gosu 0.15.1 → 1.1.0.pre1

data/dependencies/SDL_sound/SDL_sound_au.c

@@ -0,0 +1,352 @@
+/**
+ * SDL_sound; An abstract sound format decoding API.
+ *
+ * Please see the file LICENSE.txt in the source's root directory.
+ *
+ *  This file written by Mattias Engdegård.
+ */
+
+/*
+ * Sun/NeXT .au decoder for SDL_sound.
+ * Formats supported: 8 and 16 bit linear PCM, 8 bit µ-law.
+ * Files without valid header are assumed to be 8 bit µ-law, 8kHz, mono.
+ */
+
+#define __SDL_SOUND_INTERNAL__
+#include "SDL_sound_internal.h"
+
+#if SOUND_SUPPORTS_AU
+
+/* no init/deinit needed */
+static int AU_init(void)
+{
+    return 1;
+} /* AU_init */
+
+static void AU_quit(void)
+{
+    /* no-op. */
+} /* AU_quit */
+
+struct au_file_hdr
+{
+    Uint32 magic;
+    Uint32 hdr_size;
+    Uint32 data_size;
+    Uint32 encoding;
+    Uint32 sample_rate;
+    Uint32 channels;
+};
+
+#define HDR_SIZE 24
+
+enum
+{
+    AU_ENC_ULAW_8       = 1,        /* 8-bit ISDN µ-law */
+    AU_ENC_LINEAR_8     = 2,        /* 8-bit linear PCM */
+    AU_ENC_LINEAR_16    = 3,        /* 16-bit linear PCM */
+
+    /* the rest are unsupported (I have never seen them in the wild) */
+    AU_ENC_LINEAR_24    = 4,        /* 24-bit linear PCM */
+    AU_ENC_LINEAR_32    = 5,        /* 32-bit linear PCM  */
+    AU_ENC_FLOAT        = 6,        /* 32-bit IEEE floating point */
+    AU_ENC_DOUBLE       = 7,        /* 64-bit IEEE floating point */
+    /* more Sun formats, not supported either */
+    AU_ENC_ADPCM_G721   = 23,
+    AU_ENC_ADPCM_G722   = 24,
+    AU_ENC_ADPCM_G723_3 = 25,
+    AU_ENC_ADPCM_G723_5 = 26,
+    AU_ENC_ALAW_8       = 27
+};
+
+struct audec
+{
+    Uint32 total;
+    Uint32 remaining;
+    Uint32 start_offset;
+    int encoding;
+};
+
+
+/*
+ * Read in the AU header from disk. This makes this process safe
+ *  regardless of the processor's byte order or how the au_file_hdr
+ *  structure is packed.
+ */
+static int read_au_header(SDL_RWops *rw, struct au_file_hdr *hdr)
+{
+    if (SDL_RWread(rw, &hdr->magic, sizeof (hdr->magic), 1) != 1)
+        return 0;
+    hdr->magic = SDL_SwapBE32(hdr->magic);
+
+    if (SDL_RWread(rw, &hdr->hdr_size, sizeof (hdr->hdr_size), 1) != 1)
+        return 0;
+    hdr->hdr_size = SDL_SwapBE32(hdr->hdr_size);
+
+    if (SDL_RWread(rw, &hdr->data_size, sizeof (hdr->data_size), 1) != 1)
+        return 0;
+    hdr->data_size = SDL_SwapBE32(hdr->data_size);
+
+    if (SDL_RWread(rw, &hdr->encoding, sizeof (hdr->encoding), 1) != 1)
+        return 0;
+    hdr->encoding = SDL_SwapBE32(hdr->encoding);
+
+    if (SDL_RWread(rw, &hdr->sample_rate, sizeof (hdr->sample_rate), 1) != 1)
+        return 0;
+    hdr->sample_rate = SDL_SwapBE32(hdr->sample_rate);
+
+    if (SDL_RWread(rw, &hdr->channels, sizeof (hdr->channels), 1) != 1)
+        return 0;
+    hdr->channels = SDL_SwapBE32(hdr->channels);
+
+    return 1;
+} /* read_au_header */
+
+
+#define AU_MAGIC 0x2E736E64  /* ".snd", in ASCII (bigendian number) */
+
+static int AU_open(Sound_Sample *sample, const char *ext)
+{
+    Sound_SampleInternal *internal = sample->opaque;
+    SDL_RWops *rw = internal->rw;
+    int skip, hsize, i, bytes_per_second;
+    struct au_file_hdr hdr;
+    struct audec *dec;
+    char c;
+
+    /* read_au_header() will do byte order swapping. */
+    BAIL_IF_MACRO(!read_au_header(rw, &hdr), "AU: bad header", 0);
+
+    dec = SDL_malloc(sizeof *dec);
+    BAIL_IF_MACRO(dec == NULL, ERR_OUT_OF_MEMORY, 0);
+    internal->decoder_private = dec;
+
+    if (hdr.magic == AU_MAGIC)
+    {
+        /* valid magic */
+        dec->encoding = hdr.encoding;
+        switch(dec->encoding)
+        {
+            case AU_ENC_ULAW_8:
+                /* Convert 8-bit µ-law to 16-bit linear on the fly. This is
+                   slightly wasteful if the audio driver must convert them
+                   back, but µ-law only devices are rare (mostly _old_ Suns) */
+                sample->actual.format = AUDIO_S16SYS;
+                break;
+
+            case AU_ENC_LINEAR_8:
+                sample->actual.format = AUDIO_S8;
+                break;
+
+            case AU_ENC_LINEAR_16:
+                sample->actual.format = AUDIO_S16MSB;
+                break;
+
+            default:
+                SDL_free(dec);
+                BAIL_MACRO("AU: Unsupported .au encoding", 0);
+        } /* switch */
+
+        sample->actual.rate = hdr.sample_rate;
+        sample->actual.channels = hdr.channels;
+        dec->remaining = hdr.data_size;
+        hsize = hdr.hdr_size;
+
+        /* skip remaining part of header (input may be unseekable) */
+        for (i = HDR_SIZE; i < hsize; i++)
+        {
+            if (SDL_RWread(rw, &c, 1, 1) != 1)
+            {
+                SDL_free(dec);
+                BAIL_MACRO(ERR_IO_ERROR, 0);
+            } /* if */
+        } /* for */
+    } /* if */
+
+    else if (SDL_strcasecmp(ext, "au") == 0)
+    {
+        /*
+         * A number of files in the wild have the .au extension but no valid
+         * header; these are traditionally assumed to be 8kHz µ-law. Handle
+         * them here only if the extension is recognized.
