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Fix minor errors with the OpenAL backend.

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This commit is contained in:
David Reid 2017-11-14 21:44:57 +10:00
parent 88d2810fa3
commit 322d868841
1 changed files with 22 additions and 21 deletions
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43
src/audio.c
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@ -82,9 +82,9 @@
#include "utils.h" // Required for: fopen() Android mapping
#endif
//#if USE_MINI_AL
#include "external/mini_al.h" // Implemented in mini_al.c. Cannot implement this here because it conflicts with Win32 APIs such as CloseWindow(), etc.
//#else
#include "external/mini_al.h" // Implemented in mini_al.c. Cannot implement this here because it conflicts with Win32 APIs such as CloseWindow(), etc.
#if !defined(USE_MINI_AL) || USE_MINI_AL == 0
#if defined(__APPLE__)
#include "OpenAL/al.h" // OpenAL basic header
#include "OpenAL/alc.h" // OpenAL context header (like OpenGL, OpenAL requires a context to work)
@ -96,7 +96,7 @@
// OpenAL extension: AL_EXT_FLOAT32 - Support for 32bit float samples
// OpenAL extension: AL_EXT_MCFORMATS - Support for multi-channel formats (Quad, 5.1, 6.1, 7.1)
//#endif
#endif
#include <stdlib.h> // Required for: malloc(), free()
#include <string.h> // Required for: strcmp(), strncmp()
@ -176,22 +176,6 @@ typedef struct MusicData {
unsigned int samplesLeft; // Number of samples left to end
} MusicData;
// AudioStreamData
typedef struct AudioStreamData AudioStreamData;
struct AudioStreamData {
mal_dsp dsp; // AudioStream data needs to flow through a persistent conversion pipeline. Not doing this will result in glitches between buffer updates.
float volume;
float pitch;
bool playing;
bool paused;
bool isSubBufferProcessed[2];
unsigned int frameCursorPos;
unsigned int bufferSizeInFrames;
AudioStreamData* next;
AudioStreamData* prev;
unsigned char buffer[1];
};
#if defined(AUDIO_STANDALONE)
typedef enum { LOG_INFO = 0, LOG_ERROR, LOG_WARNING, LOG_DEBUG, LOG_OTHER } TraceLogType;
#endif
@ -245,6 +229,22 @@ struct SoundData
mal_uint8 data[1]; // Raw audio data.
};
// AudioStreamData
typedef struct AudioStreamData AudioStreamData;
struct AudioStreamData {
mal_dsp dsp; // AudioStream data needs to flow through a persistent conversion pipeline. Not doing this will result in glitches between buffer updates.
float volume;
float pitch;
bool playing;
bool paused;
bool isSubBufferProcessed[2];
unsigned int frameCursorPos;
unsigned int bufferSizeInFrames;
AudioStreamData* next;
AudioStreamData* prev;
unsigned char buffer[1];
};
static mal_context context;
static mal_device device;
static mal_bool32 isAudioInitialized = MAL_FALSE;
@ -1594,7 +1594,7 @@ float GetMusicTimePlayed(Music music)
return secondsPlayed;
}
#if USE_MINI_AL
static mal_uint32 UpdateAudioStream_OnDSPRead(mal_dsp* pDSP, mal_uint32 frameCount, void* pFramesOut, void* pUserData)
{
AudioStreamData* internalData = (AudioStreamData*)pUserData;
@ -1652,6 +1652,7 @@ static mal_uint32 UpdateAudioStream_OnDSPRead(mal_dsp* pDSP, mal_uint32 frameCou
return frameCount;
}
#endif
// Init audio stream (to stream audio pcm data)
AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels)