/*

SDL_mixer: An audio mixer library based on the SDL library

Copyright (C) 1997-2019 Sam Lantinga <slouken@libsdl.org>

This software is provided 'as-is', without any express or implied

warranty. In no event will the authors be held liable for any damages

arising from the use of this software.

Permission is granted to anyone to use this software for any purpose,

including commercial applications, and to alter it and redistribute it

freely, subject to the following restrictions:

1. The origin of this software must not be misrepresented; you must not

claim that you wrote the original software. If you use this software

in a product, an acknowledgment in the product documentation would be

appreciated but is not required.

2. Altered source versions must be plainly marked as such, and must not be

misrepresented as being the original software.

3. This notice may not be removed or altered from any source distribution.

*/

#ifdef MUSIC_WAV

/* This file supports streaming WAV files */

#include "music_wav.h"

typedef struct {

SDL_bool active;

Uint32 start;

Uint32 stop;

Uint32 initial_play_count;

Uint32 current_play_count;

} WAVLoopPoint;

typedef struct {

SDL_RWops *src;

int freesrc;

SDL_AudioSpec spec;

int volume;

int play_count;

Sint64 start;

Sint64 stop;

Sint64 samplesize;

Uint8 *buffer;

SDL_AudioStream *stream;

unsigned int numloops;

WAVLoopPoint *loops;

Uint16 encoding;

int (*decode)(void *music, int length);

} WAV_Music;

/*

Taken with permission from SDL_wave.h, part of the SDL library,

available at: http://www.libsdl.org/

and placed under the same license as this mixer library.

*/

/* WAVE files are little-endian */

/*******************************************/

/* Define values for Microsoft WAVE format */

/*******************************************/

#define RIFF 0x46464952 /* "RIFF" */

#define WAVE 0x45564157 /* "WAVE" */

#define FMT 0x20746D66 /* "fmt " */

#define DATA 0x61746164 /* "data" */

#define SMPL 0x6c706d73 /* "smpl" */

#define LIST 0x5453494c /* "LIST" */

#define ID3_ 0x20336469 /* "id3 " */

#define PCM_CODE 1 /* WAVE_FORMAT_PCM */

#define ADPCM_CODE 2 /* WAVE_FORMAT_ADPCM */

#define FLOAT_CODE 3 /* WAVE_FORMAT_IEEE_FLOAT */

#define ALAW_CODE 6 /* WAVE_FORMAT_ALAW */

#define uLAW_CODE 7 /* WAVE_FORMAT_MULAW */

#define EXT_CODE 0xFFFE /* WAVE_FORMAT_EXTENSIBLE */

#define WAVE_MONO 1

#define WAVE_STEREO 2

typedef struct {

/* Not saved in the chunk we read:

Uint32 chunkID;

Uint32 chunkLen;

*/

Uint16 encoding;

Uint16 channels; /* 1 = mono, 2 = stereo */

Uint32 frequency; /* One of 11025, 22050, or 44100 Hz */

Uint32 byterate; /* Average bytes per second */

Uint16 blockalign; /* Bytes per sample block */

Uint16 bitspersample; /* One of 8, 12, 16, or 4 for ADPCM */

} WaveFMT;

typedef struct {

Uint16 cbSize;

union {

Uint16 validbitspersample; /* bits of precision */

Uint16 samplesperblock; /* valid if wBitsPerSample==0 */

Uint16 reserved; /* If neither applies, set to zero. */

} Samples;

Uint32 channelsmask;

/* GUID subFormat 16 bytes */

Uint32 subencoding;

Uint16 sub_data2;

Uint16 sub_data3;

Uint8 sub_data[8];

} WaveFMTex;

typedef struct {

Uint32 identifier;

Uint32 type;

Uint32 start;

Uint32 end;

Uint32 fraction;

Uint32 play_count;

} SampleLoop;

typedef struct {

/* Not saved in the chunk we read:

Uint32 chunkID;

Uint32 chunkLen;

*/

Uint32 manufacturer;

Uint32 product;

Uint32 sample_period;

Uint32 MIDI_unity_note;

Uint32 MIDI_pitch_fraction;

Uint32 SMTPE_format;

Uint32 SMTPE_offset;

Uint32 sample_loops;

Uint32 sampler_data;

SampleLoop loops[1];

} SamplerChunk;

/*********************************************/

/* Define values for AIFF (IFF audio) format */

/*********************************************/

#define FORM 0x4d524f46 /* "FORM" */

#define AIFF 0x46464941 /* "AIFF" */

#define AIFC 0x43464941 /* "AIFС" */

#define FVER 0x52455646 /* "FVER" */

#define SSND 0x444e5353 /* "SSND" */

#define COMM 0x4d4d4f43 /* "COMM" */

#define AIFF_ID3_ 0x20334449 /* "ID3 " */

#define MARK 0x4B52414D /* "MARK" */

#define INST 0x54534E49 /* "INST" */

#define AUTH 0x48545541 /* "AUTH" */

#define NAME 0x454D414E /* "NAME" */

#define _c__ 0x20296328 /* "(c) " */

/* Supported compression types */

#define NONE 0x454E4F4E /* "NONE" */

#define sowt 0x74776F73 /* "sowt" */

#define raw_ 0x20776172 /* "raw " */

#define ulaw 0x77616C75 /* "ulaw" */

#define alaw 0x77616C61 /* "alaw" */

#define ULAW 0x57414C55 /* "ULAW" */

#define ALAW 0x57414C41 /* "ALAW" */

#define fl32 0x32336C66 /* "fl32" */

#define fl64 0x34366C66 /* "fl64" */

#define FL32 0x32334C46 /* "FL32" */

/* Function to load the WAV/AIFF stream */

static SDL_bool LoadWAVMusic(WAV_Music *wave);

static SDL_bool LoadAIFFMusic(WAV_Music *wave);

static void WAV_Delete(void *context);

static int fetch_pcm(void *context, int length);

/* Load a WAV stream from the given RWops object */

static void *WAV_CreateFromRW(SDL_RWops *src, int freesrc)

{

WAV_Music *music;

Uint32 magic;

SDL_bool loaded = SDL_FALSE;

music = (WAV_Music *)SDL_calloc(1, sizeof(*music));

if (!music) {

SDL_OutOfMemory();

return NULL;

}

music->src = src;

music->volume = MIX_MAX_VOLUME;

/* Default decoder is PCM */

music->decode = fetch_pcm;

music->encoding = PCM_CODE;

magic = SDL_ReadLE32(src);

if (magic == RIFF || magic == WAVE) {

loaded = LoadWAVMusic(music);

} else if (magic == FORM) {

loaded = LoadAIFFMusic(music);

} else {

Mix_SetError("Unknown WAVE format");

}

if (!loaded) {

SDL_free(music);

return NULL;

}

music->buffer = (Uint8*)SDL_malloc(music->spec.size);

if (!music->buffer) {

WAV_Delete(music);

return NULL;

}

music->stream = SDL_NewAudioStream(

music->spec.format, music->spec.channels, music->spec.freq,

music_spec.format, music_spec.channels, music_spec.freq);

if (!music->stream) {

WAV_Delete(music);

return NULL;

}

music->freesrc = freesrc;

return music;

}

static void WAV_SetVolume(void *context, int volume)

{

WAV_Music *music = (WAV_Music *)context;

music->volume = volume;

}

static int WAV_GetVolume(void *context)

{

WAV_Music *music = (WAV_Music *)context;

return music->volume;

}

/* Start playback of a given WAV stream */

static int WAV_Play(void *context, int play_count)

{

WAV_Music *music = (WAV_Music *)context;

unsigned int i;

for (i = 0; i < music->numloops; ++i) {

WAVLoopPoint *loop = &music->loops[i];

loop->active = SDL_TRUE;

loop->current_play_count = loop->initial_play_count;

}

music->play_count = play_count;

if (SDL_RWseek(music->src, music->start, RW_SEEK_SET) < 0) {

return -1;

}

return 0;

}

static int fetch_pcm(void *context, int length)

{

WAV_Music *music = (WAV_Music *)context;

return (int)SDL_RWread(music->src, music->buffer, 1, (size_t)length);

}

static Uint32 PCM_S24_to_S32_BE(Uint8 *x) {

const Uint32 bits = 24;

