SDL_sound_CoreAudio: SDL_sound.h comparison

comparison SDL_sound.h @ 3:fd6cd0e04e6f

First stab at implementation complete.

author	Ryan C. Gordon <icculus@icculus.org>
date	Mon, 17 Sep 2001 18:10:16 +0000
parents	508aac690b19
children	5ff1dce70aec

comparison

equal deleted inserted replaced

-:f58a664c1557
+:fd6cd0e04e6f
 *  into this library, and SDL_sound will take that data, in one of several
 *  popular formats, and decode it into raw waveform data in the format of
 *  your choice. This gives you a nice abstraction for getting sound into your
 *  game or application; just feed it to SDL_sound, and it will handle
 *  decoding and converting, so you can just pass it to your SDL audio
-*  callback (or whatever). Since it gets data from a SDL_RWops, you can get
+*  callback (or whatever). Since it gets data from an SDL_RWops, you can get
 *  the initial sound data from any number of sources: file, memory buffer,
 *  network connection, etc.
 *
 * As the name implies, this library depends on SDL: Simple Directmedia Layer,
 *  which is a powerful, free, and cross-platform multimedia library. It can
 *  be found at http://www.libsdl.org/
 *
 * Support is in place or planned for the following sound formats:
-*   - .WAV     (Microsoft WAVfile RIFF data, internal.)
+*   - .WAV  (Microsoft WAVfile RIFF data, internal.)
-*   - .VOC     (Creative Labs' Voice format, internal.)
+*   - .VOC  (Creative Labs' Voice format, internal.)
-*   - .MP3     (MPEG-1 layer 3 support, via the SMPEG library.)
+*   - .MP3  (MPEG-1 layer 3 support, via the SMPEG library.)
-*   - .MID     (MIDI music converted to Waveform data, via Timidity.)
+*   - .MID  (MIDI music converted to Waveform data, via Timidity.)
-*   - .MOD     (MOD files, via MikMod.)
+*   - .MOD  (MOD files, via MikMod.)
-*   - .OGG     (Ogg files, via Ogg Vorbis libraries.)
+*   - .OGG  (Ogg files, via Ogg Vorbis libraries.)
-*   - .RAWDATA (Raw sound data in any format, internal.)
+*   - .RAW  (Raw sound data in any format, internal.)
-*   - .CDA     (CD audio read into a sound buffer, internal.)
+*   - .CDA  (CD audio read into a sound buffer, internal.)
 *   - .AU
 *   - .AIFF
 *
 *   (...and more to come...)
 *
 /*
 * These are flags that are used in a Sound_Sample (below) to show various
 *  states.
 *
-*   To use: "if (sample->flags & SOUND_SAMPLEFLAGS_ERROR) { dosomething(); }"
+*   To use: "if (sample->flags & SOUND_SAMPLEFLAG_ERROR) { dosomething(); }"
 */
 typedef enum __SOUND_SAMPLEFLAGS__
 {
 SOUND_SAMPLEFLAG_NONE      = 0,       /* Null flag.                      */
 /* these are set at sample creation time... */
 SOUND_SAMPLEFLAG_NEEDSEEK  = 1,       /* SDL_RWops must be able to seek. */
-SOUND_SAMPLEFLAG_STREAMING = 1 << 1,  /* source is streaming (no EOF).   */
 /* these are set during decoding... */
 SOUND_SAMPLEFLAG_EOF       = 1 << 29, /* end of input stream.            */
 SOUND_SAMPLEFLAG_ERROR     = 1 << 30, /* unrecoverable error.            */
-SOUND_SAMPLEFLAG_AGAIN     = 1 << 31  /* couldn't read without blocking. */
+SOUND_SAMPLEFLAG_EAGAIN    = 1 << 31  /* would block, or temp error.     */
 } Sound_SampleFlags;
+/*
+* Basics of a decoded sample's data structure: data format (see AUDIO_U8 and
+*  friends in SDL_audio.h), number of channels, and sample rate. If you need
+*  more explanation than that, you should stop developing sound code right
+*  now.
+*/
+typedef struct __SOUND_AUDIOINFO__
+{
+Uint16 format;
+Uint8 channels;
+Uint32 rate;
+} Sound_AudioInfo;
+/*
+* Each decoder sets up one of these structs, which can be retrieved via
+*  the Sound_AvailableDecoders() function. EVERY FIELD IN THIS IS READ-ONLY.
+*/
+typedef struct __SOUND_DECODERINFO__
+{
+const char *extension;      /* standard file extension. "MP3", "WAV"... */
+const char *description;    /* Human readable description of decoder.   */
+const char *author;         /* "Name Of Author <email@emailhost.dom>"   */
+const char *url;            /* URL specific to this decoder.            */
+} Sound_DecoderInfo;
 /*
 * The Sound_Sample structure is the heart of SDL_sound. This holds
 *  information about a source of sound data as it is being decoded.
 *  change them.
 */
 typedef struct __SOUND_SAMPLE__
 {
 void *opaque;                /* Internal use only.                   */
-Sound_DecoderInfo *decoder;  /* Decoder used for this sample.        */
