src/audio/SDL_audiocvt.c
author Ryan C. Gordon <icculus@icculus.org>
Wed, 11 Oct 2017 02:03:05 -0400
changeset 11591 c79a9f64ddb2
parent 11590 ca6aa7f5488d
child 11592 61260a51fc60
permissions -rw-r--r--
audio: Make sure audio stream resampling doesn't overflow buffers.
slouken@0
     1
/*
slouken@5535
     2
  Simple DirectMedia Layer
slouken@10737
     3
  Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
slouken@0
     4
slouken@5535
     5
  This software is provided 'as-is', without any express or implied
slouken@5535
     6
  warranty.  In no event will the authors be held liable for any damages
slouken@5535
     7
  arising from the use of this software.
slouken@0
     8
slouken@5535
     9
  Permission is granted to anyone to use this software for any purpose,
slouken@5535
    10
  including commercial applications, and to alter it and redistribute it
slouken@5535
    11
  freely, subject to the following restrictions:
slouken@0
    12
slouken@5535
    13
  1. The origin of this software must not be misrepresented; you must not
slouken@5535
    14
     claim that you wrote the original software. If you use this software
slouken@5535
    15
     in a product, an acknowledgment in the product documentation would be
slouken@5535
    16
     appreciated but is not required.
slouken@5535
    17
  2. Altered source versions must be plainly marked as such, and must not be
slouken@5535
    18
     misrepresented as being the original software.
slouken@5535
    19
  3. This notice may not be removed or altered from any source distribution.
slouken@0
    20
*/
icculus@8093
    21
#include "../SDL_internal.h"
slouken@2728
    22
slouken@0
    23
/* Functions for audio drivers to perform runtime conversion of audio format */
slouken@0
    24
icculus@11319
    25
#include "SDL.h"
slouken@0
    26
#include "SDL_audio.h"
icculus@1982
    27
#include "SDL_audio_c.h"
slouken@0
    28
slouken@10773
    29
#include "SDL_loadso.h"
icculus@6281
    30
#include "SDL_assert.h"
icculus@10757
    31
#include "../SDL_dataqueue.h"
icculus@10835
    32
#include "SDL_cpuinfo.h"
icculus@6281
    33
icculus@11583
    34
#define DEBUG_AUDIOSTREAM 0
icculus@11583
    35
icculus@10835
    36
#ifdef __SSE3__
icculus@10835
    37
#define HAVE_SSE3_INTRINSICS 1
icculus@10832
    38
#endif
icculus@10832
    39
icculus@10832
    40
#if HAVE_SSE3_INTRINSICS
icculus@11405
    41
/* Convert from stereo to mono. Average left and right. */
icculus@10832
    42
static void SDLCALL
icculus@10832
    43
SDL_ConvertStereoToMono_SSE3(SDL_AudioCVT * cvt, SDL_AudioFormat format)
icculus@10832
    44
{
icculus@10832
    45
    float *dst = (float *) cvt->buf;
icculus@10832
    46
    const float *src = dst;
icculus@10832
    47
    int i = cvt->len_cvt / 8;
icculus@10832
    48
icculus@10832
    49
    LOG_DEBUG_CONVERT("stereo", "mono (using SSE3)");
icculus@10832
    50
    SDL_assert(format == AUDIO_F32SYS);
icculus@10832
    51
icculus@10832
    52
    /* We can only do this if dst is aligned to 16 bytes; since src is the
icculus@10832
    53
       same pointer and it moves by 2, it can't be forcibly aligned. */
icculus@10832
    54
    if ((((size_t) dst) & 15) == 0) {
icculus@10832
    55
        /* Aligned! Do SSE blocks as long as we have 16 bytes available. */
icculus@10832
    56
        const __m128 divby2 = _mm_set1_ps(0.5f);
icculus@10832
    57
        while (i >= 4) {   /* 4 * float32 */
icculus@10832
    58
            _mm_store_ps(dst, _mm_mul_ps(_mm_hadd_ps(_mm_load_ps(src), _mm_load_ps(src+4)), divby2));
icculus@10832
    59
            i -= 4; src += 8; dst += 4;
icculus@10832
    60
        }
icculus@10832
    61
    }
icculus@10832
    62
icculus@10832
    63
    /* Finish off any leftovers with scalar operations. */
icculus@10832
    64
    while (i) {
icculus@10832
    65
        *dst = (src[0] + src[1]) * 0.5f;
icculus@10832
    66
        dst++; i--; src += 2;
icculus@10832
    67
    }
icculus@10832
    68
icculus@10832
    69
    cvt->len_cvt /= 2;
icculus@10832
    70
    if (cvt->filters[++cvt->filter_index]) {
icculus@10832
    71
        cvt->filters[cvt->filter_index] (cvt, format);
icculus@10832
    72
    }
icculus@10832
    73
}
icculus@10832
    74
#endif
icculus@10832
    75
icculus@11405
    76
/* Convert from stereo to mono. Average left and right. */
icculus@1982
    77
static void SDLCALL
icculus@10793
    78
SDL_ConvertStereoToMono(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@0
    79
{
icculus@10575
    80
    float *dst = (float *) cvt->buf;
icculus@10575
    81
    const float *src = dst;
slouken@1895
    82
    int i;
slouken@0
    83
icculus@10575
    84
    LOG_DEBUG_CONVERT("stereo", "mono");
icculus@10575
    85
    SDL_assert(format == AUDIO_F32SYS);
slouken@0
    86
icculus@10575
    87
    for (i = cvt->len_cvt / 8; i; --i, src += 2) {
icculus@10831
    88
        *(dst++) = (src[0] + src[1]) * 0.5f;
slouken@1895
    89
    }
icculus@1982
    90
slouken@1895
    91
    cvt->len_cvt /= 2;
slouken@1895
    92
    if (cvt->filters[++cvt->filter_index]) {
slouken@1895
    93
        cvt->filters[cvt->filter_index] (cvt, format);
slouken@1895
    94
    }
slouken@0
    95
}
slouken@0
    96
icculus@1982
    97
icculus@11405
    98
/* Convert from 5.1 to stereo. Average left and right, distribute center, discard LFE. */
icculus@1982
    99
static void SDLCALL
icculus@10793
   100
SDL_Convert51ToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@942
   101
{
icculus@10575
   102
    float *dst = (float *) cvt->buf;
icculus@10575
   103
    const float *src = dst;
slouken@1895
   104
    int i;
slouken@942
   105
icculus@10793
   106
    LOG_DEBUG_CONVERT("5.1", "stereo");
icculus@10575
   107
    SDL_assert(format == AUDIO_F32SYS);
slouken@942
   108
icculus@11405
   109
    /* SDL's 5.1 layout: FL+FR+FC+LFE+BL+BR */
icculus@10575
   110
    for (i = cvt->len_cvt / (sizeof (float) * 6); i; --i, src += 6, dst += 2) {
icculus@11405
   111
        const float front_center_distributed = src[2] * 0.5f;
icculus@11405
   112
        dst[0] = (src[0] + front_center_distributed + src[4]) / 2.5f;  /* left */
icculus@11405
   113
        dst[1] = (src[1] + front_center_distributed + src[5]) / 2.5f;  /* right */
icculus@1982
   114
    }
slouken@942
   115
slouken@1895
   116
    cvt->len_cvt /= 3;
slouken@1895
   117
    if (cvt->filters[++cvt->filter_index]) {
slouken@1895
   118
        cvt->filters[cvt->filter_index] (cvt, format);
slouken@1895
   119
    }
slouken@942
   120
}
slouken@942
   121
slouken@942
   122
icculus@11405
   123
/* Convert from quad to stereo. Average left and right. */
icculus@11405
   124
static void SDLCALL
icculus@11405
   125
SDL_ConvertQuadToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
icculus@11405
   126
{
icculus@11405
   127
    float *dst = (float *) cvt->buf;
icculus@11405
   128
    const float *src = dst;
icculus@11405
   129
    int i;
icculus@11405
   130
icculus@11405
   131
    LOG_DEBUG_CONVERT("quad", "stereo");
icculus@11405
   132
    SDL_assert(format == AUDIO_F32SYS);
icculus@11405
   133
icculus@11405
   134
    for (i = cvt->len_cvt / (sizeof (float) * 4); i; --i, src += 4, dst += 2) {
icculus@11405
   135
        dst[0] = (src[0] + src[2]) * 0.5f; /* left */
icculus@11405
   136
        dst[1] = (src[1] + src[3]) * 0.5f; /* right */
icculus@11405
   137
    }
icculus@11405
   138
icculus@11405
   139
    cvt->len_cvt /= 3;
icculus@11405
   140
    if (cvt->filters[++cvt->filter_index]) {
icculus@11405
   141
        cvt->filters[cvt->filter_index] (cvt, format);
icculus@11405
   142
    }
icculus@11405
   143
}
icculus@11405
   144
icculus@11405
   145
icculus@11405
   146
/* Convert from 7.1 to 5.1. Distribute sides across front and back. */
icculus@11405
   147
static void SDLCALL
icculus@11405
   148
SDL_Convert71To51(SDL_AudioCVT * cvt, SDL_AudioFormat format)
icculus@11405
   149
{
icculus@11405
   150
    float *dst = (float *) cvt->buf;
icculus@11405
   151
    const float *src = dst;
icculus@11405
   152
    int i;
icculus@11405
   153
icculus@11405
   154
    LOG_DEBUG_CONVERT("7.1", "5.1");
icculus@11405
   155
    SDL_assert(format == AUDIO_F32SYS);
icculus@11405
   156
icculus@11405
   157
    for (i = cvt->len_cvt / (sizeof (float) * 8); i; --i, src += 8, dst += 6) {
icculus@11405
   158
        const float surround_left_distributed = src[6] * 0.5f;
icculus@11405
   159
        const float surround_right_distributed = src[7] * 0.5f;
icculus@11405
   160
        dst[0] = (src[0] + surround_left_distributed) / 1.5f;  /* FL */
icculus@11405
   161
        dst[1] = (src[1] + surround_right_distributed) / 1.5f;  /* FR */
icculus@11405
   162
        dst[2] = src[2] / 1.5f; /* CC */
icculus@11405
   163
        dst[3] = src[3] / 1.5f; /* LFE */
icculus@11405
   164
        dst[4] = (src[4] + surround_left_distributed) / 1.5f;  /* BL */
icculus@11405
   165
        dst[5] = (src[5] + surround_right_distributed) / 1.5f;  /* BR */
icculus@11405
   166
    }
icculus@11405
   167
icculus@11405
   168
    cvt->len_cvt /= 8;
icculus@11405
   169
    cvt->len_cvt *= 6;
icculus@11405
   170
    if (cvt->filters[++cvt->filter_index]) {
icculus@11405
   171
        cvt->filters[cvt->filter_index] (cvt, format);
icculus@11405
   172
    }
icculus@11405
   173
}
icculus@11405
   174
icculus@11405
   175
icculus@11405
   176
/* Convert from 5.1 to quad. Distribute center across front, discard LFE. */
icculus@1982
   177
static void SDLCALL
icculus@10793
   178