+         */
+
+        SNDDBG(("AU: Invalid header, assuming raw 8kHz µ-law.\n"));
+        /* if seeking fails, we lose 24 samples. big deal */
+        SDL_RWseek(rw, -HDR_SIZE, SEEK_CUR);
+        dec->encoding = AU_ENC_ULAW_8;
+        dec->remaining = (Uint32)-1; 		/* no limit */
+        sample->actual.format = AUDIO_S16SYS;
+        sample->actual.rate = 8000;
+        sample->actual.channels = 1;
+    } /* else if */
+
+    else
+    {
+        SDL_free(dec);
+        BAIL_MACRO("AU: Not an .AU stream.", 0);
+    } /* else */    
+
+    bytes_per_second = ( ( dec->encoding == AU_ENC_LINEAR_16 ) ? 2 : 1 )
+        * sample->actual.rate * sample->actual.channels ;
+    internal->total_time = ((dec->remaining == -1) ? (-1) :
+                            ( ( dec->remaining / bytes_per_second ) * 1000 ) +
+                            ( ( dec->remaining % bytes_per_second ) * 1000 /
+                              bytes_per_second ) );
+
+    sample->flags = SOUND_SAMPLEFLAG_CANSEEK;
+    dec->total = dec->remaining;
+    dec->start_offset = SDL_RWtell(rw);
+
+    SNDDBG(("AU: Accepting data stream.\n"));
+    return 1;
+} /* AU_open */
+
+
+static void AU_close(Sound_Sample *sample)
+{
+    Sound_SampleInternal *internal = sample->opaque;
+    SDL_free(internal->decoder_private);
+} /* AU_close */
+
+
+/* table to convert from µ-law encoding to signed 16-bit samples,
+   generated by a throwaway perl script */
+static Sint16 ulaw_to_linear[256] = {
+    -32124,-31100,-30076,-29052,-28028,-27004,-25980,-24956,
+    -23932,-22908,-21884,-20860,-19836,-18812,-17788,-16764,
+    -15996,-15484,-14972,-14460,-13948,-13436,-12924,-12412,
+    -11900,-11388,-10876,-10364, -9852, -9340, -8828, -8316,
+     -7932, -7676, -7420, -7164, -6908, -6652, -6396, -6140,
+     -5884, -5628, -5372, -5116, -4860, -4604, -4348, -4092,
+     -3900, -3772, -3644, -3516, -3388, -3260, -3132, -3004,
+     -2876, -2748, -2620, -2492, -2364, -2236, -2108, -1980,
+     -1884, -1820, -1756, -1692, -1628, -1564, -1500, -1436,
+     -1372, -1308, -1244, -1180, -1116, -1052,  -988,  -924,
+      -876,  -844,  -812,  -780,  -748,  -716,  -684,  -652,
+      -620,  -588,  -556,  -524,  -492,  -460,  -428,  -396,
+      -372,  -356,  -340,  -324,  -308,  -292,  -276,  -260,
+      -244,  -228,  -212,  -196,  -180,  -164,  -148,  -132,
+      -120,  -112,  -104,   -96,   -88,   -80,   -72,   -64,
+       -56,   -48,   -40,   -32,   -24,   -16,    -8,     0,
+     32124, 31100, 30076, 29052, 28028, 27004, 25980, 24956,
+     23932, 22908, 21884, 20860, 19836, 18812, 17788, 16764,
+     15996, 15484, 14972, 14460, 13948, 13436, 12924, 12412,
+     11900, 11388, 10876, 10364,  9852,  9340,  8828,  8316,
+      7932,  7676,  7420,  7164,  6908,  6652,  6396,  6140,
+      5884,  5628,  5372,  5116,  4860,  4604,  4348,  4092,
+      3900,  3772,  3644,  3516,  3388,  3260,  3132,  3004,
+      2876,  2748,  2620,  2492,  2364,  2236,  2108,  1980,
+      1884,  1820,  1756,  1692,  1628,  1564,  1500,  1436,
+      1372,  1308,  1244,  1180,  1116,  1052,   988,   924,
+       876,   844,   812,   780,   748,   716,   684,   652,
+       620,   588,   556,   524,   492,   460,   428,   396,
+       372,   356,   340,   324,   308,   292,   276,   260,
+       244,   228,   212,   196,   180,   164,   148,   132,
+       120,   112,   104,    96,    88,    80,    72,    64,
+        56,    48,    40,    32,    24,    16,     8,     0
+};
+
+
+static Uint32 AU_read(Sound_Sample *sample)
+{
+    int ret;
+    Sound_SampleInternal *internal = sample->opaque;
+    struct audec *dec = internal->decoder_private;
+    int maxlen;
+    Uint8 *buf;
+
+    maxlen = internal->buffer_size;
+    buf = internal->buffer;
+    if (dec->encoding == AU_ENC_ULAW_8)
+    {
+        /* We read µ-law samples into the second half of the buffer, so
+           we can expand them to 16-bit samples afterwards */
+        maxlen >>= 1;
+        buf += maxlen;
+    } /* if */
+
+    if (maxlen > dec->remaining)
+        maxlen = dec->remaining;
+    ret = SDL_RWread(internal->rw, buf, 1, maxlen);
+    if (ret == 0)
+        sample->flags |= SOUND_SAMPLEFLAG_EOF;
+    else if (ret == -1)
+        sample->flags |= SOUND_SAMPLEFLAG_ERROR;
