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music_wav.c
1110 lines (993 loc) · 33.9 KB
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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) 1997-2019 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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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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}
/* 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;
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loop = NULL;
502
503
for (i = 0; i < music->numloops; ++i) {
loop = &music->loops[i];
504
if (loop->active) {
505
const int bytes_per_sample = (SDL_AUDIO_BITSIZE(music->spec.format) / 8) * music->spec.channels;
506
507
loop_start = music->start + loop->start * (Uint32)bytes_per_sample;
loop_stop = music->start + (loop->stop + 1) * (Uint32)bytes_per_sample;
508
if (pos >= loop_start && pos < loop_stop) {
509
510
stop = loop_stop;
break;
511
}
512
513
514
515
}
loop = NULL;
}
516
amount = (int)music->spec.size;
517
518
519
if ((stop - pos) < amount) {
amount = (int)(stop - pos);
}
520
521
amount = music->decode(music, amount);
522
523
524
525
if (amount > 0) {
result = SDL_AudioStreamPut(music->stream, music->buffer, amount);
if (result < 0) {
return -1;
526
527
}
} else {
528
/* We might be looping, continue */
529
at_end = SDL_TRUE;
530
531
}
532
if (loop && SDL_RWtell(music->src) >= stop) {
533
534
if (loop->current_play_count == 1) {
loop->active = SDL_FALSE;
535
} else {
536
537
if (loop->current_play_count > 0) {
--loop->current_play_count;
538
}
539
SDL_RWseek(music->src, loop_start, RW_SEEK_SET);
540
looped = SDL_TRUE;
541
542
}
}
543
544
if (!looped && (at_end || SDL_RWtell(music->src) >= music->stop)) {
545
546
547
548
549
550
551
552
553
554
555
556
557
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;
}
}
}
558
559
560
561
/* We'll get called again in the case where we looped or have more data */
return 0;
}
562
563
564
565
566
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);
567
568
}
569
570
571
572
573
574
575
576
577
578
579
580
581
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;
}
582
583
584
585
586
587
588
589
/* 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;
}
590
/* Close the given WAV stream */
591
static void WAV_Delete(void *context)
592
{
593
WAV_Music *music = (WAV_Music *)context;
594
595
/* Clean up associated data */
596
597
598
599
600
if (music->loops) {
SDL_free(music->loops);
}
if (music->stream) {
SDL_FreeAudioStream(music->stream);
601
}
602
603
if (music->buffer) {
SDL_free(music->buffer);
604
}
605
606
if (music->freesrc) {
SDL_RWclose(music->src);
607
}
608
SDL_free(music);
609
610
}
611
static SDL_bool ParseFMT(WAV_Music *wave, Uint32 chunk_length)
612
{
613
614
SDL_AudioSpec *spec = &wave->spec;
WaveFMT *format;
615
WaveFMTex *formatEx = NULL;
616
Uint8 *data;
617
Uint16 bitsamplerate;
618
619
620
621
622
SDL_bool loaded = SDL_FALSE;
if (chunk_length < sizeof(*format)) {
Mix_SetError("Wave format chunk too small");
return SDL_FALSE;
623
624
}
625
626
627
628
629
630
631
632
633
634
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;
635
636
637
638
639
640
641
642
wave->encoding = SDL_SwapLE16(format->encoding);
if (wave->encoding == EXT_CODE) {
formatEx = (WaveFMTex*)(data + sizeof(WaveFMT));
wave->encoding = (Uint16)SDL_SwapLE32(formatEx->subencoding);
}
643
/* Decode the audio data format */
644
switch (wave->encoding) {
645
case PCM_CODE:
646
case FLOAT_CODE:
647
/* We can understand this */
648
649
650
651
652
653
654
655
656
wave->decode = fetch_pcm;
break;
case uLAW_CODE:
/* , this */
wave->decode = fetch_ulaw;
break;
case ALAW_CODE:
/* , and this */
wave->decode = fetch_alaw;
657
658
break;
default:
659
/* but NOT this */
660
661
662
Mix_SetError("Unknown WAVE data format");
goto done;
}
663
664
665
spec->freq = (int)SDL_SwapLE32(format->frequency);
bitsamplerate = SDL_SwapLE16(format->bitspersample);
switch (bitsamplerate) {
666
case 8:
667
668
669
670
671
672
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;
}
673
674
break;
case 16:
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
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;
}
703
704
break;
default:
705
706
unknown_length:
Mix_SetError("Unknown PCM data format of %d-bit length", (int)bitsamplerate);
707
708
709
710
goto done;
}
spec->channels = (Uint8) SDL_SwapLE16(format->channels);
spec->samples = 4096; /* Good default buffer size */
711
wave->samplesize = spec->channels * (bitsamplerate / 8);
712
713
714
715
/* SDL_CalculateAudioSpec */
spec->size = SDL_AUDIO_BITSIZE(spec->format) / 8;
spec->size *= spec->channels;
spec->size *= spec->samples;
716
717
718
719
720
721
722
723
loaded = SDL_TRUE;
done:
SDL_free(data);
return loaded;
}
724
static SDL_bool ParseDATA(WAV_Music *wave, Uint32 chunk_length)
725
726
727
728
729
730
731
{
wave->start = SDL_RWtell(wave->src);
