--- a/misc/winutils/include/SDL_audio.h Tue Jan 21 22:38:13 2014 +0100
+++ b/misc/winutils/include/SDL_audio.h Tue Jan 21 22:43:06 2014 +0100
@@ -72,24 +72,24 @@
*
*/
typedef struct SDL_AudioSpec {
- int freq; /**< DSP frequency -- samples per second */
- Uint16 format; /**< Audio data format */
- Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */
- Uint8 silence; /**< Audio buffer silence value (calculated) */
- Uint16 samples; /**< Audio buffer size in samples (power of 2) */
- Uint16 padding; /**< Necessary for some compile environments */
- Uint32 size; /**< Audio buffer size in bytes (calculated) */
- /**
- * This function is called when the audio device needs more data.
- *
- * @param[out] stream A pointer to the audio data buffer
- * @param[in] len The length of the audio buffer in bytes.
- *
- * Once the callback returns, the buffer will no longer be valid.
- * Stereo samples are stored in a LRLRLR ordering.
- */
- void (SDLCALL *callback)(void *userdata, Uint8 *stream, int len);
- void *userdata;
+ int freq; /**< DSP frequency -- samples per second */
+ Uint16 format; /**< Audio data format */
+ Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */
+ Uint8 silence; /**< Audio buffer silence value (calculated) */
+ Uint16 samples; /**< Audio buffer size in samples (power of 2) */
+ Uint16 padding; /**< Necessary for some compile environments */
+ Uint32 size; /**< Audio buffer size in bytes (calculated) */
+ /**
+ * This function is called when the audio device needs more data.
+ *
+ * @param[out] stream A pointer to the audio data buffer
+ * @param[in] len The length of the audio buffer in bytes.
+ *
+ * Once the callback returns, the buffer will no longer be valid.
+ * Stereo samples are stored in a LRLRLR ordering.
+ */
+ void (SDLCALL *callback)(void *userdata, Uint8 *stream, int len);
+ void *userdata;
} SDL_AudioSpec;
/**
@@ -97,25 +97,25 @@
* defaults to LSB byte order
*/
/*@{*/
-#define AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */
-#define AUDIO_S8 0x8008 /**< Signed 8-bit samples */
-#define AUDIO_U16LSB 0x0010 /**< Unsigned 16-bit samples */
-#define AUDIO_S16LSB 0x8010 /**< Signed 16-bit samples */
-#define AUDIO_U16MSB 0x1010 /**< As above, but big-endian byte order */
-#define AUDIO_S16MSB 0x9010 /**< As above, but big-endian byte order */
-#define AUDIO_U16 AUDIO_U16LSB
-#define AUDIO_S16 AUDIO_S16LSB
+#define AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */
+#define AUDIO_S8 0x8008 /**< Signed 8-bit samples */
+#define AUDIO_U16LSB 0x0010 /**< Unsigned 16-bit samples */
+#define AUDIO_S16LSB 0x8010 /**< Signed 16-bit samples */
+#define AUDIO_U16MSB 0x1010 /**< As above, but big-endian byte order */
+#define AUDIO_S16MSB 0x9010 /**< As above, but big-endian byte order */
+#define AUDIO_U16 AUDIO_U16LSB
+#define AUDIO_S16 AUDIO_S16LSB
/**
* @name Native audio byte ordering
*/
/*@{*/
#if SDL_BYTEORDER == SDL_LIL_ENDIAN
-#define AUDIO_U16SYS AUDIO_U16LSB
-#define AUDIO_S16SYS AUDIO_S16LSB
+#define AUDIO_U16SYS AUDIO_U16LSB
+#define AUDIO_S16SYS AUDIO_S16LSB
#else
-#define AUDIO_U16SYS AUDIO_U16MSB
-#define AUDIO_S16SYS AUDIO_S16MSB
+#define AUDIO_U16SYS AUDIO_U16MSB
+#define AUDIO_S16SYS AUDIO_S16MSB
#endif
/*@}*/
@@ -124,17 +124,17 @@
/** A structure to hold a set of audio conversion filters and buffers */
typedef struct SDL_AudioCVT {
- int needed; /**< Set to 1 if conversion possible */
- Uint16 src_format; /**< Source audio format */
- Uint16 dst_format; /**< Target audio format */
- double rate_incr; /**< Rate conversion increment */
- Uint8 *buf; /**< Buffer to hold entire audio data */
- int len; /**< Length of original audio buffer */