SDL_Convert51ToQuad(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@942
   179
{
icculus@10575
   180
    float *dst = (float *) cvt->buf;
icculus@10575
   181
    const float *src = dst;
slouken@1895
   182
    int i;
slouken@942
   183
icculus@10793
   184
    LOG_DEBUG_CONVERT("5.1", "quad");
icculus@10575
   185
    SDL_assert(format == AUDIO_F32SYS);
slouken@942
   186
icculus@11405
   187
    /* SDL's 4.0 layout: FL+FR+BL+BR */
icculus@11405
   188
    /* SDL's 5.1 layout: FL+FR+FC+LFE+BL+BR */
icculus@10575
   189
    for (i = cvt->len_cvt / (sizeof (float) * 6); i; --i, src += 6, dst += 4) {
icculus@11405
   190
        const float front_center_distributed = src[2] * 0.5f;
icculus@11405
   191
        dst[0] = (src[0] + front_center_distributed) / 1.5f;  /* FL */
icculus@11405
   192
        dst[1] = (src[1] + front_center_distributed) / 1.5f;  /* FR */
icculus@11405
   193
        dst[2] = src[4] / 1.5f;  /* BL */
icculus@11405
   194
        dst[3] = src[5] / 1.5f;  /* BR */
icculus@1982
   195
    }
slouken@942
   196
icculus@1982
   197
    cvt->len_cvt /= 6;
icculus@1982
   198
    cvt->len_cvt *= 4;
slouken@1895
   199
    if (cvt->filters[++cvt->filter_index]) {
slouken@1895
   200
        cvt->filters[cvt->filter_index] (cvt, format);
slouken@1895
   201
    }
slouken@942
   202
}
slouken@0
   203
icculus@10793
   204
icculus@11405
   205
/* Upmix mono to stereo (by duplication) */
icculus@1982
   206
static void SDLCALL
icculus@10793
   207
SDL_ConvertMonoToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@0
   208
{
icculus@10575
   209
    const float *src = (const float *) (cvt->buf + cvt->len_cvt);
icculus@10575
   210
    float *dst = (float *) (cvt->buf + cvt->len_cvt * 2);
slouken@1895
   211
    int i;
slouken@0
   212
icculus@10575
   213
    LOG_DEBUG_CONVERT("mono", "stereo");
icculus@10575
   214
    SDL_assert(format == AUDIO_F32SYS);
slouken@0
   215
icculus@10575
   216
    for (i = cvt->len_cvt / sizeof (float); i; --i) {
icculus@10575
   217
        src--;
icculus@10575
   218
        dst -= 2;
icculus@10575
   219
        dst[0] = dst[1] = *src;
icculus@1982
   220
    }
slouken@0
   221
slouken@1895
   222
    cvt->len_cvt *= 2;
slouken@1895
   223
    if (cvt->filters[++cvt->filter_index]) {
slouken@1895
   224
        cvt->filters[cvt->filter_index] (cvt, format);
slouken@1895
   225
    }
slouken@0
   226
}
slouken@0
   227
slouken@942
   228
icculus@11405
   229
/* Upmix stereo to a pseudo-5.1 stream */
icculus@1982
   230
static void SDLCALL
icculus@10793
   231
SDL_ConvertStereoTo51(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@942
   232
{
slouken@1895
   233
    int i;
icculus@10575
   234
    float lf, rf, ce;
icculus@10575
   235
    const float *src = (const float *) (cvt->buf + cvt->len_cvt);
icculus@10575
   236
    float *dst = (float *) (cvt->buf + cvt->len_cvt * 3);
slouken@942
   237
icculus@10575
   238
    LOG_DEBUG_CONVERT("stereo", "5.1");
icculus@10575
   239
    SDL_assert(format == AUDIO_F32SYS);
slouken@942
   240
icculus@11405
   241
    for (i = cvt->len_cvt / (sizeof(float) * 2); i; --i) {
icculus@10575
   242
        dst -= 6;
icculus@10575
   243
        src -= 2;
icculus@10575
   244
        lf = src[0];
icculus@10575
   245
        rf = src[1];
icculus@10793
   246
        ce = (lf + rf) * 0.5f;
icculus@11405
   247
        /* !!! FIXME: FL and FR may clip */
icculus@10793
   248
        dst[0] = lf + (lf - ce);  /* FL */
icculus@10793
   249
        dst[1] = rf + (rf - ce);  /* FR */
icculus@10793
   250
        dst[2] = ce;  /* FC */
icculus@11405
   251
        dst[3] = 0;   /* LFE (only meant for special LFE effects) */
icculus@10793
   252
        dst[4] = lf;  /* BL */
icculus@10793
   253
        dst[5] = rf;  /* BR */
icculus@10575
   254
    }
slouken@942
   255
slouken@1895
   256
    cvt->len_cvt *= 3;
slouken@1895
   257
    if (cvt->filters[++cvt->filter_index]) {
slouken@1895
   258
        cvt->filters[cvt->filter_index] (cvt, format);
slouken@1895
   259
    }
slouken@942
   260
}
slouken@942
   261
slouken@942
   262
icculus@11405
   263
/* Upmix quad to a pseudo-5.1 stream */
icculus@11405
   264
static void SDLCALL
icculus@11405
   265
SDL_ConvertQuadTo51(SDL_AudioCVT * cvt, SDL_AudioFormat format)
icculus@11405
   266
{
icculus@11405
   267
    int i;
icculus@11405
   268
    float lf, rf, lb, rb, ce;
icculus@11405
   269
    const float *src = (const float *) (cvt->buf + cvt->len_cvt);
icculus@11405
   270
    float *dst = (float *) (cvt->buf + cvt->len_cvt * 3 / 2);
icculus@11405
   271
icculus@11405
   272
    LOG_DEBUG_CONVERT("quad", "5.1");
icculus@11405
   273
    SDL_assert(format == AUDIO_F32SYS);
icculus@11405
   274
    SDL_assert(cvt->len_cvt % (sizeof(float) * 4) == 0);
icculus@11405
   275
icculus@11405
   276
    for (i = cvt->len_cvt / (sizeof(float) * 4); i; --i) {
icculus@11405
   277
        dst -= 6;
icculus@11405
   278
        src -= 4;
icculus@11405
   279
        lf = src[0];
icculus@11405
   280
        rf = src[1];
icculus@11405
   281
        lb = src[2];
icculus@11405
   282
        rb = src[3];
icculus@11405
   283
        ce = (lf + rf) * 0.5f;
icculus@11405
   284
        /* !!! FIXME: FL and FR may clip */
icculus@11405
   285
        dst[0] = lf + (lf - ce);  /* FL */
icculus@11405
   286
        dst[1] = rf + (rf - ce);  /* FR */
icculus@11405
   287
        dst[2] = ce;  /* FC */
icculus@11405
   288
        dst[3] = 0;   /* LFE (only meant for special LFE effects) */
icculus@11405
   289
        dst[4] = lb;  /* BL */
icculus@11405
   290
        dst[5] = rb;  /* BR */
icculus@11405
   291
    }
icculus@11405
   292
icculus@11405
   293
    cvt->len_cvt = cvt->len_cvt * 3 / 2;
icculus@11405
   294
    if (cvt->filters[++cvt->filter_index]) {
icculus@11405
   295
        cvt->filters[cvt->filter_index] (cvt, format);
icculus@11405
   296
    }
icculus@11405
   297
}
icculus@11405
   298
icculus@11405
   299
icculus@11405
   300
/* Upmix stereo to a pseudo-4.0 stream (by duplication) */
icculus@1982
   301
static void SDLCALL
icculus@10793
   302
SDL_ConvertStereoToQuad(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@942
   303
{
icculus@10575
   304
    const float *src = (const float *) (cvt->buf + cvt->len_cvt);
icculus@10575
   305
    float *dst = (float *) (cvt->buf + cvt->len_cvt * 2);
icculus@10793
   306
    float lf, rf;
slouken@1895
   307
    int i;
slouken@942
   308
icculus@10575
   309
    LOG_DEBUG_CONVERT("stereo", "quad");
icculus@10575
   310
    SDL_assert(format == AUDIO_F32SYS);
slouken@942
   311
icculus@11405
   312
    for (i = cvt->len_cvt / (sizeof(float) * 2); i; --i) {
icculus@10575
   313
        dst -= 4;
icculus@10575
   314
        src -= 2;
icculus@10575
   315
        lf = src[0];
icculus@10575
   316
        rf = src[1];
icculus@10793
   317
        dst[0] = lf;  /* FL */
icculus@10793
   318
        dst[1] = rf;  /* FR */
icculus@10793
   319
        dst[2] = lf;  /* BL */
icculus@10793
   320
        dst[3] = rf;  /* BR */
icculus@10575
   321
    }
slouken@942
   322
slouken@1895
   323
    cvt->len_cvt *= 2;
slouken@1895
   324
    if (cvt->filters[++cvt->filter_index]) {
slouken@1895
   325
        cvt->filters[cvt->filter_index] (cvt, format);
slouken@1895
   326
    }
slouken@0
   327
}
slouken@0
   328
icculus@11405
   329
icculus@11405
   330
/* Upmix 5.1 to 7.1 */
icculus@11405
   331
static void SDLCALL
icculus@11405
   332
SDL_Convert51To71(SDL_AudioCVT * cvt, SDL_AudioFormat format)
icculus@11405
   333
{
icculus@11405
   334
    float lf, rf, lb, rb, ls, rs;
icculus@11405
   335
    int i;
icculus@11405
   336
    const float *src = (const float *) (cvt->buf + cvt->len_cvt);
icculus@11405
   337
    float *dst = (float *) (cvt->buf + cvt->len_cvt * 4 / 3);
icculus@11405
   338
icculus@11405
   339
    LOG_DEBUG_CONVERT("5.1", "7.1");
icculus@11405
   340
    SDL_assert(format == AUDIO_F32SYS);
icculus@11405
   341
    SDL_assert(cvt->len_cvt % (sizeof(float) * 6) == 0);
icculus@11405
   342
icculus@11405
   343
    for (i = cvt->len_cvt / (sizeof(float) * 6); i; --i) {
icculus@11405
   344
        dst -= 8;
icculus@11405
   345
        src -= 6;
icculus@11405
   346
        lf = src[0];
icculus@11405
   347
        rf = src[1];
icculus@11405
   348
        lb = src[4];
icculus@11405
   349
        rb = src[5];
icculus@11405
   350
        ls = (lf + lb) * 0.5f;
icculus@11405
   351
        rs = (rf + rb) * 0.5f;
icculus@11405
   352
        /* !!! FIXME: these four may clip */
icculus@11405
   353
        lf += lf - ls;
icculus@11405
   354
        rf += rf - ls;
icculus@11405
   355
        lb += lb - ls;
icculus@11405
   356
        rb += rb - ls;
icculus@11405
   357
        dst[3] = src[3];  /* LFE */
icculus@11405
   358
        dst[2] = src[2];  /* FC */
icculus@11405
   359
        dst[7] = rs; /* SR */
icculus@11405
   360
        dst[6] = ls; /* SL */
icculus@11405
   361
        dst[5] = rb;  /* BR */
icculus@11405
   362
        dst[4] = lb;  /* BL */
icculus@11405
   363
        dst[1] = rf;  /* FR */
icculus@11405
   364
        dst[0] = lf;  /* FL */
icculus@11405
   365
    }
icculus@11405
   366
icculus@11405
   367
    cvt->len_cvt = cvt->len_cvt * 4 / 3;
icculus@11405
   368
icculus@11405
   369
    if (cvt->filters[++cvt->filter_index]) {
icculus@11405
   370
        cvt->filters[cvt->filter_index] (cvt, format);
icculus@11405
   371
    }
icculus@11405
   372
}
icculus@11405
   373
icculus@11508
   374
/* SDL's resampler uses a "bandlimited interpolation" algorithm:
