if NULL is passed to data then zeros are played
-
 Sound LoadSound(char *fileName);                                // Load sound to memory
 Sound LoadSoundFromWave(Wave wave);                             // Load sound to memory from wave data
 Sound LoadSoundFromRES(const char *rresName, int resId);        // Load sound to memory from rRES file (raylib Resource)
@@ -112,6 +102,9 @@ float GetMusicTimePlayed(int index);                            // Get current m
 int getMusicStreamCount(void);
 void SetMusicPitch(int index, float pitch);
 
+RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint);
+void CloseRawAudioContext(RawAudioContext ctx);
+
 #ifdef __cplusplus
 }
 #endif
diff --git a/src/raylib.h b/src/raylib.h
index ea9fbfcb..6efde710 100644
--- a/src/raylib.h
+++ b/src/raylib.h
@@ -451,10 +451,7 @@ typedef struct Wave {
     short channels;
 } Wave;
 
-// Audio Context, used to create custom audio streams that are not bound to a sound file. There can be
-// no more than 4 concurrent audio contexts in use. This is due to each active context being tied to
-// a dedicated mix channel.
-typedef void* AudioContext;
+typedef int RawAudioContext;
 
 // Texture formats
 // NOTE: Support depends on OpenGL version and platform
@@ -876,13 +873,6 @@ void InitAudioDevice(void);                                     // Initialize au
 void CloseAudioDevice(void);                                    // Close the audio device and context (and music stream)
 bool IsAudioDeviceReady(void);                                  // True if call to InitAudioDevice() was successful and CloseAudioDevice() has not been called yet
 
-// Audio contexts are for outputing custom audio waveforms, This will shut down any other sound sources currently playing
-// The mixChannel is what mix 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 InitAudioContext(48000, 0, 2, true); // mixchannel 1, 48khz, stereo, floating point
-AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint);
-void CloseAudioContext(AudioContext ctx);                       // Frees audio context
-unsigned short UpdateAudioContext(AudioContext ctx, void *data, unsigned short numberElements); // Pushes more audio data into context mix channel, if NULL is passed to data then zeros are played
-
 Sound LoadSound(char *fileName);                                // Load sound to memory
 Sound LoadSoundFromWave(Wave wave);                             // Load sound to memory from wave data
 Sound LoadSoundFromRES(const char *rresName, int resId);        // Load sound to memory from rRES file (raylib Resource)
@@ -906,6 +896,9 @@ float GetMusicTimePlayed(int index);                            // Get current m
 int getMusicStreamCount(void);
 void SetMusicPitch(int index, float pitch);
 
+RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint); // used to output raw audio streams, returns negative numbers on error
+void CloseRawAudioContext(RawAudioContext ctx);
+
 #ifdef __cplusplus
 }
 #endif

From b10425492ac692d7c53001290e3d0b678df634b0 Mon Sep 17 00:00:00 2001
From: Joshua Reisenauer <kd7tck@msn.com>
Date: Thu, 19 May 2016 15:22:12 -0700
Subject: [PATCH 12/14] name correction

---
 src/audio.c | 24 ++++++++++++------------
 1 file changed, 12 insertions(+), 12 deletions(-)

diff --git a/src/audio.c b/src/audio.c
index 584d3ad1..6d8f876c 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -59,9 +59,9 @@
 //----------------------------------------------------------------------------------
 // Defines and Macros
 //----------------------------------------------------------------------------------
-#define MAX_STREAM_BUFFERS          2
-#define MAX_AUDIO_CONTEXTS          4             // Number of open AL sources
-#define MAX_MUSIC_STREAMS           2
+#define MAX_STREAM_BUFFERS          2             // Number of buffers for each alSource
+#define MAX_MIX_CHANNELS            4             // Number of open AL sources
+#define MAX_MUSIC_STREAMS           2             // Number of simultanious music sources
 
