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music_wav.c
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/*
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SDL_mixer: An audio mixer library based on the SDL library
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Copyright (C) 1997-2020 Sam Lantinga <slouken@libsdl.org>
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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.
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*/
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#ifdef MUSIC_WAV
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/* This file supports streaming WAV files */
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#include "music_wav.h"
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typedef struct {
SDL_bool active;
Uint32 start;
Uint32 stop;
Uint32 initial_play_count;
Uint32 current_play_count;
} WAVLoopPoint;
typedef struct {
SDL_RWops *src;
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int freesrc;
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SDL_AudioSpec spec;
int volume;
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int play_count;
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Sint64 start;
Sint64 stop;
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Sint64 samplesize;
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Uint8 *buffer;
SDL_AudioStream *stream;
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unsigned int numloops;
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WAVLoopPoint *loops;
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Mix_MusicMetaTags tags;
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Uint16 encoding;
int (*decode)(void *music, int length);
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} WAV_Music;
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/*
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 */
/*******************************************/
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#define RIFF 0x46464952 /* "RIFF" */
#define WAVE 0x45564157 /* "WAVE" */
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#define FMT 0x20746D66 /* "fmt " */
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#define DATA 0x61746164 /* "data" */
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#define SMPL 0x6c706d73 /* "smpl" */
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#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 */
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#define WAVE_MONO 1
#define WAVE_STEREO 2
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typedef struct {
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/* Not saved in the chunk we read:
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Uint32 chunkID;
Uint32 chunkLen;
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*/
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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 */
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} WaveFMT;
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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;
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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;
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SampleLoop loops[1];
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} SamplerChunk;
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/*********************************************/
/* Define values for AIFF (IFF audio) format */
/*********************************************/
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#define FORM 0x4d524f46 /* "FORM" */
#define AIFF 0x46464941 /* "AIFF" */
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#define AIFC 0x43464941 /* "AIFС" */
#define FVER 0x52455646 /* "FVER" */
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#define SSND 0x444e5353 /* "SSND" */
#define COMM 0x4d4d4f43 /* "COMM" */
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#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" */
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/* Function to load the WAV/AIFF stream */
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static SDL_bool LoadWAVMusic(WAV_Music *wave);
static SDL_bool LoadAIFFMusic(WAV_Music *wave);
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static void WAV_Delete(void *context);
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static int fetch_pcm(void *context, int length);
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/* Load a WAV stream from the given RWops object */
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static void *WAV_CreateFromRW(SDL_RWops *src, int freesrc)
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{