icculus@11508
   375
     https://ccrma.stanford.edu/~jos/resample/ */
icculus@11508
   376
icculus@11508
   377
#define RESAMPLER_ZERO_CROSSINGS 5
icculus@11508
   378
#define RESAMPLER_BITS_PER_SAMPLE 16
icculus@11508
   379
#define RESAMPLER_SAMPLES_PER_ZERO_CROSSING  (1 << ((RESAMPLER_BITS_PER_SAMPLE / 2) + 1))
icculus@11508
   380
#define RESAMPLER_FILTER_SIZE ((RESAMPLER_SAMPLES_PER_ZERO_CROSSING * RESAMPLER_ZERO_CROSSINGS) + 1)
icculus@11508
   381
icculus@11508
   382
/* This is a "modified" bessel function, so you can't use POSIX j0() */
icculus@11508
   383
static double
icculus@11508
   384
bessel(const double x)
icculus@11508
   385
{
icculus@11508
   386
    const double xdiv2 = x / 2.0;
icculus@11508
   387
    double i0 = 1.0f;
icculus@11508
   388
    double f = 1.0f;
icculus@11508
   389
    int i = 1;
icculus@11508
   390
icculus@11508
   391
    while (SDL_TRUE) {
icculus@11508
   392
        const double diff = SDL_pow(xdiv2, i * 2) / SDL_pow(f, 2);
icculus@11508
   393
        if (diff < 1.0e-21f) {
icculus@11508
   394
            break;
icculus@11508
   395
        }
icculus@11508
   396
        i0 += diff;
icculus@11508
   397
        i++;
icculus@11508
   398
        f *= (double) i;
icculus@11508
   399
    }
icculus@11508
   400
icculus@11508
   401
    return i0;
icculus@11508
   402
}
icculus@11508
   403
icculus@11508
   404
/* build kaiser table with cardinal sine applied to it, and array of differences between elements. */
icculus@11508
   405
static void
icculus@11508
   406
kaiser_and_sinc(float *table, float *diffs, const int tablelen, const double beta)
icculus@11508
   407
{
icculus@11508
   408
    const int lenm1 = tablelen - 1;
icculus@11508
   409
    const int lenm1div2 = lenm1 / 2;
icculus@11508
   410
    int i;
icculus@11508
   411
icculus@11508
   412
    table[0] = 1.0f;
icculus@11508
   413
    for (i = 1; i < tablelen; i++) {
icculus@11508
   414
        const double kaiser = bessel(beta * SDL_sqrt(1.0 - SDL_pow(((i - lenm1) / 2.0) / lenm1div2, 2.0))) / bessel(beta);
icculus@11508
   415
        table[tablelen - i] = (float) kaiser;
icculus@11508
   416
    }
icculus@11508
   417
icculus@11508
   418
    for (i = 1; i < tablelen; i++) {
icculus@11508
   419
        const float x = (((float) i) / ((float) RESAMPLER_SAMPLES_PER_ZERO_CROSSING)) * ((float) M_PI);
icculus@11508
   420
        table[i] *= SDL_sinf(x) / x;
icculus@11508
   421
        diffs[i - 1] = table[i] - table[i - 1];
icculus@11508
   422
    }
icculus@11508
   423
    diffs[lenm1] = 0.0f;
icculus@11508
   424
}
icculus@11508
   425
icculus@11508
   426
icculus@11508
   427
static SDL_SpinLock ResampleFilterSpinlock = 0;
icculus@11508
   428
static float *ResamplerFilter = NULL;
icculus@11508
   429
static float *ResamplerFilterDifference = NULL;
icculus@11508
   430
icculus@11508
   431
int
icculus@11508
   432
SDL_PrepareResampleFilter(void)
icculus@11508
   433
{
icculus@11508
   434
    SDL_AtomicLock(&ResampleFilterSpinlock);
icculus@11508
   435
    if (!ResamplerFilter) {
icculus@11508
   436
        /* if dB > 50, beta=(0.1102 * (dB - 8.7)), according to Matlab. */
icculus@11508
   437
        const double dB = 80.0;
icculus@11508
   438
        const double beta = 0.1102 * (dB - 8.7);
icculus@11508
   439
        const size_t alloclen = RESAMPLER_FILTER_SIZE * sizeof (float);
icculus@11508
   440
icculus@11508
   441
        ResamplerFilter = (float *) SDL_malloc(alloclen);
icculus@11508
   442
        if (!ResamplerFilter) {
icculus@11508
   443
            SDL_AtomicUnlock(&ResampleFilterSpinlock);
icculus@11508
   444
            return SDL_OutOfMemory();
icculus@11508
   445
        }
icculus@11508
   446
icculus@11508
   447
        ResamplerFilterDifference = (float *) SDL_malloc(alloclen);
icculus@11508
   448
        if (!ResamplerFilterDifference) {
icculus@11508
   449
            SDL_free(ResamplerFilter);
icculus@11508
   450
            ResamplerFilter = NULL;
icculus@11508
   451
            SDL_AtomicUnlock(&ResampleFilterSpinlock);
icculus@11508
   452
            return SDL_OutOfMemory();
icculus@11508
   453
        }
icculus@11508
   454
        kaiser_and_sinc(ResamplerFilter, ResamplerFilterDifference, RESAMPLER_FILTER_SIZE, beta);
icculus@11508
   455
    }
icculus@11508
   456
    SDL_AtomicUnlock(&ResampleFilterSpinlock);
icculus@11508
   457
    return 0;
icculus@11508
   458
}
icculus@11508
   459
icculus@11508
   460
void
icculus@11508
   461
SDL_FreeResampleFilter(void)
icculus@11508
   462
{
icculus@11508
   463
    SDL_free(ResamplerFilter);
icculus@11508
   464
    SDL_free(ResamplerFilterDifference);
icculus@11508
   465
    ResamplerFilter = NULL;
icculus@11508
   466
    ResamplerFilterDifference = NULL;
icculus@11508
   467
}
icculus@11508
   468
icculus@11517
   469
static int
icculus@11517
   470
ResamplerPadding(const int inrate, const int outrate)
icculus@11517
   471
{
icculus@11583
   472
    if (inrate == outrate) {
icculus@11583
   473
        return 0;
icculus@11583
   474
    } else if (inrate > outrate) {
icculus@11583
   475
        return (int) SDL_ceil(((float) (RESAMPLER_SAMPLES_PER_ZERO_CROSSING * inrate) / ((float) outrate)));
icculus@11583
   476
    }
icculus@11583
   477
    return RESAMPLER_SAMPLES_PER_ZERO_CROSSING;
icculus@11517
   478
}
icculus@11405
   479
icculus@11517
   480
/* lpadding and rpadding are expected to be buffers of (ResamplePadding(inrate, outrate) * chans * sizeof (float)) bytes. */
icculus@10799
   481
static int
icculus@11508
   482
SDL_ResampleAudio(const int chans, const int inrate, const int outrate,
icculus@11583
   483
                        const float *lpadding, const float *rpadding,
icculus@11583
   484
                        const float *inbuf, const int inbuflen,
icculus@11583
   485
                        float *outbuf, const int outbuflen)
icculus@10799
   486
{
icculus@11508
   487
    const float outtimeincr = 1.0f / ((float) outrate);
icculus@11508
   488
    const float ratio = ((float) outrate) / ((float) inrate);
icculus@11517
   489
    const int paddinglen = ResamplerPadding(inrate, outrate);
icculus@10817
   490
    const int framelen = chans * (int)sizeof (float);
icculus@11508
   491
    const int inframes = inbuflen / framelen;
icculus@11508
   492
    const int wantedoutframes = (int) ((inbuflen / framelen) * ratio);  /* outbuflen isn't total to write, it's total available. */
icculus@11508
   493
    const int maxoutframes = outbuflen / framelen;
icculus@11583
   494
    const int outframes = SDL_min(wantedoutframes, maxoutframes);
icculus@11508
   495
    float *dst = outbuf;
icculus@11508
   496
    float outtime = 0.0f;
icculus@11508
   497
    int i, j, chan;
icculus@10799
   498
icculus@11508
   499
    for (i = 0; i < outframes; i++) {
icculus@11508
   500
        const int srcindex = (int) (outtime * inrate);
icculus@11508
   501
        const float finrate = (float) inrate;
icculus@11508
   502
        const float intime = ((float) srcindex) / finrate;
icculus@11508
   503
        const float innexttime = ((float) (srcindex + 1)) / finrate;
icculus@10799
   504
icculus@11508
   505
        const float interpolation1 = 1.0f - (innexttime - outtime) / (innexttime - intime);
icculus@11508
   506
        const int filterindex1 = (int) (interpolation1 * RESAMPLER_SAMPLES_PER_ZERO_CROSSING);
icculus@11508
   507
        const float interpolation2 = 1.0f - interpolation1;
icculus@11541
   508
        const int filterindex2 = (int) (interpolation2 * RESAMPLER_SAMPLES_PER_ZERO_CROSSING);
icculus@10833
   509
icculus@11508
   510
        for (chan = 0; chan < chans; chan++) {
icculus@11508
   511
            float outsample = 0.0f;
icculus@11508
   512
icculus@11508
   513
            /* do this twice to calculate the sample, once for the "left wing" and then same for the right. */
icculus@11508
   514
            /* !!! FIXME: do both wings in one loop */
icculus@11508
   515
            for (j = 0; (filterindex1 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)) < RESAMPLER_FILTER_SIZE; j++) {
icculus@11508
   516
                const int srcframe = srcindex - j;
icculus@11517
   517
                /* !!! FIXME: we can bubble this conditional out of here by doing a pre loop. */
icculus@11517
   518
                const float insample = (srcframe < 0) ? lpadding[((paddinglen + srcframe) * chans) + chan] : inbuf[(srcframe * chans) + chan];
icculus@11508
   519
                outsample += (insample * (ResamplerFilter[filterindex1 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)] + (interpolation1 * ResamplerFilterDifference[filterindex1 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)])));
icculus@10840
   520
            }
icculus@11508
   521
icculus@11508
   522
            for (j = 0; (filterindex2 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)) < RESAMPLER_FILTER_SIZE; j++) {
icculus@11508
   523
                const int srcframe = srcindex + 1 + j;
icculus@11517
   524
                /* !!! FIXME: we can bubble this conditional out of here by doing a post loop. */
icculus@11517
   525
                const float insample = (srcframe >= inframes) ? rpadding[((srcframe - inframes) * chans) + chan] : inbuf[(srcframe * chans) + chan];
icculus@11508
   526
                outsample += (insample * (ResamplerFilter[filterindex2 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)] + (interpolation2 * ResamplerFilterDifference[filterindex2 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)])));
icculus@10840
   527
            }
icculus@11508
   528
            *(dst++) = outsample;
icculus@10799
   529
        }
icculus@10833
   530
icculus@11508
   531
        outtime += outtimeincr;
icculus@10799
   532
    }
icculus@10799
   533
icculus@11508
   534
    return outframes * chans * sizeof (float);
icculus@10799
   535
}
icculus@10799
   536
slouken@1895
   537
int
slouken@1895
   538
SDL_ConvertAudio(SDL_AudioCVT * cvt)
slouken@0
   539
{
icculus@3021
   540
    /* !!! FIXME: (cvt) should be const; stack-copy it here. */
icculus@3021
   541
    /* !!! FIXME: (actually, we can't...len_cvt needs to be updated. Grr.) */
icculus@3021
   542
slouken@1895
   543
    /* Make sure there's data to convert */
slouken@1895
   544
    if (cvt->buf == NULL) {
icculus@10575
   545
        return SDL_SetError("No buffer allocated for conversion");
slouken@1895
   546
    }
icculus@10575
   547
slouken@1895
   548
    /* Return okay if no conversion is necessary */
slouken@1895
   549
    cvt->len_cvt = cvt->len;
slouken@1895
   550
    if (cvt->filters[0] == NULL) {
icculus@10575
   551
        return 0;
slouken@1895
   552
    }
slouken@0
   553
slouken@1895
   554
    /* Set up the conversion and go! */
slouken@1895
   555
    cvt->filter_index = 0;
slouken@1895
   556
    cvt->filters[0] (cvt, cvt->src_format);
icculus@10575
   557
    return 0;
slouken@0
   558
}
slouken@0
   559
icculus@10575
   560
static void SDLCALL
icculus@10575
   561
SDL_Convert_Byteswap(SDL_AudioCVT *cvt, SDL_AudioFormat format)
icculus@10575
   562
{
slouken@10579
   563
#if DEBUG_CONVERT
slouken@10579
   564
    printf("Converting byte order\n");
slouken@10579
   565
#endif
icculus@1982
   566
icculus@10575
   567
    switch (SDL_AUDIO_BITSIZE(format)) {
icculus@10575
   568
        #define CASESWAP(b) \
icculus@10575
   569
            case b: { \
icculus@10575
   570
                Uint##b *ptr = (Uint##b *) cvt->buf; \
icculus@10575
   571
                int i; \
icculus@10575
   572
                for (i = cvt->len_cvt / sizeof (*ptr); i; --i, ++ptr) { \
icculus@10575
   573
                    *ptr = SDL_Swap##b(*ptr); \
icculus@10575
   574
                } \
icculus@10575
   575
                break; \
icculus@10575
   576
            }
icculus@1982
   577
icculus@10575
   578
        CASESWAP(16);
icculus@10575
   579
        CASESWAP(32);
icculus@10575
   580
        CASESWAP(64);
icculus@10575
   581
icculus@10575
   582
        #undef CASESWAP
icculus@10575
   583
icculus@10575
   584
        default: SDL_assert(!"unhandled byteswap datatype!"); break;
icculus@10575
   585
    }
icculus@10575
   586
icculus@10575
   587
    if (cvt->filters[++cvt->filter_index]) {
icculus@10575
   588
        /* flip endian flag for data. */
icculus@10575
   589
        if (format & SDL_AUDIO_MASK_ENDIAN) {
icculus@10575
   590
            format &= ~SDL_AUDIO_MASK_ENDIAN;
icculus@10575
   591
        } else {
icculus@10575
   592
            format |= SDL_AUDIO_MASK_ENDIAN;
icculus@10575
   593
        }
icculus@10575
   594
        cvt->filters[cvt->filter_index](cvt, format);
icculus@10575
   595
    }
icculus@1982
   596
}
icculus@1982
   597
slouken@11096
   598
static int
slouken@11096
   599
SDL_AddAudioCVTFilter(SDL_AudioCVT *cvt, const SDL_AudioFilter filter)
slouken@11096
   600
{
slouken@11096
   601
    if (cvt->filter_index >= SDL_AUDIOCVT_MAX_FILTERS) {
slouken@11096
   602
        return SDL_SetError("Too many filters needed for conversion, exceeded maximum of %d", SDL_AUDIOCVT_MAX_FILTERS);
slouken@11096
   603
    }
slouken@11096
   604
    if (filter == NULL) {
slouken@11096
   605
        return SDL_SetError("Audio filter pointer is NULL");
slouken@11096
   606
    }
slouken@11096
   607
    cvt->filters[cvt->filter_index++] = filter;
slouken@11096
   608
    cvt->filters[cvt->filter_index] = NULL; /* Moving terminator */
slouken@11096
   609
    return 0;
slouken@11096
   610
}
icculus@1982
   611
icculus@1982
   612
static int
icculus@10575
   613
SDL_BuildAudioTypeCVTToFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat src_fmt)
icculus@1982
   614
{
icculus@10575
   615
    int retval = 0;  /* 0 == no conversion necessary. */
icculus@1982
   616
icculus@10575
   617
    if ((SDL_AUDIO_ISBIGENDIAN(src_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
slouken@11096
   618
        if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
slouken@11096
   619
            return -1;
slouken@11096
   620
        }
icculus@10575
   621
        retval = 1;  /* added a converter. */
icculus@10575
   622
    }
icculus@1982
   623
icculus@10575
   624
    if (!SDL_AUDIO_ISFLOAT(src_fmt)) {
icculus@10576
   625
        const Uint16 src_bitsize = SDL_AUDIO_BITSIZE(src_fmt);
icculus@10576
   626
        const Uint16 dst_bitsize = 32;
icculus@10575
   627
        SDL_AudioFilter filter = NULL;
icculus@10576
   628
icculus@10575
   629
        switch (src_fmt & ~SDL_AUDIO_MASK_ENDIAN) {
icculus@10575
   630
            case AUDIO_S8: filter = SDL_Convert_S8_to_F32; break;
icculus@10575
   631
            case AUDIO_U8: filter = SDL_Convert_U8_to_F32; break;
icculus@10575
   632
            case AUDIO_S16: filter = SDL_Convert_S16_to_F32; break;
philipp@10591
   633
            case AUDIO_U16: filter = SDL_Convert_U16_to_F32; break;
icculus@10575
   634
            case AUDIO_S32: filter = SDL_Convert_S32_to_F32; break;
icculus@10575
   635
            default: SDL_assert(!"Unexpected audio format!"); break;
icculus@1982
   636
        }
icculus@1982
   637
icculus@10575
   638
        if (!filter) {
icculus@11319
   639
            return SDL_SetError("No conversion from source format to float available");
icculus@10575
   640
        }
icculus@10575
   641
slouken@11096
   642
        if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
slouken@11096
   643
            return -1;
slouken@11096
   644
        }
icculus@1982
   645
        if (src_bitsize < dst_bitsize) {
icculus@1982
   646
            const int mult = (dst_bitsize / src_bitsize);
icculus@1982
   647
            cvt->len_mult *= mult;
icculus@1982
   648
            cvt->len_ratio *= mult;
icculus@1982
   649
        } else if (src_bitsize > dst_bitsize) {
icculus@1982
   650
            cvt->len_ratio /= (src_bitsize / dst_bitsize);
icculus@1982
   651
        }
icculus@10576
   652
icculus@10575
   653
        retval = 1;  /* added a converter. */
icculus@1982
   654
    }
icculus@1982
   655
icculus@10575
   656
    return retval;
icculus@1982
   657
}
icculus@1982
   658
icculus@10575
   659
static int
icculus@10575
   660
SDL_BuildAudioTypeCVTFromFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat dst_fmt)
icculus@10575
   661
{
icculus@10575
   662
    int retval = 0;  /* 0 == no conversion necessary. */
icculus@3021
   663
icculus@10575
   664
    if (!SDL_AUDIO_ISFLOAT(dst_fmt)) {
icculus@10577
   665
        const Uint16 dst_bitsize = SDL_AUDIO_BITSIZE(dst_fmt);
icculus@10577
   666
        const Uint16 src_bitsize = 32;
icculus@10575
   667
        SDL_AudioFilter filter = NULL;
icculus@10575
   668
        switch (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN) {
icculus@10575
   669
            case AUDIO_S8: filter = SDL_Convert_F32_to_S8; break;
icculus@10575
   670
            case AUDIO_U8: filter = SDL_Convert_F32_to_U8; break;
icculus@10575
   671
            case AUDIO_S16: filter = SDL_Convert_F32_to_S16; break;
philipp@10591
   672
            case AUDIO_U16: filter = SDL_Convert_F32_to_U16; break;
icculus@10575
   673
            case AUDIO_S32: filter = SDL_Convert_F32_to_S32; break;
icculus@10575
   674
            default: SDL_assert(!"Unexpected audio format!"); break;
icculus@10575
   675
        }
slouken@2716
   676
icculus@10575
   677
        if (!filter) {
icculus@11319
   678
            return SDL_SetError("No conversion from float to destination format available");
icculus@10575
   679
        }
icculus@10575
   680
slouken@11096
   681
        if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
slouken@11096
   682
            return -1;
slouken@11096
   683
        }
icculus@10575
   684
        if (src_bitsize < dst_bitsize) {
icculus@10575
   685
            const int mult = (dst_bitsize / src_bitsize);
icculus@10575
   686
            cvt->len_mult *= mult;
icculus@10575
   687
            cvt->len_ratio *= mult;
icculus@10575
   688
        } else if (src_bitsize > dst_bitsize) {
icculus@10575
   689
            cvt->len_ratio /= (src_bitsize / dst_bitsize);
icculus@10575
   690
        }
icculus@10575
   691
        retval = 1;  /* added a converter. */
icculus@10575
   692
    }
icculus@10575
   693
icculus@10575
   694
    if ((SDL_AUDIO_ISBIGENDIAN(dst_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
slouken@11096
   695
        if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
slouken@11096
   696
            return -1;
slouken@11096
   697
        }
icculus@10575
   698
        retval = 1;  /* added a converter. */
icculus@10575
   699
    }
icculus@10575
   700
icculus@10575
   701
    return retval;
icculus@3021
   702
}
slouken@2716
   703
icculus@10799
   704
static void
icculus@10799
   705
SDL_ResampleCVT(SDL_AudioCVT *cvt, const int chans, const SDL_AudioFormat format)
icculus@10799
   706
{
icculus@11508
   707
    /* !!! FIXME in 2.1: there are ten slots in the filter list, and the theoretical maximum we use is six (seven with NULL terminator).
icculus@11508
   708
       !!! FIXME in 2.1:   We need to store data for this resampler, because the cvt structure doesn't store the original sample rates,
icculus@11508
   709
       !!! FIXME in 2.1:   so we steal the ninth and tenth slot.  :( */
icculus@11517
   710
    const int inrate = (int) (size_t) cvt->filters[SDL_AUDIOCVT_MAX_FILTERS-1];
icculus@11517
   711
    const int outrate = (int) (size_t) cvt->filters[SDL_AUDIOCVT_MAX_FILTERS];
icculus@10799
   712
    const float *src = (const float *) cvt->buf;
icculus@10799
   713
    const int srclen = cvt->len_cvt;
icculus@11508
   714
    /*float *dst = (float *) cvt->buf;
icculus@11508
   715
    const int dstlen = (cvt->len * cvt->len_mult);*/
icculus@11508
   716
    /* !!! FIXME: remove this if we can get the resampler to work in-place again. */
icculus@11508
   717
    float *dst = (float *) (cvt->buf + srclen);
icculus@11508
   718
    const int dstlen = (cvt->len * cvt->len_mult) - srclen;
icculus@11517
   719
    const int paddingsamples = (ResamplerPadding(inrate, outrate) * chans);
slouken@11519
   720
    float *padding;
icculus@10756
   721
icculus@10799
   722
    SDL_assert(format == AUDIO_F32SYS);
icculus@10799
   723
icculus@11517
   724
    /* we keep no streaming state here, so pad with silence on both ends. */
icculus@11585
   725
    padding = (float *) SDL_calloc(paddingsamples, sizeof (float));
slouken@11519
   726
    if (!padding) {
slouken@11519
   727
        SDL_OutOfMemory();
slouken@11519
   728
        return;
slouken@11519
   729
    }
icculus@10799
   730
icculus@11517
   731
    cvt->len_cvt = SDL_ResampleAudio(chans, inrate, outrate, padding, padding, src, srclen, dst, dstlen);
icculus@11508
   732
icculus@11585
   733
    SDL_free(padding);
slouken@11519
   734
icculus@11586
   735
    SDL_memmove(cvt->buf, dst, cvt->len_cvt);  /* !!! FIXME: remove this if we can get the resampler to work in-place again. */
icculus@11508
   736
icculus@10799
   737
    if (cvt->filters[++cvt->filter_index]) {
icculus@10799
   738
        cvt->filters[cvt->filter_index](cvt, format);
icculus@10799
   739
    }
icculus@10799
   740
}
icculus@10799
   741
icculus@10799
   742
/* !!! FIXME: We only have this macro salsa because SDL_AudioCVT doesn't
icculus@10799
   743
   !!! FIXME:  store channel info, so we have to have function entry
icculus@10799
   744
   !!! FIXME:  points for each supported channel count and multiple
icculus@10799
   745
   !!! FIXME:  vs arbitrary. When we rev the ABI, clean this up. */
icculus@10756
   746
#define RESAMPLER_FUNCS(chans) \
icculus@10756
   747
    static void SDLCALL \
icculus@10799
   748
    SDL_ResampleCVT_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
icculus@10799
   749
        SDL_ResampleCVT(cvt, chans, format); \
icculus@10756
   750
    }
icculus@10756
   751
RESAMPLER_FUNCS(1)
icculus@10756
   752
RESAMPLER_FUNCS(2)
icculus@10756
   753
RESAMPLER_FUNCS(4)
icculus@10756
   754
RESAMPLER_FUNCS(6)
icculus@10756
   755
RESAMPLER_FUNCS(8)
icculus@10756
   756
#undef RESAMPLER_FUNCS
icculus@10756
   757
icculus@10799
   758
static SDL_AudioFilter
icculus@10799
   759
ChooseCVTResampler(const int dst_channels)
icculus@3021
   760
{
icculus@10799
   761
    switch (dst_channels) {
icculus@10799
   762
        case 1: return SDL_ResampleCVT_c1;
icculus@10799
   763
        case 2: return SDL_ResampleCVT_c2;
icculus@10799
   764
        case 4: return SDL_ResampleCVT_c4;
icculus@10799
   765
        case 6: return SDL_ResampleCVT_c6;
icculus@10799
   766
        case 8: return SDL_ResampleCVT_c8;
icculus@10799
   767
        default: break;
icculus@3021
   768
    }
slouken@2716
   769
icculus@10799
   770
    return NULL;
icculus@10756
   771
}
icculus@10575
   772
icculus@3021
   773
static int
icculus@10756
   774
SDL_BuildAudioResampleCVT(SDL_AudioCVT * cvt, const int dst_channels,
icculus@10756
   775
                          const int src_rate, const int dst_rate)
icculus@3021
   776
{
icculus@10756
   777
    SDL_AudioFilter filter;
icculus@3021
   778
icculus@10756
   779
    if (src_rate == dst_rate) {
icculus@10756
   780
        return 0;  /* no conversion necessary. */
slouken@2716
   781
    }
slouken@2716
   782
icculus@10799
   783
    filter = ChooseCVTResampler(dst_channels);
icculus@10756
   784
    if (filter == NULL) {
icculus@10756
   785
        return SDL_SetError("No conversion available for these rates");
icculus@10756
   786
    }
icculus@10756
   787
icculus@11508
   788
    if (SDL_PrepareResampleFilter() < 0) {
icculus@11508
   789
        return -1;
icculus@11508
   790
    }
icculus@11508
   791
icculus@10756
   792
    /* Update (cvt) with filter details... */
slouken@11096
   793
    if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
slouken@11096
   794
        return -1;
slouken@11096
   795
    }
icculus@11508
   796
icculus@11508
   797
    /* !!! FIXME in 2.1: there are ten slots in the filter list, and the theoretical maximum we use is six (seven with NULL terminator).
icculus@11508
   798
       !!! FIXME in 2.1:   We need to store data for this resampler, because the cvt structure doesn't store the original sample rates,
icculus@11508
   799
       !!! FIXME in 2.1:   so we steal the ninth and tenth slot.  :( */
icculus@11508
   800
    if (cvt->filter_index >= (SDL_AUDIOCVT_MAX_FILTERS-2)) {
icculus@11508
   801
        return SDL_SetError("Too many filters needed for conversion, exceeded maximum of %d", SDL_AUDIOCVT_MAX_FILTERS-2);
icculus@11508
   802
    }
icculus@11508
   803
    cvt->filters[SDL_AUDIOCVT_MAX_FILTERS-1] = (SDL_AudioFilter) (size_t) src_rate;
icculus@11508
   804
    cvt->filters[SDL_AUDIOCVT_MAX_FILTERS] = (SDL_AudioFilter) (size_t) dst_rate;
icculus@11508
   805
icculus@10756
   806
    if (src_rate < dst_rate) {
icculus@10756
   807
        const double mult = ((double) dst_rate) / ((double) src_rate);
icculus@10756
   808
        cvt->len_mult *= (int) SDL_ceil(mult);
icculus@10756
   809
        cvt->len_ratio *= mult;
icculus@10756
   810
    } else {
icculus@10756
   811
        cvt->len_ratio /= ((double) src_rate) / ((double) dst_rate);
icculus@10756
   812
    }
icculus@10756
   813
icculus@11508
   814
    /* !!! FIXME: remove this if we can get the resampler to work in-place again. */
icculus@11508
   815
    /* the buffer is big enough to hold the destination now, but
icculus@11508
   816
       we need it large enough to hold a separate scratch buffer. */
icculus@11508
   817
    cvt->len_mult *= 2;
icculus@11508
   818
icculus@10756
   819
    return 1;               /* added a converter. */
slouken@2716
   820
}
icculus@1982
   821
icculus@11097
   822
static SDL_bool
icculus@11097
   823
SDL_SupportedAudioFormat(const SDL_AudioFormat fmt)
icculus@11097
   824
{
icculus@11097
   825
    switch (fmt) {
icculus@11097
   826
        case AUDIO_U8:
icculus@11097
   827
        case AUDIO_S8:
icculus@11097
   828
        case AUDIO_U16LSB:
icculus@11097
   829
        case AUDIO_S16LSB:
icculus@11097
   830
        case AUDIO_U16MSB:
icculus@11097
   831
        case AUDIO_S16MSB:
icculus@11097
   832
        case AUDIO_S32LSB:
icculus@11097
   833
        case AUDIO_S32MSB:
icculus@11097
   834
        case AUDIO_F32LSB:
icculus@11097
   835
        case AUDIO_F32MSB:
icculus@11097
   836
            return SDL_TRUE;  /* supported. */
icculus@11097
   837
icculus@11097
   838
        default:
icculus@11097
   839
            break;
icculus@11097
   840
    }
icculus@11097
   841
icculus@11097
   842
    return SDL_FALSE;  /* unsupported. */
icculus@11097
   843
}
icculus@11097
   844
icculus@11097
   845
static SDL_bool
icculus@11097
   846
SDL_SupportedChannelCount(const int channels)
icculus@11097
   847
{
icculus@11097
   848
    switch (channels) {
icculus@11097
   849
        case 1:  /* mono */
icculus@11097
   850
        case 2:  /* stereo */
icculus@11097
   851
        case 4:  /* quad */
icculus@11097
   852
        case 6:  /* 5.1 */
icculus@11405
   853
        case 8:  /* 7.1 */
icculus@11405
   854
          return SDL_TRUE;  /* supported. */
icculus@11097
   855
icculus@11097
   856
        default:
icculus@11097
   857
            break;
icculus@11097
   858
    }
icculus@11097
   859
icculus@11097
   860
    return SDL_FALSE;  /* unsupported. */
icculus@11097
   861
}
icculus@11097
   862
icculus@1982
   863
icculus@1982
   864
/* Creates a set of audio filters to convert from one format to another.
icculus@11319
   865
   Returns 0 if no conversion is needed, 1 if the audio filter is set up,
icculus@11319
   866
   or -1 if an error like invalid parameter, unsupported format, etc. occurred.
slouken@0
   867
*/
slouken@1895
   868
slouken@1895
   869
int
slouken@1895
   870
SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
icculus@1982
   871
                  SDL_AudioFormat src_fmt, Uint8 src_channels, int src_rate,
icculus@1982
   872
                  SDL_AudioFormat dst_fmt, Uint8 dst_channels, int dst_rate)
slouken@0
   873
{
aschiffler@6819
   874
    /* Sanity check target pointer */
aschiffler@6819
   875
    if (cvt == NULL) {
icculus@7037
   876
        return SDL_InvalidParamError("cvt");
aschiffler@6819
   877
    }
slouken@7191
   878
slouken@10767
   879
    /* Make sure we zero out the audio conversion before error checking */
slouken@10767
   880
    SDL_zerop(cvt);
slouken@10767
   881
icculus@11097
   882
    if (!SDL_SupportedAudioFormat(src_fmt)) {
icculus@7037
   883
        return SDL_SetError("Invalid source format");
icculus@11097
   884
    } else if (!SDL_SupportedAudioFormat(dst_fmt)) {
icculus@7037
   885
        return SDL_SetError("Invalid destination format");
icculus@11097
   886
    } else if (!SDL_SupportedChannelCount(src_channels)) {
icculus@11097
   887
        return SDL_SetError("Invalid source channels");
icculus@11097
   888
    } else if (!SDL_SupportedChannelCount(dst_channels)) {
icculus@11097
   889
        return SDL_SetError("Invalid destination channels");
icculus@11097
   890
    } else if (src_rate == 0) {
icculus@11097
   891
        return SDL_SetError("Source rate is zero");
icculus@11097
   892
    } else if (dst_rate == 0) {
icculus@11097
   893
        return SDL_SetError("Destination rate is zero");
icculus@1982
   894
    }
icculus@3021
   895
slouken@10579
   896
#if DEBUG_CONVERT
icculus@1982
   897
    printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
slouken@1985
   898
           src_fmt, dst_fmt, src_channels, dst_channels, src_rate, dst_rate);
slouken@1985
   899
#endif
icculus@1982
   900
slouken@1895
   901
    /* Start off with no conversion necessary */
icculus@1982
   902
    cvt->src_format = src_fmt;
icculus@1982
   903
    cvt->dst_format = dst_fmt;
slouken@1895
   904
    cvt->needed = 0;
slouken@1895
   905
    cvt->filter_index = 0;
icculus@11508
   906
    SDL_zero(cvt->filters);
slouken@1895
   907
    cvt->len_mult = 1;
slouken@1895
   908
    cvt->len_ratio = 1.0;
icculus@3021
   909
    cvt->rate_incr = ((double) dst_rate) / ((double) src_rate);
slouken@0
   910
slouken@11406
   911
    /* Make sure we've chosen audio conversion functions (MMX, scalar, etc.) */
slouken@11406
   912
    SDL_ChooseAudioConverters();
slouken@11406
   913
icculus@10575
   914
    /* Type conversion goes like this now:
icculus@10575
   915
        - byteswap to CPU native format first if necessary.
icculus@10575
   916
        - convert to native Float32 if necessary.
icculus@10575
   917
        - resample and change channel count if necessary.
icculus@10575
   918
        - convert back to native format.
icculus@10575
   919
        - byteswap back to foreign format if necessary.
icculus@10575
   920
icculus@10575
   921
       The expectation is we can process data faster in float32
icculus@10575
   922
       (possibly with SIMD), and making several passes over the same
icculus@10756
   923
       buffer is likely to be CPU cache-friendly, avoiding the
icculus@10575
   924
       biggest performance hit in modern times. Previously we had
icculus@10575
   925
       (script-generated) custom converters for every data type and
icculus@10575
   926
       it was a bloat on SDL compile times and final library size. */
icculus@10575
   927
slouken@10767
   928
    /* see if we can skip float conversion entirely. */
slouken@10767
   929
    if (src_rate == dst_rate && src_channels == dst_channels) {
slouken@10767
   930
        if (src_fmt == dst_fmt) {
slouken@10767
   931
            return 0;
slouken@10767
   932
        }
slouken@10767
   933
slouken@10767
   934
        /* just a byteswap needed? */
slouken@10767
   935
        if ((src_fmt & ~SDL_AUDIO_MASK_ENDIAN) == (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN)) {
slouken@11096
   936
            if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
slouken@11096
   937
                return -1;
slouken@11096
   938
            }
slouken@10767
   939
            cvt->needed = 1;
slouken@10767
   940
            return 1;
slouken@10767
   941
        }
icculus@10575
   942
    }
icculus@10575
   943
icculus@1982
   944
    /* Convert data types, if necessary. Updates (cvt). */
slouken@10767
   945
    if (SDL_BuildAudioTypeCVTToFloat(cvt, src_fmt) < 0) {
slouken@1985
   946
        return -1;              /* shouldn't happen, but just in case... */
icculus@3021
   947
    }
slouken@0
   948
icculus@1982
   949
    /* Channel conversion */
icculus@11405
   950
    if (src_channels < dst_channels) {
icculus@11405
   951
        /* Upmixing */
icculus@11405
   952
        /* Mono -> Stereo [-> ...] */
slouken@1895
   953
        if ((src_channels == 1) && (dst_channels > 1)) {
slouken@11096
   954
            if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertMonoToStereo) < 0) {
slouken@11096
   955
                return -1;
slouken@11096
   956
            }
slouken@1895
   957
            cvt->len_mult *= 2;
slouken@1895
   958
            src_channels = 2;
slouken@1895
   959
            cvt->len_ratio *= 2;
slouken@1895
   960
        }
icculus@11405
   961
        /* [Mono ->] Stereo -> 5.1 [-> 7.1] */
icculus@11405
   962
        if ((src_channels == 2) && (dst_channels >= 6)) {
slouken@11096
   963
            if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertStereoTo51) < 0) {
slouken@11096
   964
                return -1;
slouken@11096
   965
            }
slouken@1895
   966
            src_channels = 6;
slouken@1895
   967
            cvt->len_mult *= 3;
slouken@1895
   968
            cvt->len_ratio *= 3;
slouken@1895
   969
        }
icculus@11405
   970
        /* Quad -> 5.1 [-> 7.1] */
icculus@11405
   971
        if ((src_channels == 4) && (dst_channels >= 6)) {
icculus@11405