+    else
+    {
+        dec->remaining -= ret;
+        if (ret < maxlen)
+            sample->flags |= SOUND_SAMPLEFLAG_EAGAIN;
+
+        if (dec->encoding == AU_ENC_ULAW_8)
+        {
+            int i;
+            Sint16 *dst = internal->buffer;
+            for (i = 0; i < ret; i++)
+                dst[i] = ulaw_to_linear[buf[i]];
+            ret <<= 1;                  /* return twice as much as read */
+        } /* if */
+    } /* else */
+
+    return ret;
+} /* AU_read */
+
+
+static int AU_rewind(Sound_Sample *sample)
+{
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    struct audec *dec = (struct audec *) internal->decoder_private;
+    int rc = SDL_RWseek(internal->rw, dec->start_offset, SEEK_SET);
+    BAIL_IF_MACRO(rc != dec->start_offset, ERR_IO_ERROR, 0);
+    dec->remaining = dec->total;
+    return 1;
+} /* AU_rewind */
+
+
+static int AU_seek(Sound_Sample *sample, Uint32 ms)
+{
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    struct audec *dec = (struct audec *) internal->decoder_private;
+    int offset = __Sound_convertMsToBytePos(&sample->actual, ms);
+    int rc;
+    int pos;
+
+    if (dec->encoding == AU_ENC_ULAW_8)
+        offset >>= 1;  /* halve the byte offset for compression. */
+
+    pos = (int) (dec->start_offset + offset);
+    rc = SDL_RWseek(internal->rw, pos, SEEK_SET);
+    BAIL_IF_MACRO(rc != pos, ERR_IO_ERROR, 0);
+    dec->remaining = dec->total - offset;
+    return 1;
+} /* AU_seek */
+
+/*
+ * Sometimes the extension ".snd" is used for these files (mostly on the NeXT),
+ * and the magic number comes from this. However it may clash with other
+ * formats and is somewhat of an anachronism, so only .au is used here.
+ */
+static const char *extensions_au[] = { "AU", NULL };
+const Sound_DecoderFunctions __Sound_DecoderFunctions_AU =
+{
+    {
+        extensions_au,
+        "Sun/NeXT audio file format",
+        "Mattias Engdegård <f91-men@nada.kth.se>",
+        "https://icculus.org/SDL_sound/"
+    },
+
+    AU_init,        /*   init() method */
+    AU_quit,        /*   quit() method */
+    AU_open,        /*   open() method */
+    AU_close,       /*  close() method */
+    AU_read,        /*   read() method */
+    AU_rewind,      /* rewind() method */
+    AU_seek         /*   seek() method */
+};
+
+#endif /* SOUND_SUPPORTS_AU */
+
+/* end of SDL_sound_au.c ... */
+

data/dependencies/SDL_sound/SDL_sound_coreaudio.c

@@ -0,0 +1,747 @@
+/**
+ * SDL_sound; An abstract sound format decoding API.
+ *
+ * Please see the file LICENSE.txt in the source's root directory.
+ *
+ *  This file written by Eric Wing.
+ */
+
+#define __SDL_SOUND_INTERNAL__
+#include "SDL_sound_internal.h"
+
+#if SOUND_SUPPORTS_COREAUDIO
+
+#include <stddef.h> /* NULL */
+#include <arpa/inet.h> /* htonl */
+#include <AudioToolbox/AudioToolbox.h>
+
+typedef struct CoreAudioFileContainer
+{
+	AudioFileID* audioFileID;
+	ExtAudioFileRef extAudioFileRef;
+	AudioStreamBasicDescription* outputFormat;
+} CoreAudioFileContainer;
+
+
+static int CoreAudio_init(void)
+{
+    return 1;  /* always succeeds. */
+} /* CoreAudio_init */
+
+
+static void CoreAudio_quit(void)
+{
+    /* it's a no-op. */
+} /* CoreAudio_quit */
+
+/*
+   https://developer.apple.com/library/ios/#documentation/MusicAudio/Reference/AudioFileConvertRef/Reference/reference.html
+   kAudioFileAIFFType = 'AIFF',
+   kAudioFileAIFCType            = 'AIFC',
+   kAudioFileWAVEType            = 'WAVE',
+   kAudioFileSoundDesigner2Type  = 'Sd2f',
+   kAudioFileNextType            = 'NeXT',
+   kAudioFileMP3Type             = 'MPG3',
+   kAudioFileMP2Type             = 'MPG2',
+   kAudioFileMP1Type             = 'MPG1',
+   kAudioFileAC3Type             = 'ac-3',
+   kAudioFileAAC_ADTSType        = 'adts',
+   kAudioFileMPEG4Type           = 'mp4f',
+   kAudioFileM4AType             = 'm4af',
+   kAudioFileCAFType             = 'caff',
+   kAudioFile3GPType             = '3gpp',
+   kAudioFile3GP2Type            = '3gp2',
+   kAudioFileAMRType             = 'amrf'
+*/
+static AudioFileTypeID CoreAudio_GetAudioTypeForExtension(const char* file_extension)
+{
+	if( (SDL_strcasecmp(file_extension, "aif") == 0)
+		|| (SDL_strcasecmp(file_extension, "aiff") == 0)
+		|| (SDL_strcasecmp(file_extension, "aifc") == 0)
+	)
+	{
+		return kAudioFileAIFCType;
+	}
+	else if( (SDL_strcasecmp(file_extension, "wav") == 0)
+		|| (SDL_strcasecmp(file_extension, "wave") == 0)
+	)
+	{
+		return kAudioFileWAVEType;
+	}
+	else if( (SDL_strcasecmp(file_extension, "mp3") == 0)
+	)
+	{
+		return kAudioFileMP3Type;
+	}
+	else if( (SDL_strcasecmp(file_extension, "mp4") == 0)
+	)
+	{
+		return kAudioFileMPEG4Type;
+	}
+	else if( (SDL_strcasecmp(file_extension, "m4a") == 0)
+	)
+	{
+		return kAudioFileM4AType;
+	}
+	else if( (SDL_strcasecmp(file_extension, "aac") == 0)
+	)
+	{
+		return kAudioFileAAC_ADTSType;
+	}
+	else if( (SDL_strcasecmp(file_extension, "aac") == 0)
+	)
+	{
+		return kAudioFileAAC_ADTSType;
+	}
+	else if( (SDL_strcasecmp(file_extension, "caf") == 0)
+		|| (SDL_strcasecmp(file_extension, "caff") == 0)
+	)
+	{
+		return kAudioFileCAFType;
+	}
+	else if( (SDL_strcasecmp(file_extension, "Sd2f") == 0)
+		|| (SDL_strcasecmp(file_extension, "sd2") == 0)
+	)
+	{
+		return kAudioFileSoundDesigner2Type;
+	}
+	else if( (SDL_strcasecmp(file_extension, "au") == 0)
+		|| (SDL_strcasecmp(file_extension, "next") == 0)
+	)
+	{
+		return kAudioFileNextType;
+	}
+	else if( (SDL_strcasecmp(file_extension, "mp2") == 0)
+	)
+	{
+		return kAudioFileMP2Type;
+	}
+	else if( (SDL_strcasecmp(file_extension, "mp1") == 0)
+	)
+	{
+		return kAudioFileMP1Type;
+	}
+	else if( (SDL_strcasecmp(file_extension, "ac3") == 0)
+	)
+	{
+		return kAudioFileAC3Type;
+	}
+	else if( (SDL_strcasecmp(file_extension, "3gpp") == 0)
+	)
+	{
+		return kAudioFile3GPType;
+	}
+	else if( (SDL_strcasecmp(file_extension, "3gp2") == 0)
+	)
+	{
+		return kAudioFile3GP2Type;
+	}
+	else if( (SDL_strcasecmp(file_extension, "amrf") == 0)
+		|| (SDL_strcasecmp(file_extension, "amr") == 0)
+	)
+	{
+		return kAudioFileAMRType;
+	}
+	else if( (SDL_strcasecmp(file_extension, "ima4") == 0)
+		|| (SDL_strcasecmp(file_extension, "ima") == 0)
+	)
+	{
+		/* not sure about this one */
+		return kAudioFileCAFType;
+	}
+	else
+	{
+		return 0;
+	}
+
+}
+
+static const char* CoreAudio_FourCCToString(int32_t error_code)
+{
+	static char return_string[16];
+	uint32_t big_endian_code = htonl(error_code);
+	char* big_endian_str = (char*)&big_endian_code;
+	// see if it appears to be a 4-char-code
+	if(isprint(big_endian_str[0])
+	   && isprint(big_endian_str[1])
+	   && isprint(big_endian_str[2])
+	   && isprint (big_endian_str[3]))
+	{
+		return_string[0] = '\'';
+		return_string[1] = big_endian_str[0];
+		return_string[2] = big_endian_str[1];
+		return_string[3] = big_endian_str[2];
+		return_string[4] = big_endian_str[3];
+		return_string[5] = '\'';
+		return_string[6] = '\0';
+	}
+	else if(error_code > -200000 && error_code < 200000)
+	{
+		// no, format it as an integer
+		snprintf(return_string, 16, "%d", error_code);
+	}
+	else
+	{
+		// no, format it as an integer but in hex
+		snprintf(return_string, 16, "0x%x", error_code);
+	}
+	return return_string;
+}
+
+
+
+SInt64 CoreAudio_SizeCallback(void* inClientData)
+{
+	SDL_RWops* rw_ops = (SDL_RWops*)inClientData;
+	SInt64 current_position = SDL_RWtell(rw_ops);
+	SInt64 end_position = SDL_RWseek(rw_ops, 0, SEEK_END);
+	SDL_RWseek(rw_ops, current_position, SEEK_SET);
+//	fprintf(stderr, "CoreAudio_SizeCallback:%d\n", end_position);
+
+	return end_position;
+}
+
+OSStatus CoreAudio_ReadCallback(
+	void* inClientData,
+	SInt64 inPosition,
+	UInt32 requestCount,
+	void* data_buffer,
+	UInt32* actualCount
+)
+{
+	SDL_RWops* rw_ops = (SDL_RWops*)inClientData;
+	SDL_RWseek(rw_ops, inPosition, SEEK_SET);
+	size_t bytes_actually_read = SDL_RWread(rw_ops, data_buffer, 1, requestCount);
+	// Not sure how to test for a read error with SDL_RWops
+//	fprintf(stderr, "CoreAudio_ReadCallback:%d, %d\n", requestCount, bytes_actually_read);
+
+	*actualCount = bytes_actually_read;
+	return noErr;
+}
+
+
+static int CoreAudio_open(Sound_Sample *sample, const char *ext)
+{
+	CoreAudioFileContainer* core_audio_file_container;
+	AudioFileID* audio_file_id;
+	OSStatus error_result;
+	Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+	AudioStreamBasicDescription	actual_format;
+	AudioStreamBasicDescription	output_format;
+	Float64 estimated_duration;