Uint32 in = (((Uint32)x[0] << 0) & 0x0000FF) |

(((Uint32)x[1] << 8) & 0x00FF00) |

(((Uint32)x[2] << 16) & 0xFF0000);

Uint32 m = 1u << (bits - 1);

return (in ^ m) - m;

}

static Uint32 PCM_S24_to_S32_LE(Uint8 *x) {

const Uint32 bits = 24;

Uint32 in = (((Uint32)x[2] << 0) & 0x0000FF) |

(((Uint32)x[1] << 8) & 0x00FF00) |

(((Uint32)x[0] << 16) & 0xFF0000);

Uint32 m = 1u << (bits - 1);

return (in ^ m) - m;

}

static int fetch_pcm24be(void *context, int length)

{

WAV_Music *music = (WAV_Music *)context;

int i = 0, o = 0;

length = (int)SDL_RWread(music->src, music->buffer, 1, (size_t)((length / 4) * 3));

if (length % music->samplesize != 0) {

length -= length % music->samplesize;

}

for (i = length - 3, o = ((length - 3) / 3) * 4; i >= 0; i -= 3, o -= 4) {

Uint32 decoded = PCM_S24_to_S32_BE(music->buffer + i);

music->buffer[o + 0] = (decoded >> 0) & 0xFF;

music->buffer[o + 1] = (decoded >> 8) & 0xFF;

music->buffer[o + 2] = (decoded >> 16) & 0xFF;

music->buffer[o + 3] = (decoded >> 24) & 0xFF;

}

return (length / 3) * 4;

}

static int fetch_pcm24le(void *context, int length)

{

WAV_Music *music = (WAV_Music *)context;

int i = 0, o = 0;

length = (int)SDL_RWread(music->src, music->buffer, 1, (size_t)((length / 4) * 3));

if (length % music->samplesize != 0) {

length -= length % music->samplesize;

}

for (i = length - 3, o = ((length - 3) / 3) * 4; i >= 0; i -= 3, o -= 4) {

Uint32 decoded = PCM_S24_to_S32_LE(music->buffer + i);

music->buffer[o + 3] = (decoded >> 0) & 0xFF;

music->buffer[o + 2] = (decoded >> 8) & 0xFF;

music->buffer[o + 1] = (decoded >> 16) & 0xFF;

music->buffer[o + 0] = (decoded >> 24) & 0xFF;

}

return (length / 3) * 4;

}

SDL_FORCE_INLINE double

Mix_SwapDouble(double x)

{

union

{

double f;

Uint64 ui64;

} swapper;

swapper.f = x;

swapper.ui64 = SDL_Swap64(swapper.ui64);

return swapper.f;

}

#if SDL_BYTEORDER == SDL_LIL_ENDIAN

#define Mix_SwapDoubleLE(X) (X)

#define Mix_SwapDoubleBE(X) Mix_SwapDouble(X)

#else

#define Mix_SwapDoubleLE(X) Mix_SwapDouble(X)

#define Mix_SwapDoubleBE(X) (X)

#endif

static int fetch_float64be(void *context, int length)

{

WAV_Music *music = (WAV_Music *)context;

int i = 0, o = 0;

length = (int)SDL_RWread(music->src, music->buffer, 1, (size_t)(length));

if (length % music->samplesize != 0) {

length -= length % music->samplesize;

}

for (i = 0, o = 0; i <= length; i += 8, o += 4) {

union

{

float f;

Uint32 ui32;

} sample;

sample.f = (float)Mix_SwapDoubleBE(*(double*)(music->buffer + i));

music->buffer[o + 0] = (sample.ui32 >> 0) & 0xFF;

music->buffer[o + 1] = (sample.ui32 >> 8) & 0xFF;

music->buffer[o + 2] = (sample.ui32 >> 16) & 0xFF;

music->buffer[o + 3] = (sample.ui32 >> 24) & 0xFF;

}

return length / 2;

}

static int fetch_float64le(void *context, int length)

{

WAV_Music *music = (WAV_Music *)context;

int i = 0, o = 0;

length = (int)SDL_RWread(music->src, music->buffer, 1, (size_t)(length));

if (length % music->samplesize != 0) {

length -= length % music->samplesize;

}

for (i = 0, o = 0; i <= length; i += 8, o += 4) {

union

{

float f;

Uint32 ui32;

} sample;

sample.f = (float)Mix_SwapDoubleLE(*(double*)(music->buffer + i));

music->buffer[o + 0] = (sample.ui32 >> 0) & 0xFF;

music->buffer[o + 1] = (sample.ui32 >> 8) & 0xFF;

music->buffer[o + 2] = (sample.ui32 >> 16) & 0xFF;

music->buffer[o + 3] = (sample.ui32 >> 24) & 0xFF;

}

return length / 2;

}

/*

G711 decode tables taken from SDL2 (src/audio/SDL_wave.c)

*/

#ifdef SDL_WAVE_LAW_LUT

static const Sint16 alaw_lut[256] = {

-5504, -5248, -6016, -5760, -4480, -4224, -4992, -4736, -7552, -7296, -8064, -7808, -6528, -6272, -7040, -6784, -2752,

-2624, -3008, -2880, -2240, -2112, -2496, -2368, -3776, -3648, -4032, -3904, -3264, -3136, -3520, -3392, -22016,

-20992, -24064, -23040, -17920, -16896, -19968, -18944, -30208, -29184, -32256, -31232, -26112, -25088, -28160, -27136, -11008,

-10496, -12032, -11520, -8960, -8448, -9984, -9472, -15104, -14592, -16128, -15616, -13056, -12544, -14080, -13568, -344,

-328, -376, -360, -280, -264, -312, -296, -472, -456, -504, -488, -408, -392, -440, -424, -88,

-72, -120, -104, -24, -8, -56, -40, -216, -200, -248, -232, -152, -136, -184, -168, -1376,

-1312, -1504, -1440, -1120, -1056, -1248, -1184, -1888, -1824, -2016, -1952, -1632, -1568, -1760, -1696, -688,

-656, -752, -720, -560, -528, -624, -592, -944, -912, -1008, -976, -816, -784, -880, -848, 5504,

5248, 6016, 5760, 4480, 4224, 4992, 4736, 7552, 7296, 8064, 7808, 6528, 6272, 7040, 6784, 2752,

2624, 3008, 2880, 2240, 2112, 2496, 2368, 3776, 3648, 4032, 3904, 3264, 3136, 3520, 3392, 22016,

20992, 24064, 23040, 17920, 16896, 19968, 18944, 30208, 29184, 32256, 31232, 26112, 25088, 28160, 27136, 11008,

10496, 12032, 11520, 8960, 8448, 9984, 9472, 15104, 14592, 16128, 15616, 13056, 12544, 14080, 13568, 344,

328, 376, 360, 280, 264, 312, 296, 472, 456, 504, 488, 408, 392, 440, 424, 88,

72, 120, 104, 24, 8, 56, 40, 216, 200, 248, 232, 152, 136, 184, 168, 1376,

1312, 1504, 1440, 1120, 1056, 1248, 1184, 1888, 1824, 2016, 1952, 1632, 1568, 1760, 1696, 688,

656, 752, 720, 560, 528, 624, 592, 944, 912, 1008, 976, 816, 784, 880, 848

};

static const Sint16 mulaw_lut[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

};

#endif

static Sint16 uLAW_To_PCM16(Uint8 u_val)

{

#ifdef SDL_WAVE_LAW_LUT

return mulaw_lut[u_val];

#else

Uint8 nibble = ~u_val;

Sint16 mantissa = nibble & 0xf;

Uint8 exponent = (nibble >> 4) & 0x7;

Sint16 step = (Sint16)(4 << (exponent + 1));

mantissa = (Sint16)(0x80 << exponent) + step * mantissa + step / 2 - 132;

return nibble & 0x80 ? -mantissa : mantissa;

#endif

}

static Sint16 ALAW_To_PCM16(Uint8 a_val)

{

#ifdef SDL_WAVE_LAW_LUT

return alaw_lut[a_val];

#else

Uint8 nibble = a_val;

Uint8 exponent = (nibble & 0x7f) ^ 0x55;

Sint16 mantissa = exponent & 0xf;

exponent >>= 4;

if (exponent > 0) {

mantissa |= 0x10;

}

mantissa = (Sint16)(mantissa << 4) | 0x8;

if (exponent > 1) {

mantissa <<= exponent - 1;

}

return nibble & 0x80 ? mantissa : -mantissa;

#endif

}

static int fetch_xlaw(Sint16 (*decode_sample)(Uint8), void *context, int length)

{

WAV_Music *music = (WAV_Music *)context;

int i = 0, o = 0;

length = (int)SDL_RWread(music->src, music->buffer, 1, (size_t)(length / 2));

if (length % music->samplesize != 0) {

length -= length % music->samplesize;

}

for (i = length - 1, o = (length - 1) * 2; i >= 0; i--, o -= 2) {

Uint16 decoded = (Uint16)decode_sample(music->buffer[i]);

music->buffer[o] = decoded & 0xFF;

music->buffer[o + 1] = (decoded >> 8) & 0xFF;

}

return length * 2;

}

static int fetch_ulaw(void *context, int length)

{

return fetch_xlaw(uLAW_To_PCM16, context, length);

}

static int fetch_alaw(void *context, int length)

{

return fetch_xlaw(ALAW_To_PCM16, context, length);

}

/* Play some of a stream previously started with WAV_Play() */

static int WAV_GetSome(void *context, void *data, int bytes, SDL_bool *done)

{

WAV_Music *music = (WAV_Music *)context;

Sint64 pos, stop;

WAVLoopPoint *loop;

Sint64 loop_start = music->start;

Sint64 loop_stop = music->stop;

SDL_bool looped = SDL_FALSE;

SDL_bool at_end = SDL_FALSE;

unsigned int i;

int filled, amount, result;

filled = SDL_AudioStreamGet(music->stream, data, bytes);

if (filled != 0) {

return filled;

}

if (!music->play_count) {

/* All done */

*done = SDL_TRUE;

return 0;

}

pos = SDL_RWtell(music->src);

stop = music->stop;

loop = NULL;

for (i = 0; i < music->numloops; ++i) {

loop = &music->loops[i];

if (loop->active) {

const int bytes_per_sample = (SDL_AUDIO_BITSIZE(music->spec.format) / 8) * music->spec.channels;

loop_start = music->start + loop->start * (Uint32)bytes_per_sample;

loop_stop = music->start + (loop->stop + 1) * (Uint32)bytes_per_sample;

if (pos >= loop_start && pos < loop_stop) {

stop = loop_stop;

break;

}

}

loop = NULL;

}

amount = (int)music->spec.size;

if ((stop - pos) < amount) {

amount = (int)(stop - pos);

}

amount = music->decode(music, amount);

if (amount > 0) {

result = SDL_AudioStreamPut(music->stream, music->buffer, amount);

if (result < 0) {

return -1;

}

} else {

/* We might be looping, continue */

at_end = SDL_TRUE;

}

if (loop && SDL_RWtell(music->src) >= stop) {

if (loop->current_play_count == 1) {

loop->active = SDL_FALSE;

} else {

if (loop->current_play_count > 0) {

--loop->current_play_count;

}

SDL_RWseek(music->src, loop_start, RW_SEEK_SET);

looped = SDL_TRUE;

}

}

if (!looped && (at_end || SDL_RWtell(music->src) >= music->stop)) {

if (music->play_count == 1) {

music->play_count = 0;

SDL_AudioStreamFlush(music->stream);

} else {

int play_count = -1;

if (music->play_count > 0) {

play_count = (music->play_count - 1);

}

if (WAV_Play(music, play_count) < 0) {

return -1;

}

}

}

/* We'll get called again in the case where we looped or have more data */

return 0;

}

static int WAV_GetAudio(void *context, void *data, int bytes)

{

WAV_Music *music = (WAV_Music *)context;

return music_pcm_getaudio(context, data, bytes, music->volume, WAV_GetSome);

}

static int WAV_Seek(void *context, double position)

{

WAV_Music *music = (WAV_Music *)context;

Sint64 sample_size = music->spec.freq * music->samplesize;

Sint64 dest_offset = (Sint64)(position * (double)music->spec.freq * music->samplesize);

Sint64 destpos = music->start + dest_offset;

destpos -= dest_offset % sample_size;

if (destpos > music->stop)

return -1;

SDL_RWseek(music->src, destpos, RW_SEEK_SET);

return 0;

}

/* Return music duration in seconds */

static double WAV_Duration(void *context)

{

WAV_Music *music = (WAV_Music *)context;

Sint64 sample_size = music->spec.freq * music->samplesize;

return (double)(music->stop - music->start) / sample_size;

}

/* Close the given WAV stream */

static void WAV_Delete(void *context)

{

WAV_Music *music = (WAV_Music *)context;

/* Clean up associated data */

if (music->loops) {

SDL_free(music->loops);

}

if (music->stream) {

SDL_FreeAudioStream(music->stream);

}

if (music->buffer) {

SDL_free(music->buffer);

}

if (music->freesrc) {

SDL_RWclose(music->src);

}

SDL_free(music);

}

static SDL_bool ParseFMT(WAV_Music *wave, Uint32 chunk_length)

{

SDL_AudioSpec *spec = &wave->spec;

WaveFMT *format;

WaveFMTex *formatEx = NULL;

Uint8 *data;

Uint16 bitsamplerate;

SDL_bool loaded = SDL_FALSE;

if (chunk_length < sizeof(*format)) {

Mix_SetError("Wave format chunk too small");

return SDL_FALSE;

}

data = (Uint8 *)SDL_malloc(chunk_length);

if (!data) {

Mix_SetError("Out of memory");

return SDL_FALSE;

}

if (!SDL_RWread(wave->src, data, chunk_length, 1)) {

Mix_SetError("Couldn't read %d bytes from WAV file", chunk_length);

return SDL_FALSE;

}

format = (WaveFMT *)data;

wave->encoding = SDL_SwapLE16(format->encoding);

if (wave->encoding == EXT_CODE) {

formatEx = (WaveFMTex*)(data + sizeof(WaveFMT));

wave->encoding = (Uint16)SDL_SwapLE32(formatEx->subencoding);

}

/* Decode the audio data format */

switch (wave->encoding) {

case PCM_CODE:

case FLOAT_CODE:

/* We can understand this */

wave->decode = fetch_pcm;

break;

case uLAW_CODE:

/* , this */

wave->decode = fetch_ulaw;

break;

case ALAW_CODE:

/* , and this */

wave->decode = fetch_alaw;

break;

default:

/* but NOT this */

Mix_SetError("Unknown WAVE data format");

goto done;

}

spec->freq = (int)SDL_SwapLE32(format->frequency);

bitsamplerate = SDL_SwapLE16(format->bitspersample);

switch (bitsamplerate) {

case 8:

switch(wave->encoding) {

case PCM_CODE: spec->format = AUDIO_U8; break;

case ALAW_CODE: spec->format = AUDIO_S16; break;

case uLAW_CODE: spec->format = AUDIO_S16; break;

default: goto unknown_length;

}

break;

case 16:

switch(wave->encoding) {

case PCM_CODE: spec->format = AUDIO_S16; break;

default: goto unknown_length;

}

break;

case 24:

switch(wave->encoding) {

case PCM_CODE:

wave->decode = fetch_pcm24le;

spec->format = AUDIO_S32;

break;

default: goto unknown_length;

}

case 32:

switch(wave->encoding) {

case PCM_CODE: spec->format = AUDIO_S32; break;

case FLOAT_CODE: spec->format = AUDIO_F32; break;

default: goto unknown_length;

}

break;

case 64:

switch(wave->encoding) {

case FLOAT_CODE:

wave->decode = fetch_float64le;

spec->format = AUDIO_F32;

break;

default: goto unknown_length;

}

break;

default:

unknown_length:

Mix_SetError("Unknown PCM data format of %d-bit length", (int)bitsamplerate);

goto done;

}

spec->channels = (Uint8) SDL_SwapLE16(format->channels);

spec->samples = 4096; /* Good default buffer size */

wave->samplesize = spec->channels * (bitsamplerate / 8);

/* SDL_CalculateAudioSpec */

spec->size = SDL_AUDIO_BITSIZE(spec->format) / 8;

spec->size *= spec->channels;

spec->size *= spec->samples;

loaded = SDL_TRUE;

done:

SDL_free(data);

return loaded;

}

static SDL_bool ParseDATA(WAV_Music *wave, Uint32 chunk_length)

{

wave->start = SDL_RWtell(wave->src);

wave->stop = wave->start + chunk_length;

SDL_RWseek(wave->src, chunk_length, RW_SEEK_CUR);

return SDL_TRUE;

}

static SDL_bool AddLoopPoint(WAV_Music *wave, Uint32 play_count, Uint32 start, Uint32 stop)

{

WAVLoopPoint *loop;

WAVLoopPoint *loops = SDL_realloc(wave->loops, (wave->numloops + 1) * sizeof(*wave->loops));

if (!loops) {

Mix_SetError("Out of memory");

return SDL_FALSE;

}

loop = &loops[ wave->numloops ];

loop->start = start;

loop->stop = stop;

loop->initial_play_count = play_count;

loop->current_play_count = play_count;

wave->loops = loops;

++wave->numloops;

return SDL_TRUE;

}

static SDL_bool ParseSMPL(WAV_Music *wave, Uint32 chunk_length)

{

SamplerChunk *chunk;

Uint8 *data;

Uint32 i;

SDL_bool loaded = SDL_FALSE;

data = (Uint8 *)SDL_malloc(chunk_length);

if (!data) {

Mix_SetError("Out of memory");

return SDL_FALSE;

}

if (!SDL_RWread(wave->src, data, chunk_length, 1)) {

Mix_SetError("Couldn't read %d bytes from WAV file", chunk_length);

return SDL_FALSE;

}

chunk = (SamplerChunk *)data;

for (i = 0; i < SDL_SwapLE32(chunk->sample_loops); ++i) {

const Uint32 LOOP_TYPE_FORWARD = 0;

Uint32 loop_type = SDL_SwapLE32(chunk->loops[i].type);

if (loop_type == LOOP_TYPE_FORWARD) {

AddLoopPoint(wave, SDL_SwapLE32(chunk->loops[i].play_count), SDL_SwapLE32(chunk->loops[i].start), SDL_SwapLE32(chunk->loops[i].end));

}

}

loaded = SDL_TRUE;

SDL_free(data);

return loaded;

}

static SDL_bool LoadWAVMusic(WAV_Music *wave)

{

SDL_RWops *src = wave->src;

Uint32 chunk_type;

Uint32 chunk_length;

SDL_bool found_FMT = SDL_FALSE;

SDL_bool found_DATA = SDL_FALSE;

/* WAV magic header */

Uint32 wavelen;

Uint32 WAVEmagic;

/* Check the magic header */

wavelen = SDL_ReadLE32(src);

WAVEmagic = SDL_ReadLE32(src);

(void)wavelen; /* unused */

(void)WAVEmagic; /* unused */

/* Read the chunks */

for (; ;) {

chunk_type = SDL_ReadLE32(src);

chunk_length = SDL_ReadLE32(src);

if (chunk_length == 0)

break;

switch (chunk_type)

{

case FMT:

found_FMT = SDL_TRUE;

if (!ParseFMT(wave, chunk_length))

return SDL_FALSE;

break;

case DATA:

found_DATA = SDL_TRUE;

if (!ParseDATA(wave, chunk_length))

return SDL_FALSE;

break;

case SMPL:

if (!ParseSMPL(wave, chunk_length))

return SDL_FALSE;

break;

default:

SDL_RWseek(src, chunk_length, RW_SEEK_CUR);

break;

}

}

if (!found_FMT) {

Mix_SetError("Bad WAV file (no FMT chunk)");

return SDL_FALSE;

}

if (!found_DATA) {

Mix_SetError("Bad WAV file (no DATA chunk)");

return SDL_FALSE;

}

return SDL_TRUE;

}

/* I couldn't get SANE_to_double() to work, so I stole this from libsndfile.

* I don't pretend to fully understand it.