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WAV_Music *music;
Uint32 magic;
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SDL_bool loaded = SDL_FALSE;
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music = (WAV_Music *)SDL_calloc(1, sizeof(*music));
if (!music) {
SDL_OutOfMemory();
return NULL;
}
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music->src = src;
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music->volume = MIX_MAX_VOLUME;
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/* Default decoder is PCM */
music->decode = fetch_pcm;
music->encoding = PCM_CODE;
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magic = SDL_ReadLE32(src);
if (magic == RIFF || magic == WAVE) {
loaded = LoadWAVMusic(music);
} else if (magic == FORM) {
loaded = LoadAIFFMusic(music);
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} else {
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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;
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}
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music->freesrc = freesrc;
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return music;
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}
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static void WAV_SetVolume(void *context, int volume)
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{
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WAV_Music *music = (WAV_Music *)context;
music->volume = volume;
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}
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static int WAV_GetVolume(void *context)
{
WAV_Music *music = (WAV_Music *)context;
return music->volume;
}
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/* Start playback of a given WAV stream */
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static int WAV_Play(void *context, int play_count)
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{
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WAV_Music *music = (WAV_Music *)context;
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unsigned int i;
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for (i = 0; i < music->numloops; ++i) {
WAVLoopPoint *loop = &music->loops[i];
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loop->active = SDL_TRUE;
loop->current_play_count = loop->initial_play_count;
}
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music->play_count = play_count;
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if (SDL_RWseek(music->src, music->start, RW_SEEK_SET) < 0) {
return -1;
}
return 0;
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}
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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);
}
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/* Play some of a stream previously started with WAV_Play() */
static int WAV_GetSome(void *context, void *data, int bytes, SDL_bool *done)
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{
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WAV_Music *music = (WAV_Music *)context;
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Sint64 pos, stop;
WAVLoopPoint *loop;
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Sint64 loop_start = music->start;
Sint64 loop_stop = music->stop;
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SDL_bool looped = SDL_FALSE;
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SDL_bool at_end = SDL_FALSE;
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unsigned int i;
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int filled, amount, result;
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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;
508
loop = NULL;
509
510
for (i = 0; i < music->numloops; ++i) {
loop = &music->loops[i];
511
if (loop->active) {
512
const int bytes_per_sample = (SDL_AUDIO_BITSIZE(music->spec.format) / 8) * music->spec.channels;
513
514
loop_start = music->start + loop->start * (Uint32)bytes_per_sample;
loop_stop = music->start + (loop->stop + 1) * (Uint32)bytes_per_sample;
515
if (pos >= loop_start && pos < loop_stop) {
516
517
stop = loop_stop;
break;
518
}
519
520
521
522
}
loop = NULL;
}
523
amount = (int)music->spec.size;
524
525
526
if ((stop - pos) < amount) {
amount = (int)(stop - pos);
}
527
528
amount = music->decode(music, amount);
529
530
531
532
if (amount > 0) {
result = SDL_AudioStreamPut(music->stream, music->buffer, amount);
if (result < 0) {
return -1;
533
534
}
} else {
535
/* We might be looping, continue */
536
at_end = SDL_TRUE;
537
538
}
539
if (loop && SDL_RWtell(music->src) >= stop) {
540
541
if (loop->current_play_count == 1) {
loop->active = SDL_FALSE;
542
} else {
543
544
if (loop->current_play_count > 0) {
--loop->current_play_count;
545
}
546