   972
            if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertQuadTo51) < 0) {
icculus@11405
   973
                return -1;
icculus@11405
   974
            }
icculus@11405
   975
            src_channels = 6;
icculus@11405
   976
            cvt->len_mult = (cvt->len_mult * 3 + 1) / 2;
icculus@11405
   977
            cvt->len_ratio *= 1.5;
icculus@11405
   978
        }
icculus@11405
   979
        /* [[Mono ->] Stereo ->] 5.1 -> 7.1 */
icculus@11405
   980
        if ((src_channels == 6) && (dst_channels == 8)) {
icculus@11405
   981
            if (SDL_AddAudioCVTFilter(cvt, SDL_Convert51To71) < 0) {
icculus@11405
   982
                return -1;
icculus@11405
   983
            }
icculus@11405
   984
            src_channels = 8;
icculus@11405
   985
            cvt->len_mult = (cvt->len_mult * 4 + 2) / 3;
icculus@11405
   986
            /* Should be numerically exact with every valid input to this
icculus@11405
   987
               function */
icculus@11405
   988
            cvt->len_ratio = cvt->len_ratio * 4 / 3;
icculus@11405
   989
        }
icculus@11405
   990
        /* [Mono ->] Stereo -> Quad */
slouken@1895
   991
        if ((src_channels == 2) && (dst_channels == 4)) {
slouken@11096
   992
            if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertStereoToQuad) < 0) {
slouken@11096
   993
                return -1;
slouken@11096
   994
            }
slouken@1895
   995
            src_channels = 4;
slouken@1895
   996
            cvt->len_mult *= 2;
slouken@1895
   997
            cvt->len_ratio *= 2;
slouken@1895
   998
        }
icculus@11405
   999
    } else if (src_channels > dst_channels) {
icculus@11405
  1000
        /* Downmixing */
icculus@11405
  1001
        /* 7.1 -> 5.1 [-> Stereo [-> Mono]] */
icculus@11405
  1002
        /* 7.1 -> 5.1 [-> Quad] */
icculus@11405
  1003
        if ((src_channels == 8) && (dst_channels <= 6)) {
icculus@11405
  1004
            if (SDL_AddAudioCVTFilter(cvt, SDL_Convert71To51) < 0) {
slouken@11096
  1005
                return -1;
slouken@11096
  1006
            }
icculus@11405
  1007
            src_channels = 6;
icculus@11405
  1008
            cvt->len_ratio *= 0.75;
slouken@1895
  1009
        }
icculus@11405
  1010
        /* [7.1 ->] 5.1 -> Stereo [-> Mono] */
slouken@1895
  1011
        if ((src_channels == 6) && (dst_channels <= 2)) {
slouken@11096
  1012
            if (SDL_AddAudioCVTFilter(cvt, SDL_Convert51ToStereo) < 0) {
slouken@11096
  1013
                return -1;
slouken@11096
  1014
            }
slouken@1895
  1015
            src_channels = 2;
slouken@1895
  1016
            cvt->len_ratio /= 3;
slouken@1895
  1017
        }
icculus@11405
  1018
        /* 5.1 -> Quad */
slouken@1895
  1019
        if ((src_channels == 6) && (dst_channels == 4)) {
slouken@11096
  1020
            if (SDL_AddAudioCVTFilter(cvt, SDL_Convert51ToQuad) < 0) {
slouken@11096
  1021
                return -1;
slouken@11096
  1022
            }
slouken@1895
  1023
            src_channels = 4;
icculus@11405
  1024
            cvt->len_ratio = cvt->len_ratio * 2 / 3;
icculus@11405
  1025
        }
icculus@11405
  1026
        /* Quad -> Stereo [-> Mono] */
icculus@11405
  1027
        if ((src_channels == 4) && (dst_channels <= 2)) {
icculus@11405
  1028
            if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertQuadToStereo) < 0) {
icculus@11405
  1029
                return -1;
icculus@11405
  1030
            }
icculus@11405
  1031
            src_channels = 2;
slouken@1895
  1032
            cvt->len_ratio /= 2;
slouken@1895
  1033
        }
icculus@11405
  1034
        /* [... ->] Stereo -> Mono */
icculus@11405
  1035
        if ((src_channels == 2) && (dst_channels == 1)) {
icculus@10832
  1036
            SDL_AudioFilter filter = NULL;
icculus@10832
  1037
icculus@10832
  1038
            #if HAVE_SSE3_INTRINSICS
icculus@10832
  1039
            if (SDL_HasSSE3()) {
icculus@10832
  1040
                filter = SDL_ConvertStereoToMono_SSE3;
icculus@10832
  1041
            }
icculus@10832
  1042
            #endif
icculus@10832
  1043
icculus@10832
  1044
            if (!filter) {
icculus@10832
  1045
                filter = SDL_ConvertStereoToMono;
icculus@10832
  1046
            }
icculus@10832
  1047
slouken@11096
  1048
            if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
slouken@11096
  1049
                return -1;
slouken@11096
  1050
            }
icculus@10832
  1051
icculus@11405
  1052
            src_channels = 1;
slouken@1895
  1053
            cvt->len_ratio /= 2;
slouken@1895
  1054
        }
slouken@1895
  1055
    }
slouken@0
  1056
icculus@11405
  1057
    if (src_channels != dst_channels) {
icculus@11405
  1058
        /* All combinations of supported channel counts should have been
icculus@11405
  1059
           handled by now, but let's be defensive */
icculus@11405
  1060
      return SDL_SetError("Invalid channel combination");
icculus@11405
  1061
    }
icculus@11405
  1062
    
icculus@3021
  1063
    /* Do rate conversion, if necessary. Updates (cvt). */
slouken@10767
  1064
    if (SDL_BuildAudioResampleCVT(cvt, dst_channels, src_rate, dst_rate) < 0) {
icculus@3021
  1065
        return -1;              /* shouldn't happen, but just in case... */
slouken@2716
  1066
    }
slouken@2716
  1067
icculus@10756
  1068
    /* Move to final data type. */
slouken@10767
  1069
    if (SDL_BuildAudioTypeCVTFromFloat(cvt, dst_fmt) < 0) {
icculus@10575
  1070
        return -1;              /* shouldn't happen, but just in case... */
slouken@1895
  1071
    }
icculus@10575
  1072
icculus@10575
  1073
    cvt->needed = (cvt->filter_index != 0);
slouken@1895
  1074
    return (cvt->needed);
slouken@0
  1075
}
slouken@1895
  1076
icculus@10842
  1077
typedef int (*SDL_ResampleAudioStreamFunc)(SDL_AudioStream *stream, const void *inbuf, const int inbuflen, void *outbuf, const int outbuflen);
slouken@10773
  1078
typedef void (*SDL_ResetAudioStreamResamplerFunc)(SDL_AudioStream *stream);
slouken@10773
  1079
typedef void (*SDL_CleanupAudioStreamResamplerFunc)(SDL_AudioStream *stream);
icculus@10757
  1080
icculus@10757
  1081
struct SDL_AudioStream
icculus@10757
  1082
{
icculus@10757
  1083
    SDL_AudioCVT cvt_before_resampling;
icculus@10757
  1084
    SDL_AudioCVT cvt_after_resampling;
icculus@10757
  1085
    SDL_DataQueue *queue;
icculus@11583
  1086
    SDL_bool first_run;
icculus@10844
  1087
    Uint8 *work_buffer_base;  /* maybe unaligned pointer from SDL_realloc(). */
icculus@10757
  1088
    int work_buffer_len;
icculus@10757
  1089
    int src_sample_frame_size;
icculus@10757
  1090
    SDL_AudioFormat src_format;
icculus@10757
  1091
    Uint8 src_channels;
icculus@10757
  1092
    int src_rate;
icculus@10757
  1093
    int dst_sample_frame_size;
icculus@10757
  1094
    SDL_AudioFormat dst_format;
icculus@10757
  1095
    Uint8 dst_channels;
icculus@10757
  1096
    int dst_rate;
icculus@10757
  1097
    double rate_incr;
icculus@10757
  1098
    Uint8 pre_resample_channels;
slouken@10773
  1099
    int packetlen;
icculus@11583
  1100
    int resampler_padding_samples;
icculus@11583
  1101
    float *resampler_padding;
slouken@10773
  1102
    void *resampler_state;
slouken@10773
  1103
    SDL_ResampleAudioStreamFunc resampler_func;
slouken@10773
  1104
    SDL_ResetAudioStreamResamplerFunc reset_resampler_func;
slouken@10773
  1105
    SDL_CleanupAudioStreamResamplerFunc cleanup_resampler_func;
slouken@10773
  1106
};
slouken@10773
  1107
icculus@10851
  1108
static Uint8 *
icculus@10851
  1109
EnsureStreamBufferSize(SDL_AudioStream *stream, const int newlen)
icculus@10851
  1110
{
icculus@10851
  1111
    Uint8 *ptr;
icculus@10851
  1112
    size_t offset;
icculus@10851
  1113
icculus@10851
  1114
    if (stream->work_buffer_len >= newlen) {
icculus@10851
  1115
        ptr = stream->work_buffer_base;
icculus@10851
  1116
    } else {
icculus@10851
  1117
        ptr = (Uint8 *) SDL_realloc(stream->work_buffer_base, newlen + 32);
icculus@10851
  1118
        if (!ptr) {
icculus@10851
  1119
            SDL_OutOfMemory();
icculus@10851
  1120
            return NULL;
icculus@10851
  1121
        }
icculus@10851
  1122
        /* Make sure we're aligned to 16 bytes for SIMD code. */
icculus@10851
  1123
        stream->work_buffer_base = ptr;
icculus@10851
  1124
        stream->work_buffer_len = newlen;
icculus@10851
  1125
    }
icculus@10851
  1126
icculus@10851
  1127
    offset = ((size_t) ptr) & 15;
icculus@10851
  1128
    return offset ? ptr + (16 - offset) : ptr;
icculus@10851
  1129
}
icculus@10851
  1130
slouken@10777
  1131
#ifdef HAVE_LIBSAMPLERATE_H
slouken@10773
  1132
static int
icculus@10842
  1133
SDL_ResampleAudioStream_SRC(SDL_AudioStream *stream, const void *_inbuf, const int inbuflen, void *_outbuf, const int outbuflen)
slouken@10773
  1134
{
icculus@10842
  1135
    const float *inbuf = (const float *) _inbuf;
icculus@10842
  1136
    float *outbuf = (float *) _outbuf;
icculus@10799
  1137
    const int framelen = sizeof(float) * stream->pre_resample_channels;
icculus@10790
  1138
    SRC_STATE *state = (SRC_STATE *)stream->resampler_state;
slouken@10773
  1139
    SRC_DATA data;
slouken@10773
  1140
    int result;
slouken@10773
  1141
icculus@11583
  1142
    SDL_assert(inbuf != ((const float *) outbuf));  /* SDL_AudioStreamPut() shouldn't allow in-place resamples. */
icculus@10851
  1143
slouken@10777
  1144
    data.data_in = (float *)inbuf; /* Older versions of libsamplerate had a non-const pointer, but didn't write to it */
icculus@10799
  1145
    data.input_frames = inbuflen / framelen;
slouken@10773
  1146
    data.input_frames_used = 0;
slouken@10773