+const Sound_DecoderInfo *decoder;  /* Decoder used for this sample.  */
-SDL_AudioSpec desired;       /* Desired audio format for conversion. */
+Sound_AudioInfo desired;     /* Desired audio format for conversion. */
-SDL_AudioSpec actual;        /* Actual audio format of sample.       */
+Sound_AudioInfo actual;      /* Actual audio format of sample.       */
 void *buffer;                /* Decoded sound data lands in here.    */
 Uint32 buffer_size;          /* Current size of (buffer), in bytes.  */
 Sound_SampleFlags flags;     /* Flags relating to this sample.       */
 } Sound_Sample;
-/*
-* Each decoder sets up one of these structs, which can be retrieved via
-*  the Sound_AvailableDecoders() function.
-*/
-typedef struct __PHYSFS_ARCHIVEINFO__
-{
-const char *extension;      /* Case sensitive standard file extension. */
-const char *description;    /* Human readable description of decoder.  */
-const char *author;         /* "Name Of Author (email@emailhost.dom)"  */
-const char *url;            /* URL specific to this decoder.           */
-} PHYSFS_ArchiveInfo;
 /*
 * Just what it says: a x.y.z style version number...
 */
 *  sound sources, and frees any resources in use by SDL_sound.
 *
 * All Sound_Sample pointers you had prior to this call are INVALIDATED.
 *
 * Once successfully deinitialized, Sound_Init() can be called again to
-*  restart the subsystem. All defaults API states are restored at this
+*  restart the subsystem. All default API states are restored at this
 *  point.
 *
 * You should call this BEFORE SDL_Quit(). This will NOT call SDL_Quit()
 *  for you!
 *
 *  with that thread. It is safe to call this function at anytime, even
 *  before Sound_Init().
 *
 *   @return READ ONLY string of last error message.
 */
-extern DECLSPEC const char *Sound_GetLastError(void);
+extern DECLSPEC const char *Sound_GetError(void);
+/**
+* Clear the current error message, so the next call to Sound_GetError() will
+*  return NULL.
+*/
+extern DECLSPEC void Sound_ClearError(void);
 /**
 * Start decoding a new sound sample. The data is read via an SDL_RWops
 *  structure (see SDL_rwops.h in the SDL include directory), so it may be
 *  coming from memory, disk, network stream, etc. The (ext) parameter is
 *  merely a hint to determining the correct decoder; if you specify, for
 *  example, "mp3" for an extension, and one of the decoders lists that
-*  (case sensitive) as a handled extension, then that decoder is given
+*  as a handled extension, then that decoder is given first shot at trying
-*  first shot at trying to claim the data for decoding. If none of the
+*  to claim the data for decoding. If none of the extensions match (or the
-*  extensions match (or the extension is NULL), then every decoder examines
+*  extension is NULL), then every decoder examines the data to determine if
-*  the data to determine if it can handle it, until one accepts it.
+*  it can handle it, until one accepts it. In such a case your SDL_RWops will
+*  need to be capable of rewinding to the start of the stream.
 * If no decoders can handle the data, a NULL value is returned, and a human
 *  readable error message can be fetched from Sound_GetLastError().
 * Optionally, a desired audio format can be specified. If the incoming data
 *  is in a different format, SDL_sound will convert it to the desired format
 *  on the fly. Note that this can be an expensive operation, so it may be
 *  make sure your data is initially in the format that you need it in.
 *  If you don't want to convert the data, you can specify NULL for a desired
 *  format. The incoming format of the data, preconversion, can be found
 *  in the Sound_Sample structure.
 * Note that the raw sound data "decoder" needs you to specify both the
-*  extension "RAWDATA" and a "desired" format, or it will refuse to handle
+*  extension "RAW" and a "desired" format, or it will refuse to handle
-*  the data.
+*  the data. This is to prevent it from catching all formats unsupported
+*  by the other decoders.
 * Finally, specify an initial buffer size; this is the number of bytes that
 *  will be allocated to store each read from the sound buffer. The more you
 *  can safely allocate, the more decoding can be done in one block, but the
 *  more resources you have to use up, and the longer each decoding call will