+	UInt32 format_size;
+	
+
+	core_audio_file_container = (CoreAudioFileContainer*)SDL_malloc(sizeof(CoreAudioFileContainer));
+	BAIL_IF_MACRO(core_audio_file_container == NULL, ERR_OUT_OF_MEMORY, 0);
+
+
+	audio_file_id = (AudioFileID*)SDL_malloc(sizeof(AudioFileID));
+	BAIL_IF_MACRO(audio_file_id == NULL, ERR_OUT_OF_MEMORY, 0);
+
+	error_result = AudioFileOpenWithCallbacks(
+		internal->rw,
+		CoreAudio_ReadCallback,
+		NULL,
+		CoreAudio_SizeCallback,
+		NULL,
+		CoreAudio_GetAudioTypeForExtension(ext),
+		audio_file_id
+	);
+	if (error_result != noErr)
+	{
+		AudioFileClose(*audio_file_id);
+		SDL_free(audio_file_id);
+		SDL_free(core_audio_file_container);
+		SNDDBG(("Core Audio: can't grok data. reason: [%s].\n", CoreAudio_FourCCToString(error_result)));
+		BAIL_MACRO("Core Audio: Not valid audio data.", 0);
+	} /* if */
+	
+    format_size = sizeof(actual_format);
+    error_result = AudioFileGetProperty(
+		*audio_file_id,
+		kAudioFilePropertyDataFormat,
+		&format_size,
+		&actual_format
+	);
+    if (error_result != noErr)
+	{
+		AudioFileClose(*audio_file_id);
+		SDL_free(audio_file_id);
+		SDL_free(core_audio_file_container);
+		SNDDBG(("Core Audio: AudioFileGetProperty failed. reason: [%s]", CoreAudio_FourCCToString(error_result)));
+		BAIL_MACRO("Core Audio: Not valid audio data.", 0);
+	} /* if */
+
+    format_size = sizeof(estimated_duration);
+    error_result = AudioFileGetProperty(
+		*audio_file_id,
+		kAudioFilePropertyEstimatedDuration,
+		&format_size,
+		&estimated_duration
+	);
+    if (error_result != noErr)
+	{
+		AudioFileClose(*audio_file_id);
+		SDL_free(audio_file_id);
+		SDL_free(core_audio_file_container);
+		SNDDBG(("Core Audio: AudioFileGetProperty failed. reason: [%s].\n", CoreAudio_FourCCToString(error_result)));
+		BAIL_MACRO("Core Audio: Not valid audio data.", 0);
+	} /* if */
+
+
+	core_audio_file_container->audioFileID = audio_file_id;
+	
+	internal->decoder_private = core_audio_file_container;
+	
+	sample->flags = SOUND_SAMPLEFLAG_CANSEEK;
+	sample->actual.rate = (UInt32) actual_format.mSampleRate;
+	sample->actual.channels = (UInt8)actual_format.mChannelsPerFrame;
+	internal->total_time = (SInt32)(estimated_duration * 1000.0 + 0.5);
+
+#if 0
+	/* FIXME: Both Core Audio and SDL 1.3 support float and 32-bit formats */
+	if(actual_format.mFormatFlags & kAudioFormatFlagIsBigEndian)
+	{
+		if(16 == actual_format.mBitsPerChannel)
+		{
+			if(kAudioFormatFlagIsSignedInteger & actual_format.mFormatFlags)
+			{
+				sample->actual.format = AUDIO_S16MSB;
+			}
+			else
+			{
+				sample->actual.format = AUDIO_U16MSB;				
+			}
+		}
+		else if(8 == actual_format.mBitsPerChannel)
+		{
+			if(kAudioFormatFlagIsSignedInteger & actual_format.mFormatFlags)
+			{
+				sample->actual.format = AUDIO_S8;
+			}
+			else
+			{
+				sample->actual.format = AUDIO_U8;				
+			}
+		}
+		else // might be 0 for undefined? 
+		{
+			// This case seems to come up a lot for me. Maybe for file types like .m4a?
+			sample->actual.format = AUDIO_S16SYS;
+			SNDDBG(("Core Audio: Unsupported actual_format.mBitsPerChannel: [%d].\n", actual_format.mBitsPerChannel));
+			
+		}
+	}
+	else // little endian
+	{
+		if(16 == actual_format.mBitsPerChannel)
+		{
+			if(kAudioFormatFlagIsSignedInteger & actual_format.mFormatFlags)
+			{
+				sample->actual.format = AUDIO_S16LSB;
+			}
+			else
+			{
+				sample->actual.format = AUDIO_U16LSB;				
+			}
+		}
+		else if(8 == actual_format.mBitsPerChannel)
+		{
+			if(kAudioFormatFlagIsSignedInteger & actual_format.mFormatFlags)
+			{
+				sample->actual.format = AUDIO_S8;
+			}
+			else
+			{
+				sample->actual.format = AUDIO_U8;				
+			}
+		}
+		else // might be 0 for undefined? 
+		{
+			sample->actual.format = AUDIO_S16SYS;
+			
+			SNDDBG(("Core Audio: Unsupported actual_format.mBitsPerChannel: [%d].\n", actual_format.mBitsPerChannel));
+		}
+
+	}
+#else
+	
+	
+	
+    /*
+     * I want to use Core Audio to do conversion and decoding for performance reasons.
+	 * This is particularly important on mobile devices like iOS.
+	 * Taking from the Ogg Vorbis decode, I pretend the "actual" format is the same 
+	 * as the desired format. 
+     */
+    sample->actual.format = (sample->desired.format == 0) ?
+	AUDIO_S16SYS : sample->desired.format;
+#endif	
+
+
+	SNDDBG(("CoreAudio: channels == (%d).\n", sample->actual.channels));
+	SNDDBG(("CoreAudio: sampling rate == (%d).\n",sample->actual.rate));
+	SNDDBG(("CoreAudio: total seconds of sample == (%d).\n", internal->total_time));
+	SNDDBG(("CoreAudio: sample->actual.format == (%d).\n", sample->actual.format));
+
+
+	
+	error_result = ExtAudioFileWrapAudioFileID(*audio_file_id,
+		false, // set to false for read-only
+		&core_audio_file_container->extAudioFileRef
+	);
+	if(error_result != noErr)
+	{
+		AudioFileClose(*audio_file_id);
+		SDL_free(audio_file_id);
+		SDL_free(core_audio_file_container);
+		SNDDBG(("Core Audio: can't wrap data. reason: [%s].\n", CoreAudio_FourCCToString(error_result)));
+		BAIL_MACRO("Core Audio: Failed to wrap data.", 0);
+	} /* if */
+
+
+	/* The output format must be linear PCM because that's the only type OpenAL knows how to deal with.
+	 * Set the client format to 16 bit signed integer (native-endian) data because that is the most
+	 * optimal format on iPhone/iPod Touch hardware.
+	 * Maintain the channel count and sample rate of the original source format.
+	 */
+	output_format.mSampleRate = actual_format.mSampleRate; // preserve the original sample rate
+	output_format.mChannelsPerFrame = actual_format.mChannelsPerFrame; // preserve the number of channels
+	output_format.mFormatID = kAudioFormatLinearPCM; // We want linear PCM data
+	output_format.mFramesPerPacket = 1; // We know for linear PCM, the definition is 1 frame per packet
+
+	if(sample->desired.format == 0)
+	{
+		// do AUDIO_S16SYS
+		output_format.mFormatFlags = kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked; // I seem to read failures problems without kAudioFormatFlagIsPacked. From a mailing list post, this seems to be a Core Audio bug.
+		output_format.mBitsPerChannel = 16; // We know we want 16-bit
+	}
+	else
+	{
+		output_format.mFormatFlags = 0; // clear flags
+		output_format.mFormatFlags |= kAudioFormatFlagIsPacked; // I seem to read failures problems without kAudioFormatFlagIsPacked. From a mailing list post, this seems to be a Core Audio bug.
+		// Mask against bitsize
+		if(0xFF & sample->desired.format)
+		{
+			output_format.mBitsPerChannel = 16; /* 16-bit */
+		}
+		else
+		{
+			output_format.mBitsPerChannel = 8; /* 8-bit */
+		}
+
+		// Mask for signed/unsigned
+		if((1<<15) & sample->desired.format)
+		{
+			output_format.mFormatFlags = output_format.mFormatFlags | kAudioFormatFlagIsSignedInteger;
+
+		}
+		else
+		{
+			// no flag set for unsigned
+		}
+		// Mask for big/little endian
+		if((1<<12) & sample->desired.format)
+		{
+			output_format.mFormatFlags = output_format.mFormatFlags | kAudioFormatFlagIsBigEndian;
+		}
+		else
+		{
+			// no flag set for little endian 
+		}
+	}
+
+	output_format.mBytesPerPacket = output_format.mBitsPerChannel/8 * output_format.mChannelsPerFrame; // e.g. 16-bits/8 * channels => so 2-bytes per channel per frame
+	output_format.mBytesPerFrame = output_format.mBitsPerChannel/8 * output_format.mChannelsPerFrame; // For PCM, since 1 frame is 1 packet, it is the same as mBytesPerPacket
+
+	
+/*
+	output_format.mSampleRate = actual_format.mSampleRate; // preserve the original sample rate
+	output_format.mChannelsPerFrame = actual_format.mChannelsPerFrame; // preserve the number of channels
+	output_format.mFormatID = kAudioFormatLinearPCM; // We want linear PCM data
+//	output_format.mFormatFlags = kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger;
+	output_format.mFormatFlags = kAudioFormatFlagsNativeEndian |  kAudioFormatFlagIsSignedInteger;
+	output_format.mFramesPerPacket = 1; // We know for linear PCM, the definition is 1 frame per packet
+	output_format.mBitsPerChannel = 16; // We know we want 16-bit
+	output_format.mBytesPerPacket = 2 * output_format.mChannelsPerFrame; // We know we are using 16-bit, so 2-bytes per channel per frame
+	output_format.mBytesPerFrame = 2 * output_format.mChannelsPerFrame; // For PCM, since 1 frame is 1 packet, it is the same as mBytesPerPacket
+*/
+	SNDDBG(("output_format: mSampleRate: %lf\n", output_format.mSampleRate)); 
+	SNDDBG(("output_format: mChannelsPerFrame: %d\n", output_format.mChannelsPerFrame)); 
+	SNDDBG(("output_format: mFormatID: %d\n", output_format.mFormatID)); 
+	SNDDBG(("output_format: mFormatFlags: %d\n", output_format.mFormatFlags)); 
+	SNDDBG(("output_format: mFramesPerPacket: %d\n", output_format.mFramesPerPacket)); 
+	SNDDBG(("output_format: mBitsPerChannel: %d\n", output_format.mBitsPerChannel)); 
+	SNDDBG(("output_format: mBytesPerPacket: %d\n", output_format.mBytesPerPacket)); 
+	SNDDBG(("output_format: mBytesPerFrame: %d\n", output_format.mBytesPerFrame)); 
+	
+	
+	/* Set the desired client (output) data format */
+	error_result = ExtAudioFileSetProperty(core_audio_file_container->extAudioFileRef, kExtAudioFileProperty_ClientDataFormat, sizeof(output_format), &output_format);
+	if(noErr != error_result)
+	{
+		ExtAudioFileDispose(core_audio_file_container->extAudioFileRef);
+		AudioFileClose(*audio_file_id);
+		SDL_free(audio_file_id);
+		SDL_free(core_audio_file_container);
+		SNDDBG(("Core Audio: ExtAudioFileSetProperty(kExtAudioFileProperty_ClientDataFormat) failed, reason: [%s].\n", CoreAudio_FourCCToString(error_result)));
+		BAIL_MACRO("Core Audio: Not valid audio data.", 0);
+	}	
+
+
+	core_audio_file_container->outputFormat = (AudioStreamBasicDescription*)SDL_malloc(sizeof(AudioStreamBasicDescription));
+	BAIL_IF_MACRO(core_audio_file_container->outputFormat == NULL, ERR_OUT_OF_MEMORY, 0);
+
+
+	
+	/* Copy the output format to the audio_description that was passed in so the 
+	 * info will be returned to the user.
+	 */
+	SDL_memcpy(core_audio_file_container->outputFormat, &output_format, sizeof(AudioStreamBasicDescription));
+
+	
+
+	return 1;
+} /* CoreAudio_open */
+
+
+static void CoreAudio_close(Sound_Sample *sample)
+{
+	Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+	CoreAudioFileContainer* core_audio_file_container = (CoreAudioFileContainer *) internal->decoder_private;
+
+	SDL_free(core_audio_file_container->outputFormat);
+	ExtAudioFileDispose(core_audio_file_container->extAudioFileRef);
+	AudioFileClose(*core_audio_file_container->audioFileID);
+	SDL_free(core_audio_file_container->audioFileID);
+	SDL_free(core_audio_file_container);
+} /* CoreAudio_close */
+
+
+static Uint32 CoreAudio_read(Sound_Sample *sample)
+{
+	OSStatus error_result = noErr;	
+	/* Documentation/example shows SInt64, but is problematic for big endian
+	 * on 32-bit cast for ExtAudioFileRead() because it takes the upper
+	 * bits which turn to 0.
+	 */
+	UInt32 buffer_size_in_frames = 0;
+	UInt32 buffer_size_in_frames_remaining = 0;
+	UInt32 total_frames_read = 0;
+	UInt32 data_buffer_size = 0;
+	UInt32 bytes_remaining = 0;
+	size_t total_bytes_read = 0;
+	Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+	CoreAudioFileContainer* core_audio_file_container = (CoreAudioFileContainer *) internal->decoder_private;
+	UInt32 max_buffer_size = internal->buffer_size;
+
+//	printf("internal->buffer_size=%d, internal->buffer=0x%x, sample->buffer_size=%d\n", internal->buffer_size, internal->buffer, sample->buffer_size); 
+//	printf("internal->max_buffer_size=%d\n", max_buffer_size); 
+
+	/* Compute how many frames will fit into our max buffer size */
+	/* Warning: If this is not evenly divisible, the buffer will not be completely filled which violates the SDL_sound assumption. */
+	buffer_size_in_frames = max_buffer_size / core_audio_file_container->outputFormat->mBytesPerFrame;
+//	printf("buffer_size_in_frames=%ld, internal->buffer_size=%d, internal->buffer=0x%x outputFormat->mBytesPerFrame=%d, sample->buffer_size=%d\n", buffer_size_in_frames, internal->buffer_size, internal->buffer, core_audio_file_container->outputFormat->mBytesPerFrame, sample->buffer_size); 
+
+
+//	void* temp_buffer = SDL_malloc(max_buffer_size);
+	
+	AudioBufferList audio_buffer_list;
+	audio_buffer_list.mNumberBuffers = 1;
+	audio_buffer_list.mBuffers[0].mDataByteSize = max_buffer_size;
+	audio_buffer_list.mBuffers[0].mNumberChannels = core_audio_file_container->outputFormat->mChannelsPerFrame;
+	audio_buffer_list.mBuffers[0].mData = internal->buffer;
+
+
+	bytes_remaining = max_buffer_size;
+	buffer_size_in_frames_remaining = buffer_size_in_frames;
+	
+	// oops. Due to the kAudioFormatFlagIsPacked bug, 
+	// I was misled to believe that Core Audio
+	// was not always filling my entire requested buffer. 
+	// So this while-loop might be unnecessary.
+	// However, I have not exhaustively tested all formats, 
+	// so maybe it is possible this loop is useful.
+	// It might also handle the not-evenly disvisible case above.