 #if defined(PLATFORM_RPI) || defined(PLATFORM_ANDROID)
     // NOTE: On RPI and Android should be lower to avoid frame-stalls
@@ -96,12 +96,12 @@ typedef struct MixChannel_t {
 typedef struct Music {
     stb_vorbis *stream;
     jar_xm_context_t *chipctx; // Stores jar_xm mixc
-    MixChannel_t *mixc; // mix channel
+    MixChannel_t *mixc;        // mix channel
     
     int totalSamplesLeft;
     float totalLengthSeconds;
     bool loop;
-    bool chipTune; // True if chiptune is loaded
+    bool chipTune;             // True if chiptune is loaded
 } Music;
 
 #if defined(AUDIO_STANDALONE)
@@ -111,7 +111,7 @@ typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType;
 //----------------------------------------------------------------------------------
 // Global Variables Definition
 //----------------------------------------------------------------------------------
-static MixChannel_t* mixChannelsActive_g[MAX_AUDIO_CONTEXTS];      // What mix channels are currently active
+static MixChannel_t* mixChannelsActive_g[MAX_MIX_CHANNELS];        // What mix channels are currently active
 static bool musicEnabled_g = false;
 static Music currentMusic[MAX_MUSIC_STREAMS];                      // Current music loaded, up to two can play at the same time
 
@@ -123,7 +123,7 @@ static Wave LoadOGG(char *fileName);               // Load OGG file
 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 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.
@@ -212,7 +212,7 @@ bool IsAudioDeviceReady(void)
 // 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)
 {
-    if(mixChannel >= MAX_AUDIO_CONTEXTS) return NULL;
+    if(mixChannel >= MAX_MIX_CHANNELS) return NULL;
     if(!IsAudioDeviceReady()) InitAudioDevice();
     
     if(!mixChannelsActive_g[mixChannel]){
@@ -380,10 +380,10 @@ 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_AUDIO_CONTEXTS; mixIndex++) // find empty mix channel slot
+    for(mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot
     {
         if(mixChannelsActive_g[mixIndex] == NULL) break;
-        else if(mixIndex = MAX_AUDIO_CONTEXTS - 1) return -1; // error
+        else if(mixIndex = MAX_MIX_CHANNELS - 1) return -1; // error
     }
     
     if(InitMixChannel(sampleRate, mixIndex, channels, floatingPoint))
@@ -755,10 +755,10 @@ int PlayMusicStream(int musicIndex, char *fileName)
     
     if(currentMusic[musicIndex].stream || currentMusic[musicIndex].chipctx) return 1; // error
     
-    for(mixIndex = 0; mixIndex < MAX_AUDIO_CONTEXTS; mixIndex++) // find empty mix channel slot
+    for(mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot
     {
         if(mixChannelsActive_g[mixIndex] == NULL) break;
-        else if(mixIndex = MAX_AUDIO_CONTEXTS - 1) return 2; // error
+        else if(mixIndex = MAX_MIX_CHANNELS - 1) return 2; // error
     }
     
     if (strcmp(GetExtension(fileName),"ogg") == 0)

From 41c5f3a0178027e9b74e563ab102f603a53e35bf Mon Sep 17 00:00:00 2001
From: Joshua Reisenauer <kd7tck@msn.com>
Date: Thu, 19 May 2016 20:44:09 -0700
Subject: [PATCH 13/14] Buffer for raw audio

---
 src/audio.c  | 22 ++++++++++++++++++----
 src/audio.h  |  4 ++++
 src/raylib.h |  6 +++++-
 3 files changed, 27 insertions(+), 5 deletions(-)

diff --git a/src/audio.c b/src/audio.c
index 6d8f876c..43e8be14 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -128,8 +128,8 @@ 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 unsigned short BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements); // Pushes more audio data into mixc mix channel, if NULL is passed it pauses
-static unsigned short FillAlBufferWithSilence(MixChannel_t *mixc, ALuint buffer); // Fill buffer with zeros, returns number processed
+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
 static int IsMusicStreamReadyForBuffering(int index); // Checks if music buffer is ready to be refilled
@@ -292,7 +292,7 @@ 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 unsigned short BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements)
+static int BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements)
 {
     if(!mixc || mixChannelsActive_g[mixc->mixChannel] != mixc) return 0; // when there is two channels there must be an even number of samples
     