 *  take. Note that different data formats require more or less space to
 *  store. This buffer can be resized via Sound_SetBufferSize() ...
-*
+* The buffer size specified must be a multiple of the size of a single
+*  sample (not to be confused with a single Sound_Sample). So, if you want
+*  16-bit, stereo samples, then your sample size is (2 channels * 16 bits),
+*  or 32 bits per sample, which is four bytes. In such a case, you could
+*  specify 128 or 132 bytes for a buffer, but not 129, 130, or 131, although
+*  in reality, you'll want to specify a MUCH larger buffer.
 * When you are done with this Sound_Sample pointer, you can dispose of it
 *  via Sound_FreeSample().
+* You do not have to keep a reference to (rw) around. If this function
+*  suceeds, it stores (rw) internally (and disposes of it during the call
+*  to Sound_FreeSample()). If this function fails, it will dispose of the
+*  SDL_RWops for you.
 *
 *    @param rw SDL_RWops with sound data.
 *    @param ext File extension normally associated with a data format.
 *               Can usually be NULL.
 *    @param desired Format to convert sound data into. Can usually be NULL,
 *   @return Sound_Sample pointer, which is used as a handle to several other
 *            SDL_sound APIs. NULL on error. If error, use
 *            Sound_GetLastError() to see what went wrong.
 */
 extern DECLSPEC Sound_Sample *Sound_NewSample(SDL_RWops *rw, const char *ext,
-SDL_AudioInfo *desired,
+Sound_AudioInfo *desired,
 Uint32 bufferSize);
+/**
+* This is identical to Sound_NewSample(), but it creates an SDL_RWops for you
+*  from the file located in (filename). Note that (filename) is specified in
+*  platform-dependent notation. ("C:\\music\\mysong.mp3" on windows, and
+*  "/home/icculus/music/mysong.mp3" or whatever on Unix, etc.)
+* Sound_NewSample()'s "ext" parameter is gleaned from the contents of
+*  (filename).
+*
+*    @param filename file containing sound data.
+*    @param desired Format to convert sound data into. Can usually be NULL,
+*                   if you don't need conversion.
+*    @param bufferSize size, in bytes, of initial read buffer.
+*   @return Sound_Sample pointer, which is used as a handle to several other
+*            SDL_sound APIs. NULL on error. If error, use
+*            Sound_GetLastError() to see what went wrong.
+*/
+extern DECLSPEC Sound_Sample *Sound_NewSampleFromFile(const char *filename,
+Sound_AudioInfo *desired,
+Uint32 bufferSize);
 /**
 * Dispose of a Sound_Sample pointer that was returned from Sound_NewSample().
 *  This will also close/dispose of the SDL_RWops that was used at creation
 *  time, so there's no need to keep a reference to that around.
 *  certainly result in a crash if you attempt to keep using it.
 *
 *    @param sample The Sound_Sample to delete.
 */
 extern DECLSPEC void Sound_FreeSample(Sound_Sample *sample);
+/**
+* Change the current buffer size for a sample. If the buffer size could
+*  be changed, then the sample->buffer and sample->buffer_size fields will
+*  reflect that. If they could not be changed, then your original sample
+*  state is preserved. If the buffer is shrinking, the data at the end of
+*  buffer is truncated. If the buffer is growing, the contents of the new
+*  space at the end is undefined until you decode more into it or initialize
+*  it yourself.
+*
+* The buffer size specified must be a multiple of the size of a single
+*  sample (not to be confused with a single Sound_Sample). So, if you want
+*  16-bit, stereo samples, then your sample size is (2 channels * 16 bits),
+*  or 32 bits per sample, which is four bytes. In such a case, you could
+*  specify 128 or 132 bytes for a buffer, but not 129, 130, or 131, although
+*  in reality, you'll want to specify a MUCH larger buffer.
+*
+*    @param sample The Sound_Sample whose buffer to modify.
+*    @param new_size The desired size, in bytes, of the new buffer.
+*   @return non-zero if buffer size changed, zero on failure.
+*/
+extern DECLSPEC int Sound_SetBufferSize(Sound_Sample *sample, Uint32 new_size);
 /**
 * Decode more of the sound data in a Sound_Sample. It will decode at most
 *  sample->buffer_size bytes into sample->buffer in the desired format, and

Mercurial > SDL_sound_CoreAudio

comparison SDL_sound.h @ 3:fd6cd0e04e6f