+	while(buffer_size_in_frames_remaining > 0 && !(sample->flags & SOUND_SAMPLEFLAG_EOF))
+	{
+		
+		data_buffer_size = (UInt32)(buffer_size_in_frames * core_audio_file_container->outputFormat->mBytesPerFrame);
+//		printf("data_buffer_size=%d\n", data_buffer_size); 
+
+		buffer_size_in_frames = buffer_size_in_frames_remaining;
+		
+//		printf("reading buffer_size_in_frames=%"PRId64"\n", buffer_size_in_frames); 
+
+
+		audio_buffer_list.mBuffers[0].mDataByteSize = bytes_remaining;
+		audio_buffer_list.mBuffers[0].mData = &(((UInt8*)internal->buffer)[total_bytes_read]);
+
+		
+		/* Read the data into an AudioBufferList */
+		error_result = ExtAudioFileRead(core_audio_file_container->extAudioFileRef, &buffer_size_in_frames, &audio_buffer_list);
+		if(error_result == noErr)
+		{
+		
+		
+			/* Success */
+			
+			total_frames_read += buffer_size_in_frames;
+			buffer_size_in_frames_remaining = buffer_size_in_frames_remaining - buffer_size_in_frames;
+			
+//			printf("read buffer_size_in_frames=%"PRId64", buffer_size_in_frames_remaining=%"PRId64"\n", buffer_size_in_frames, buffer_size_in_frames_remaining); 
+
+			/* ExtAudioFileRead returns the number of frames actually read. Need to convert back to bytes. */
+			data_buffer_size = (UInt32)(buffer_size_in_frames * core_audio_file_container->outputFormat->mBytesPerFrame);
+//			printf("data_buffer_size=%d\n", data_buffer_size); 
+
+			total_bytes_read += data_buffer_size;
+			bytes_remaining = bytes_remaining - data_buffer_size;
+
+			/* Note: 0 == buffer_size_in_frames is a legitimate value meaning we are EOF. */
+			if(0 == buffer_size_in_frames)
+			{
+				sample->flags |= SOUND_SAMPLEFLAG_EOF;			
+			}
+
+		}
+		else 
+		{
+			SNDDBG(("Core Audio: ExtAudioFileReadfailed, reason: [%s].\n", CoreAudio_FourCCToString(error_result)));
+
+			sample->flags |= SOUND_SAMPLEFLAG_ERROR;
+			break;
+			
+		}
+	}
+	
+	if( (!(sample->flags & SOUND_SAMPLEFLAG_EOF)) && (total_bytes_read < max_buffer_size))
+	{
+		SNDDBG(("Core Audio: ExtAudioFileReadfailed SOUND_SAMPLEFLAG_EAGAIN, reason: [total_bytes_read < max_buffer_size], %d, %d.\n", total_bytes_read , max_buffer_size));
+		
+		/* Don't know what to do here. */
+		sample->flags |= SOUND_SAMPLEFLAG_EAGAIN;
+	}
+	return total_bytes_read;
+} /* CoreAudio_read */
+
+
+static int CoreAudio_rewind(Sound_Sample *sample)
+{
+	OSStatus error_result = noErr;	
+	Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+	CoreAudioFileContainer* core_audio_file_container = (CoreAudioFileContainer *) internal->decoder_private;
+	
+	error_result = ExtAudioFileSeek(core_audio_file_container->extAudioFileRef, 0);
+	if(error_result != noErr)
+	{
+		sample->flags |= SOUND_SAMPLEFLAG_ERROR;
+	}
+	return 1;
+} /* CoreAudio_rewind */
+
+/* Note: I found this tech note:
+ https://developer.apple.com/library/mac/#qa/qa2009/qa1678.html
+ I don't know if this applies to us. So far, I haven't noticed the problem,
+ so I haven't applied any of the techniques.
+ */
+static int CoreAudio_seek(Sound_Sample *sample, Uint32 ms)
+{
+	OSStatus error_result = noErr;	
+	Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+	CoreAudioFileContainer* core_audio_file_container = (CoreAudioFileContainer *) internal->decoder_private;
+	SInt64 frame_offset = 0;
+	AudioStreamBasicDescription	actual_format;
+	UInt32 format_size;
+
+	
+	/* I'm confused. The Apple documentation says this:
+	"Seek position is specified in the sample rate and frame count of the file’s audio data format
+	— not your application’s audio data format."
+	My interpretation is that I want to get the "actual format of the file and compute the frame offset.
+	But when I try that, seeking goes to the wrong place.
+	When I use outputFormat, things seem to work correctly.
+	I must be misinterpreting the documentation or doing something wrong.
+	*/
+#if 0 /* not working */
+    format_size = sizeof(AudioStreamBasicDescription);
+    error_result = AudioFileGetProperty(
+		*core_audio_file_container->audioFileID,
+		kAudioFilePropertyDataFormat,
+		&format_size,
+		&actual_format
+	);
+    if(error_result != noErr)
+	{
+		sample->flags |= SOUND_SAMPLEFLAG_ERROR;
+		BAIL_MACRO("Core Audio: Could not GetProperty for kAudioFilePropertyDataFormat.", 0);
+	} /* if */
+
+	// packetIndex = (pos * sampleRate) / framesPerPacket
+	//	frame_offset = (SInt64)((ms/1000.0 * actual_format.mSampleRate) / actual_format.mFramesPerPacket);
+#else /* seems to work, but I'm confused */
+	// packetIndex = (pos * sampleRate) / framesPerPacket
+	frame_offset = (SInt64)((ms/1000.0 * core_audio_file_container->outputFormat->mSampleRate) / core_audio_file_container->outputFormat->mFramesPerPacket);	
+#endif
+
+	// computed against actual format and not the client format
+	error_result = ExtAudioFileSeek(core_audio_file_container->extAudioFileRef, frame_offset);
+	if(error_result != noErr)
+	{
+		sample->flags |= SOUND_SAMPLEFLAG_ERROR;
+	}
+	
+	return 1;
+} /* CoreAudio_seek */
+
+
+static const char *extensions_coreaudio[] =
+{
+	"aif",
+	"aiff",
+	"aifc",
+	"wav",
+	"wave",
+	"mp3",
+	"mp4",
+	"m4a",
+	"aac",
+	"caf",
+	"Sd2f",
+	"Sd2",
+	"au",
+	"next",
+	"mp2",
+	"mp1",
+	"ac3",
+	"3gpp",
+	"3gp2",
+	"amrf",
+	"amr",
+	"ima4",
+	"ima",
+	NULL 
+};
+const Sound_DecoderFunctions __Sound_DecoderFunctions_CoreAudio =
+{
+    {
+        extensions_coreaudio,
+        "Decode audio through Core Audio through",
+        "Eric Wing <ewing . public @ playcontrol.net>",
+        "https://playcontrol.net"
+    },
+	
+    CoreAudio_init,       /*   init() method */
+    CoreAudio_quit,       /*   quit() method */
+    CoreAudio_open,       /*   open() method */
+    CoreAudio_close,      /*  close() method */
+    CoreAudio_read,       /*   read() method */
+    CoreAudio_rewind,     /* rewind() method */
+    CoreAudio_seek        /*   seek() method */
+};
+
+#endif /* SOUND_SUPPORTS_COREAUDIO */
+
+/* end of SDL_sound_coreaudio.c ... */
+