wave->stop = wave->start + chunk_length;
SDL_RWseek(wave->src, chunk_length, RW_SEEK_CUR);
return SDL_TRUE;
}
732
static SDL_bool AddLoopPoint(WAV_Music *wave, Uint32 play_count, Uint32 start, Uint32 stop)
733
734
{
WAVLoopPoint *loop;
735
WAVLoopPoint *loops = SDL_realloc(wave->loops, (wave->numloops + 1) * sizeof(*wave->loops));
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
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;
}
752
static SDL_bool ParseSMPL(WAV_Music *wave, Uint32 chunk_length)
753
754
755
{
SamplerChunk *chunk;
Uint8 *data;
756
Uint32 i;
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
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));
775
}
776
777
778
779
780
781
782
}
loaded = SDL_TRUE;
SDL_free(data);
return loaded;
}
783
static SDL_bool LoadWAVMusic(WAV_Music *wave)
784
785
786
787
788
789
790
791
792
793
794
795
796
797
{
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);
798
799
800
(void)wavelen; /* unused */
(void)WAVEmagic; /* unused */
801
802
803
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
/* 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;
}
829
}
830
831
832
833
if (!found_FMT) {
Mix_SetError("Bad WAV file (no FMT chunk)");
return SDL_FALSE;
834
}
835
836
837
838
839
840
841
if (!found_DATA) {
Mix_SetError("Bad WAV file (no DATA chunk)");
return SDL_FALSE;
}
return SDL_TRUE;
842
843
}
844
845
846
847
848
/* 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)
849
{
850
851
852
/* Negative number? */
if (sanebuf[0] & 0x80)
return 0;
853
854
855
856
/* Less than 1? */
if (sanebuf[0] <= 0x3F)
return 1;
857
858
859
860
/* Way too big? */
if (sanebuf[0] > 0x40)
return 0x4000000;
861
862
863
864
/* Still too big? */
if (sanebuf[0] == 0x40 && sanebuf[1] > 0x1C)
return 800000000;
865
866
867
return (Uint32)(((sanebuf[2] << 23) | (sanebuf[3] << 15) | (sanebuf[4] << 7) |
(sanebuf[5] >> 1)) >> (29 - sanebuf[1]));
868
869
}
870
static SDL_bool LoadAIFFMusic(WAV_Music *wave)
871
{
872
873
874
875
SDL_RWops *src = wave->src;
SDL_AudioSpec *spec = &wave->spec;
SDL_bool found_SSND = SDL_FALSE;
SDL_bool found_COMM = SDL_FALSE;
876
877
SDL_bool found_FVER = SDL_FALSE;
SDL_bool is_AIFC = SDL_FALSE;
878
879
880
Uint32 chunk_type;
Uint32 chunk_length;
881
882
883
Sint64 next_chunk = 0;
Sint64 file_length;
884
885
886
887
888
889
890
891
892
893
894
/* 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;
895
896
Uint32 AIFCVersion1 = 0;
Uint32 compressionType = 0;
897
898
899
file_length = SDL_RWsize(src);
900
/* Check the magic header */
901
902
chunk_length = SDL_ReadBE32(src);
AIFFmagic = SDL_ReadLE32(src);
903
904
if (AIFFmagic != AIFF && AIFFmagic != AIFC) {
Mix_SetError("Unrecognized file type (not AIFF or AIFC)");
905
return SDL_FALSE;
906
}
907
908
909
if (AIFFmagic == AIFC) {
is_AIFC = SDL_TRUE;
}
910
911
/* From what I understand of the specification, chunks may appear in
912
* any order, and we should just ignore unknown ones.
913
914
915
*
* TODO: Better sanity-checking. E.g. what happens if the AIFF file
* contains compressed sound data?
916
*/
917
918
919
920
do {
chunk_type = SDL_ReadLE32(src);
chunk_length = SDL_ReadBE32(src);
next_chunk = SDL_RWtell(src) + chunk_length;
921
922
923
924
if (chunk_length % 2) {
next_chunk++;
}
925
926
switch (chunk_type) {
927
case SSND:
928
929
930
931
found_SSND = SDL_TRUE;
offset = SDL_ReadBE32(src);
blocksize = SDL_ReadBE32(src);
wave->start = SDL_RWtell(src) + offset;
932
933
934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
(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 */
950
951
952
break;
case COMM:
953
found_COMM = SDL_TRUE;
954
955
/* Read the audio data format chunk */
956
957
958
channels = SDL_ReadBE16(src);
numsamples = SDL_ReadBE32(src);
samplesize = SDL_ReadBE16(src);
959
SDL_RWread(src, sane_freq, sizeof(sane_freq), 1);
960
frequency = SANE_to_Uint32(sane_freq);
961
962
963
964
if (is_AIFC) {
compressionType = SDL_ReadLE32(src);
/* here must be a "compressionName" which is a padded string */
}
965
966
967
break;
default:
968
/* Unknown/unsupported chunk: we just skip over */
969
970
break;
}
971
} while (next_chunk < file_length && SDL_RWseek(src, next_chunk, RW_SEEK_SET) != -1);
972
973
if (!found_SSND) {
974
Mix_SetError("Bad AIFF/AIFF-C file (no SSND chunk)");
975
return SDL_FALSE;
976
977
978
}
if (!found_COMM) {
979
Mix_SetError("Bad AIFF/AIFF-C file (no COMM chunk)");
980
return SDL_FALSE;
981
982
}
983
984
985
986
987
988
989
if (is_AIFC && !found_FVER) {
Mix_SetError("Bad AIFF-C file (no FVER chunk)");
return SDL_FALSE;
}
wave->samplesize = channels * (samplesize / 8);
990
wave->stop = wave->start + channels * numsamples * (samplesize / 8);
991
992
/* Decode the audio data format */
993
SDL_memset(spec, 0, (sizeof *spec));
994
spec->freq = (int)frequency;
995
switch (samplesize) {
996
case 8:
997
998
999
1000
if (!is_AIFC)
spec->format = AUDIO_S8;
else switch (compressionType) {
case raw_: spec->format = AUDIO_U8; break;