@@ -331,7 +331,7 @@ static unsigned short BufferMixChannel(MixChannel_t* mixc, void *data, int numbe
 }
 
 // fill buffer with zeros, returns number processed
-static unsigned short FillAlBufferWithSilence(MixChannel_t *mixc, ALuint buffer)
+static int FillAlBufferWithSilence(MixChannel_t *mixc, ALuint buffer)
 {
     if(mixc->floatingPoint){
         float pcm[MUSIC_BUFFER_SIZE_FLOAT] = {0.f};
@@ -377,6 +377,7 @@ static void ResampleByteToFloat(char *chars, float *floats, unsigned short len)
 }
 
 // used to output raw audio streams, returns negative numbers on error
+// if floating point is false the data size is 16bit short, otherwise it is float 32bit
 RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint)
 {
     int mixIndex;
@@ -398,6 +399,19 @@ void CloseRawAudioContext(RawAudioContext ctx)
         CloseMixChannel(mixChannelsActive_g[ctx]);
 }
 
+int BufferRawAudioContext(RawAudioContext ctx, void *data, int numberElements)
+{
+    int numBuffered = 0;
+    if(ctx >= 0)
+    {
+        MixChannel_t* mixc = mixChannelsActive_g[ctx];
+        numBuffered = BufferMixChannel(mixc, data, numberElements);
+    }
+    return numBuffered;
+}
+
+
+
 
 
 //----------------------------------------------------------------------------------
diff --git a/src/audio.h b/src/audio.h
index d3276bf6..1140a60a 100644
--- a/src/audio.h
+++ b/src/audio.h
@@ -102,8 +102,12 @@ float GetMusicTimePlayed(int index);                            // Get current m
 int getMusicStreamCount(void);
 void SetMusicPitch(int index, float pitch);
 
+// used to output raw audio streams, returns negative numbers on error
+// if floating point is false the data size is 16bit short, otherwise it is float 32bit
 RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint);
+
 void CloseRawAudioContext(RawAudioContext ctx);
+int BufferRawAudioContext(RawAudioContext ctx, void *data, int numberElements); // returns number of elements buffered
 
 #ifdef __cplusplus
 }
diff --git a/src/raylib.h b/src/raylib.h
index 6efde710..986dc7bf 100644
--- a/src/raylib.h
+++ b/src/raylib.h
@@ -896,8 +896,12 @@ float GetMusicTimePlayed(int index);                            // Get current m
 int getMusicStreamCount(void);
 void SetMusicPitch(int index, float pitch);
 
-RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint); // used to output raw audio streams, returns negative numbers on error
+// used to output raw audio streams, returns negative numbers on error
+// if floating point is false the data size is 16bit short, otherwise it is float 32bit
+RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint);
+
 void CloseRawAudioContext(RawAudioContext ctx);
+int BufferRawAudioContext(RawAudioContext ctx, void *data, int numberElements); // returns number of elements buffered
 
 #ifdef __cplusplus
 }

From 179f2f9e4fd9ad43e120f186484a78984a2ac063 Mon Sep 17 00:00:00 2001
From: Joshua Reisenauer <kd7tck@msn.com>
Date: Thu, 19 May 2016 20:56:38 -0700
Subject: [PATCH 14/14] windows automated compile

Only works when raylib is installed on windows system.
---
 src/windows_compile.bat | 2 ++
 1 file changed, 2 insertions(+)
 create mode 100644 src/windows_compile.bat

diff --git a/src/windows_compile.bat b/src/windows_compile.bat
new file mode 100644
index 00000000..f1d0fb29
--- /dev/null
+++ b/src/windows_compile.bat
@@ -0,0 +1,2 @@
+set PATH=C:\raylib\MinGW\bin;%PATH%
+mingw32-make
\ No newline at end of file