SDL_RWseek(music->src, loop_start, RW_SEEK_SET);
547
looped = SDL_TRUE;
548
549
}
}
550
551
if (!looped && (at_end || SDL_RWtell(music->src) >= music->stop)) {
552
553
554
555
556
557
558
559
560
561
562
563
564
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;
}
}
}
565
566
567
568
/* We'll get called again in the case where we looped or have more data */
return 0;
}
569
570
571
572
573
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);
574
575
}
576
577
578
579
580
581
582
583
584
585
586
587
588
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;
}
589
590
591
592
593
594
595
static double WAV_Tell(void *context)
{
WAV_Music *music = (WAV_Music *)context;
Sint64 phys_pos = SDL_RWtell(music->src);
return (double)(phys_pos - music->start) / (double)(music->spec.freq * music->samplesize);
}
596
597
598
599
600
601
602
603
/* 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;
}
604
605
606
607
608
609
static const char* WAV_GetMetaTag(void *context, Mix_MusicMetaTag tag_type)
{
WAV_Music *music = (WAV_Music *)context;
return meta_tags_get(&music->tags, tag_type);
}
610
/* Close the given WAV stream */
611
static void WAV_Delete(void *context)
612
{
613
WAV_Music *music = (WAV_Music *)context;
614
615
/* Clean up associated data */
616
meta_tags_clear(&music->tags);
617
618
619
620
621
if (music->loops) {
SDL_free(music->loops);
}
if (music->stream) {
SDL_FreeAudioStream(music->stream);
622
}
623
624
if (music->buffer) {
SDL_free(music->buffer);
625
}
626
627
if (music->freesrc) {
SDL_RWclose(music->src);
628
}
629
SDL_free(music);
630
631
}
632
static SDL_bool ParseFMT(WAV_Music *wave, Uint32 chunk_length)
633
{
634
635
SDL_AudioSpec *spec = &wave->spec;
WaveFMT *format;
636
WaveFMTex *formatEx = NULL;
637
Uint8 *data;
638
Uint16 bitsamplerate;
639
640
641
642
643
SDL_bool loaded = SDL_FALSE;
if (chunk_length < sizeof(*format)) {
Mix_SetError("Wave format chunk too small");
return SDL_FALSE;
644
645
}
646
647
648
649
650
651
652
653
654
655
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;
656
657
658
659
660
661
662
663
wave->encoding = SDL_SwapLE16(format->encoding);
if (wave->encoding == EXT_CODE) {
formatEx = (WaveFMTex*)(data + sizeof(WaveFMT));
wave->encoding = (Uint16)SDL_SwapLE32(formatEx->subencoding);
}
664
/* Decode the audio data format */
665
switch (wave->encoding) {
666
case PCM_CODE:
667
case FLOAT_CODE:
668
/* We can understand this */
669
670
671
672
673
674
675
676
677
wave->decode = fetch_pcm;
break;
case uLAW_CODE:
/* , this */
wave->decode = fetch_ulaw;
break;
case ALAW_CODE:
/* , and this */
wave->decode = fetch_alaw;
678
679
break;
default:
680
/* but NOT this */
681
682
683
Mix_SetError("Unknown WAVE data format");
goto done;
}
684
685
686
spec->freq = (int)SDL_SwapLE32(format->frequency);
bitsamplerate = SDL_SwapLE16(format->bitspersample);
switch (bitsamplerate) {
687
case 8:
688
689
690
691
692
693
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;
}
694
695
break;
case 16:
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
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;
}
724
725
break;
default:
726
727
unknown_length:
Mix_SetError("Unknown PCM data format of %d-bit length", (int)bitsamplerate);
728
729
730
731
goto done;
}
spec->channels = (Uint8) SDL_SwapLE16(format->channels);
spec->samples = 4096; /* Good default buffer size */
732
wave->samplesize = spec->channels * (bitsamplerate / 8);
733
734
735
736
/* SDL_CalculateAudioSpec */
spec->size = SDL_AUDIO_BITSIZE(spec->format) / 8;
spec->size *= spec->channels;
spec->size *= spec->samples;
737
738
739
740
741
742
743
744
loaded = SDL_TRUE;
done:
SDL_free(data);
return loaded;
}
745
static SDL_bool ParseDATA(WAV_Music *wave, Uint32 chunk_length)
746
747
748
749
750
751
752
{
wave->start = SDL_RWtell(wave->src);
wave->stop = wave->start + chunk_length;
SDL_RWseek(wave->src, chunk_length, RW_SEEK_CUR);
return SDL_TRUE;
}
753
static SDL_bool AddLoopPoint(WAV_Music *wave, Uint32 play_count, Uint32 start, Uint32 stop)
754
755
{
WAVLoopPoint *loop;
756
WAVLoopPoint *loops = SDL_realloc(wave->loops, (wave->numloops + 1) * sizeof(*wave->loops));
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
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;
}
773
static SDL_bool ParseSMPL(WAV_Music *wave, Uint32 chunk_length)
774
775
776
{
SamplerChunk *chunk;
Uint8 *data;
777
Uint32 i;
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
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));
796
}
797
798
799
800
801
802
803
}
loaded = SDL_TRUE;
SDL_free(data);
return loaded;
}
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
static void read_meta_field(Mix_MusicMetaTags *tags, Mix_MusicMetaTag tag_type, size_t *i, Uint32 chunk_length, Uint8 *data, size_t fieldOffset)
{
Uint32 len = 0;
int isID3 = fieldOffset == 7;
char *field = NULL;
*i += 4;
len = isID3 ?