*/

static Uint32 SANE_to_Uint32 (Uint8 *sanebuf)

{

/* Negative number? */

if (sanebuf[0] & 0x80)

return 0;

/* Less than 1? */

if (sanebuf[0] <= 0x3F)

return 1;

/* Way too big? */

if (sanebuf[0] > 0x40)

return 0x4000000;

/* Still too big? */

if (sanebuf[0] == 0x40 && sanebuf[1] > 0x1C)

return 800000000;

return (Uint32)(((sanebuf[2] << 23) | (sanebuf[3] << 15) | (sanebuf[4] << 7) |

(sanebuf[5] >> 1)) >> (29 - sanebuf[1]));

}

static SDL_bool LoadAIFFMusic(WAV_Music *wave)

{

SDL_RWops *src = wave->src;

SDL_AudioSpec *spec = &wave->spec;

SDL_bool found_SSND = SDL_FALSE;

SDL_bool found_COMM = SDL_FALSE;

SDL_bool found_FVER = SDL_FALSE;

SDL_bool is_AIFC = SDL_FALSE;

Uint32 chunk_type;

Uint32 chunk_length;

Sint64 next_chunk = 0;

Sint64 file_length;

/* AIFF magic header */

Uint32 AIFFmagic;

/* SSND chunk */

Uint32 offset;

Uint32 blocksize;

/* COMM format chunk */

Uint16 channels = 0;

Uint32 numsamples = 0;

Uint16 samplesize = 0;

Uint8 sane_freq[10];

Uint32 frequency = 0;

Uint32 AIFCVersion1 = 0;

Uint32 compressionType = 0;

file_length = SDL_RWsize(src);

/* Check the magic header */

chunk_length = SDL_ReadBE32(src);

AIFFmagic = SDL_ReadLE32(src);

if (AIFFmagic != AIFF && AIFFmagic != AIFC) {

Mix_SetError("Unrecognized file type (not AIFF or AIFC)");

return SDL_FALSE;

}

if (AIFFmagic == AIFC) {

is_AIFC = SDL_TRUE;

}

/* From what I understand of the specification, chunks may appear in

* any order, and we should just ignore unknown ones.

*

* TODO: Better sanity-checking. E.g. what happens if the AIFF file

* contains compressed sound data?

*/

do {

chunk_type = SDL_ReadLE32(src);

chunk_length = SDL_ReadBE32(src);

next_chunk = SDL_RWtell(src) + chunk_length;

if (chunk_length % 2) {

next_chunk++;

}

switch (chunk_type) {

case SSND:

found_SSND = SDL_TRUE;

offset = SDL_ReadBE32(src);

blocksize = SDL_ReadBE32(src);

wave->start = SDL_RWtell(src) + offset;

(void)blocksize; /* unused */

break;

case FVER:

found_FVER = SDL_TRUE;

AIFCVersion1 = SDL_ReadBE32(src);

(void)AIFCVersion1; /* unused */

break;

case MARK:

case INST:

/* Just skip those chunks */

break;

case NAME:

case AUTH:

case _c__:

/* Just skip those chunks */

break;

case COMM:

found_COMM = SDL_TRUE;

/* Read the audio data format chunk */

channels = SDL_ReadBE16(src);

numsamples = SDL_ReadBE32(src);

samplesize = SDL_ReadBE16(src);

SDL_RWread(src, sane_freq, sizeof(sane_freq), 1);

frequency = SANE_to_Uint32(sane_freq);

if (is_AIFC) {

compressionType = SDL_ReadLE32(src);

/* here must be a "compressionName" which is a padded string */

}

break;

default:

/* Unknown/unsupported chunk: we just skip over */

break;

}

} while (next_chunk < file_length && SDL_RWseek(src, next_chunk, RW_SEEK_SET) != -1);

if (!found_SSND) {

Mix_SetError("Bad AIFF/AIFF-C file (no SSND chunk)");

return SDL_FALSE;

}

if (!found_COMM) {

Mix_SetError("Bad AIFF/AIFF-C file (no COMM chunk)");

return SDL_FALSE;

}

if (is_AIFC && !found_FVER) {

Mix_SetError("Bad AIFF-C file (no FVER chunk)");

return SDL_FALSE;

}

wave->samplesize = channels * (samplesize / 8);

wave->stop = wave->start + channels * numsamples * (samplesize / 8);

/* Decode the audio data format */

SDL_memset(spec, 0, (sizeof *spec));

spec->freq = (int)frequency;