- int len_cvt; /**< Length of converted audio buffer */
- int len_mult; /**< buffer must be len*len_mult big */
- double len_ratio; /**< Given len, final size is len*len_ratio */
- void (SDLCALL *filters[10])(struct SDL_AudioCVT *cvt, Uint16 format);
- int filter_index; /**< Current audio conversion function */
+ int needed; /**< Set to 1 if conversion possible */
+ Uint16 src_format; /**< Source audio format */
+ Uint16 dst_format; /**< Target audio format */
+ double rate_incr; /**< Rate conversion increment */
+ Uint8 *buf; /**< Buffer to hold entire audio data */
+ int len; /**< Length of original audio buffer */
+ int len_cvt; /**< Length of converted audio buffer */
+ int len_mult; /**< buffer must be len*len_mult big */
+ double len_ratio; /**< Given len, final size is len*len_ratio */
+ void (SDLCALL *filters[10])(struct SDL_AudioCVT *cvt, Uint16 format);
+ int filter_index; /**< Current audio conversion function */
} SDL_AudioCVT;
@@ -164,7 +164,7 @@
* structure pointed to by 'obtained'. If 'obtained' is NULL, the audio
* data passed to the callback function will be guaranteed to be in the
* requested format, and will be automatically converted to the hardware
- * audio format if necessary. This function returns -1 if it failed
+ * audio format if necessary. This function returns -1 if it failed
* to open the audio device, or couldn't set up the audio thread.
*
* The audio device starts out playing silence when it's opened, and should
@@ -178,9 +178,9 @@
extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained);
typedef enum {
- SDL_AUDIO_STOPPED = 0,
- SDL_AUDIO_PLAYING,
- SDL_AUDIO_PAUSED
+ SDL_AUDIO_STOPPED = 0,
+ SDL_AUDIO_PLAYING,
+ SDL_AUDIO_PAUSED
} SDL_audiostatus;
/** Get the current audio state */
@@ -199,24 +199,24 @@
* This function loads a WAVE from the data source, automatically freeing
* that source if 'freesrc' is non-zero. For example, to load a WAVE file,
* you could do:
- * @code SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); @endcode
+ * @code SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); @endcode
*
* If this function succeeds, it returns the given SDL_AudioSpec,
* filled with the audio data format of the wave data, and sets
* 'audio_buf' to a malloc()'d buffer containing the audio data,
* and sets 'audio_len' to the length of that audio buffer, in bytes.
- * You need to free the audio buffer with SDL_FreeWAV() when you are
+ * You need to free the audio buffer with SDL_FreeWAV() when you are
* done with it.
*
- * This function returns NULL and sets the SDL error message if the
- * wave file cannot be opened, uses an unknown data format, or is
+ * This function returns NULL and sets the SDL error message if the
+ * wave file cannot be opened, uses an unknown data format, or is
* corrupt. Currently raw and MS-ADPCM WAVE files are supported.
*/
extern DECLSPEC SDL_AudioSpec * SDLCALL SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len);
/** Compatibility convenience function -- loads a WAV from a file */
#define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
- SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)
+ SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)
/**
* This function frees data previously allocated with SDL_LoadWAV_RW()
@@ -232,8 +232,8 @@
* @return This function returns 0, or -1 if there was an error.
*/
extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT *cvt,
- Uint16 src_format, Uint8 src_channels, int src_rate,
- Uint16 dst_format, Uint8 dst_channels, int dst_rate);
+ Uint16 src_format, Uint8 src_channels, int src_rate,
+ Uint16 dst_format, Uint8 dst_channels, int dst_rate);
/**
* Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(),