SDL_SwapBE32(*((Uint32 *)(data + *i))) : /* ID3 */
SDL_SwapLE32(*((Uint32 *)(data + *i))); /* LIST */
if (len > chunk_length) {
return; /* Do nothing due to broken lenght */
}
*i += fieldOffset;
field = (char *)SDL_malloc(len + 1);
SDL_memset(field, 0, (len + 1));
SDL_strlcpy(field, (char *)(data + *i), isID3 ? len - 1 : len);
*i += len;
meta_tags_set(tags, tag_type, field);
SDL_free(field);
}
static SDL_bool ParseLIST(WAV_Music *wave, Uint32 chunk_length)
{
SDL_bool loaded = SDL_FALSE;
Uint8 *data;
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;
}
if (SDL_strncmp((char *)data, "INFO", 4) == 0) {
size_t i = 4;
for (i = 4; i < chunk_length - 4;) {
if(SDL_strncmp((char *)(data + i), "INAM", 4) == 0) {
read_meta_field(&wave->tags, MIX_META_TITLE, &i, chunk_length, data, 4);
continue;
} else if(SDL_strncmp((char *)(data + i), "IART", 4) == 0) {
read_meta_field(&wave->tags, MIX_META_ARTIST, &i, chunk_length, data, 4);
continue;
} else if(SDL_strncmp((char *)(data + i), "IALB", 4) == 0) {
read_meta_field(&wave->tags, MIX_META_ALBUM, &i, chunk_length, data, 4);
continue;
} else if (SDL_strncmp((char *)(data + i), "BCPR", 4) == 0) {
read_meta_field(&wave->tags, MIX_META_COPYRIGHT, &i, chunk_length, data, 4);
continue;
}
i++;
}
loaded = SDL_TRUE;
}
/* done: */
SDL_free(data);
return loaded;
}
868
static SDL_bool LoadWAVMusic(WAV_Music *wave)
869
870
871
872
873
874
875
876
877
878
{
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;
879
880
meta_tags_init(&wave->tags);
881
882
883
884
/* Check the magic header */
wavelen = SDL_ReadLE32(src);
WAVEmagic = SDL_ReadLE32(src);
885
886
887
(void)wavelen; /* unused */
(void)WAVEmagic; /* unused */
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
/* 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;
912
913
914
915
case LIST:
if (!ParseLIST(wave, chunk_length))
return SDL_FALSE;
break;
916
917
918
919
default:
SDL_RWseek(src, chunk_length, RW_SEEK_CUR);
break;
}
920
}
921
922
923
924
if (!found_FMT) {
Mix_SetError("Bad WAV file (no FMT chunk)");
return SDL_FALSE;
925
}
926
927
928
929
930
931
932
if (!found_DATA) {
Mix_SetError("Bad WAV file (no DATA chunk)");
return SDL_FALSE;
}
return SDL_TRUE;
933
934
}
935
936
937
938
939
/* 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)
940
{
941
942
943
/* Negative number? */
if (sanebuf[0] & 0x80)
return 0;
944
945
946
947
/* Less than 1? */
if (sanebuf[0] <= 0x3F)
return 1;
948
949
950
951
/* Way too big? */
if (sanebuf[0] > 0x40)
return 0x4000000;
952
953
954
955
/* Still too big? */
if (sanebuf[0] == 0x40 && sanebuf[1] > 0x1C)
return 800000000;
956
957
958
return (Uint32)(((sanebuf[2] << 23) | (sanebuf[3] << 15) | (sanebuf[4] << 7) |
(sanebuf[5] >> 1)) >> (29 - sanebuf[1]));
959
960
}
961
static SDL_bool LoadAIFFMusic(WAV_Music *wave)
962
{
963
964
965
966
SDL_RWops *src = wave->src;
SDL_AudioSpec *spec = &wave->spec;
SDL_bool found_SSND = SDL_FALSE;
SDL_bool found_COMM = SDL_FALSE;
967
968
SDL_bool found_FVER = SDL_FALSE;
SDL_bool is_AIFC = SDL_FALSE;
969
970
971
Uint32 chunk_type;
Uint32 chunk_length;
972
973
974
Sint64 next_chunk = 0;
Sint64 file_length;
975
976
977
978
979
980
981
982
983
984
985
/* 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;
986
987
Uint32 AIFCVersion1 = 0;
Uint32 compressionType = 0;
988
char *chunk_buffer;
989
990
991
file_length = SDL_RWsize(src);
992
/* Check the magic header */
993
994
chunk_length = SDL_ReadBE32(src);
AIFFmagic = SDL_ReadLE32(src);
995
996
if (AIFFmagic != AIFF && AIFFmagic != AIFC) {
Mix_SetError("Unrecognized file type (not AIFF or AIFC)");
997
return SDL_FALSE;
998
}
999
1000
if (AIFFmagic == AIFC) {
is_AIFC = SDL_TRUE;