  1147
slouken@10773
  1148
    data.data_out = outbuf;
icculus@10799
  1149
    data.output_frames = outbuflen / framelen;
slouken@10773
  1150
slouken@10773
  1151
    data.end_of_input = 0;
slouken@10773
  1152
    data.src_ratio = stream->rate_incr;
slouken@10773
  1153
icculus@10790
  1154
    result = SRC_src_process(state, &data);
slouken@10773
  1155
    if (result != 0) {
icculus@10790
  1156
        SDL_SetError("src_process() failed: %s", SRC_src_strerror(result));
slouken@10773
  1157
        return 0;
slouken@10773
  1158
    }
slouken@10773
  1159
slouken@10773
  1160
    /* If this fails, we need to store them off somewhere */
slouken@10773
  1161
    SDL_assert(data.input_frames_used == data.input_frames);
slouken@10773
  1162
slouken@10773
  1163
    return data.output_frames_gen * (sizeof(float) * stream->pre_resample_channels);
slouken@10773
  1164
}
slouken@10773
  1165
slouken@10773
  1166
static void
slouken@10773
  1167
SDL_ResetAudioStreamResampler_SRC(SDL_AudioStream *stream)
slouken@10773
  1168
{
icculus@10790
  1169
    SRC_src_reset((SRC_STATE *)stream->resampler_state);
slouken@10773
  1170
}
slouken@10773
  1171
slouken@10773
  1172
static void
slouken@10773
  1173
SDL_CleanupAudioStreamResampler_SRC(SDL_AudioStream *stream)
slouken@10773
  1174
{
icculus@10790
  1175
    SRC_STATE *state = (SRC_STATE *)stream->resampler_state;
slouken@10773
  1176
    if (state) {
icculus@10790
  1177
        SRC_src_delete(state);
slouken@10773
  1178
    }
slouken@10773
  1179
slouken@10773
  1180
    stream->resampler_state = NULL;
slouken@10773
  1181
    stream->resampler_func = NULL;
slouken@10773
  1182
    stream->reset_resampler_func = NULL;
slouken@10773
  1183
    stream->cleanup_resampler_func = NULL;
slouken@10773
  1184
}
slouken@10773
  1185
slouken@10773
  1186
static SDL_bool
slouken@10773
  1187
SetupLibSampleRateResampling(SDL_AudioStream *stream)
slouken@10773
  1188
{
icculus@10790
  1189
    int result = 0;
icculus@10790
  1190
    SRC_STATE *state = NULL;
slouken@10773
  1191
icculus@10790
  1192
    if (SRC_available) {
icculus@10849
  1193
        state = SRC_src_new(SRC_converter, stream->pre_resample_channels, &result);
icculus@10790
  1194
        if (!state) {
icculus@10790
  1195
            SDL_SetError("src_new() failed: %s", SRC_src_strerror(result));
icculus@10790
  1196
        }
slouken@10773
  1197
    }
slouken@10773
  1198
icculus@10790
  1199
    if (!state) {
icculus@10790
  1200
        SDL_CleanupAudioStreamResampler_SRC(stream);
slouken@10773
  1201
        return SDL_FALSE;
slouken@10773
  1202
    }
slouken@10773
  1203
slouken@10773
  1204
    stream->resampler_state = state;
slouken@10773
  1205
    stream->resampler_func = SDL_ResampleAudioStream_SRC;
slouken@10773
  1206
    stream->reset_resampler_func = SDL_ResetAudioStreamResampler_SRC;
slouken@10773
  1207
    stream->cleanup_resampler_func = SDL_CleanupAudioStreamResampler_SRC;
slouken@10773
  1208
slouken@10773
  1209
    return SDL_TRUE;
slouken@10773
  1210
}
icculus@10790
  1211
#endif /* HAVE_LIBSAMPLERATE_H */
slouken@10773
  1212
slouken@10773
  1213
slouken@10773
  1214
static int
icculus@10842
  1215
SDL_ResampleAudioStream(SDL_AudioStream *stream, const void *_inbuf, const int inbuflen, void *_outbuf, const int outbuflen)
slouken@10773
  1216
{
icculus@11583
  1217
    const Uint8 *inbufend = ((const Uint8 *) _inbuf) + inbuflen;
icculus@10842
  1218
    const float *inbuf = (const float *) _inbuf;
icculus@10842
  1219
    float *outbuf = (float *) _outbuf;
icculus@11517
  1220
    const int chans = (int) stream->pre_resample_channels;
icculus@11517
  1221
    const int inrate = stream->src_rate;
icculus@11517
  1222
    const int outrate = stream->dst_rate;
icculus@11583
  1223
    const int paddingsamples = stream->resampler_padding_samples;
icculus@11517
  1224
    const int paddingbytes = paddingsamples * sizeof (float);
icculus@11517
  1225
    float *lpadding = (float *) stream->resampler_state;
icculus@11583
  1226
    const float *rpadding = (const float *) inbufend; /* we set this up so there are valid padding samples at the end of the input buffer. */
icculus@11591
  1227
    const int cpy = SDL_min(inbuflen, paddingbytes);
icculus@11517
  1228
    int retval;
slouken@10773
  1229
icculus@11583
  1230
    SDL_assert(inbuf != ((const float *) outbuf));  /* SDL_AudioStreamPut() shouldn't allow in-place resamples. */
slouken@11519
  1231
icculus@11517
  1232
    retval = SDL_ResampleAudio(chans, inrate, outrate, lpadding, rpadding, inbuf, inbuflen, outbuf, outbuflen);
slouken@10773
  1233
icculus@11517
  1234
    /* update our left padding with end of current input, for next run. */
icculus@11591
  1235
    SDL_memcpy((lpadding + paddingsamples) - (cpy / sizeof (float)), inbufend - cpy, cpy);
icculus@11517
  1236
    return retval;
slouken@10773
  1237
}
slouken@10773
  1238
slouken@10773
  1239
static void
slouken@10773
  1240
SDL_ResetAudioStreamResampler(SDL_AudioStream *stream)
slouken@10773
  1241
{
icculus@11583
  1242
    /* set all the padding to silence. */
icculus@11583
  1243
    const int len = stream->resampler_padding_samples;
icculus@11517
  1244
    SDL_memset(stream->resampler_state, '\0', len * sizeof (float));
slouken@10773
  1245
}
slouken@10773
  1246
slouken@10773
  1247
static void
slouken@10773
  1248
SDL_CleanupAudioStreamResampler(SDL_AudioStream *stream)
slouken@10773
  1249
{
slouken@10773
  1250
    SDL_free(stream->resampler_state);
slouken@10773
  1251
}
icculus@10757
  1252
icculus@10789
  1253
SDL_AudioStream *
icculus@10789
  1254
SDL_NewAudioStream(const SDL_AudioFormat src_format,
icculus@10789
  1255
                   const Uint8 src_channels,
icculus@10789
  1256
                   const int src_rate,
icculus@10789
  1257
                   const SDL_AudioFormat dst_format,
icculus@10789
  1258
                   const Uint8 dst_channels,
icculus@10789
  1259
                   const int dst_rate)
icculus@10757
  1260
{
icculus@10757
  1261
    const int packetlen = 4096;  /* !!! FIXME: good enough for now. */
icculus@10757
  1262
    Uint8 pre_resample_channels;
icculus@10757
  1263
    SDL_AudioStream *retval;
icculus@10757
  1264
icculus@10757
  1265
    retval = (SDL_AudioStream *) SDL_calloc(1, sizeof (SDL_AudioStream));
icculus@10757
  1266
    if (!retval) {
icculus@10757
  1267
        return NULL;
icculus@10757
  1268
    }
icculus@10757
  1269
icculus@10757
  1270
    /* If increasing channels, do it after resampling, since we'd just
icculus@10757
  1271
       do more work to resample duplicate channels. If we're decreasing, do
icculus@10757
  1272
       it first so we resample the interpolated data instead of interpolating
icculus@10757
  1273
       the resampled data (!!! FIXME: decide if that works in practice, though!). */
icculus@10757
  1274
    pre_resample_channels = SDL_min(src_channels, dst_channels);
icculus@10757
  1275
icculus@11583
  1276
    retval->first_run = SDL_TRUE;
icculus@10883
  1277
    retval->src_sample_frame_size = (SDL_AUDIO_BITSIZE(src_format) / 8) * src_channels;
icculus@10757
  1278
    retval->src_format = src_format;
icculus@10757
  1279
    retval->src_channels = src_channels;
icculus@10757
  1280
    retval->src_rate = src_rate;
icculus@10883
  1281
    retval->dst_sample_frame_size = (SDL_AUDIO_BITSIZE(dst_format) / 8) * dst_channels;
icculus@10757
  1282
    retval->dst_format = dst_format;
icculus@10757
  1283
    retval->dst_channels = dst_channels;
icculus@10757
  1284
    retval->dst_rate = dst_rate;
icculus@10757
  1285
    retval->pre_resample_channels = pre_resample_channels;
icculus@10757
  1286
    retval->packetlen = packetlen;
icculus@10757
  1287
    retval->rate_incr = ((double) dst_rate) / ((double) src_rate);
icculus@11583
  1288
    retval->resampler_padding_samples = ResamplerPadding(retval->src_rate, retval->dst_rate) * pre_resample_channels;
icculus@11583
  1289
    retval->resampler_padding = (float *) SDL_calloc(retval->resampler_padding_samples, sizeof (float));
icculus@11583
  1290
icculus@11583
  1291
    if (retval->resampler_padding == NULL) {
icculus@11583
  1292
        SDL_FreeAudioStream(retval);
icculus@11583
  1293
        SDL_OutOfMemory();
icculus@11583
  1294
        return NULL;
icculus@11583
  1295
    }
icculus@10757
  1296
icculus@10757
  1297
    /* Not resampling? It's an easy conversion (and maybe not even that!). */
icculus@10757
  1298
    if (src_rate == dst_rate) {
icculus@10757
  1299
        retval->cvt_before_resampling.needed = SDL_FALSE;
slouken@10773
  1300
        if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, src_format, src_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
slouken@10773
  1301
            SDL_FreeAudioStream(retval);
icculus@10757
  1302
            return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
icculus@10757
  1303
        }
icculus@10757
  1304
    } else {
icculus@10757
  1305
        /* Don't resample at first. Just get us to Float32 format. */
icculus@10757
  1306
        /* !!! FIXME: convert to int32 on devices without hardware float. */
slouken@10773
  1307
        if (SDL_BuildAudioCVT(&retval->cvt_before_resampling, src_format, src_channels, src_rate, AUDIO_F32SYS, pre_resample_channels, src_rate) < 0) {
slouken@10773
  1308
            SDL_FreeAudioStream(retval);
icculus@10757
  1309
            return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
icculus@10757
  1310
        }
icculus@10757
  1311
slouken@10777
  1312
#ifdef HAVE_LIBSAMPLERATE_H
slouken@10773
  1313
        SetupLibSampleRateResampling(retval);
slouken@10773
  1314
#endif
slouken@10773
  1315
slouken@10773
  1316
        if (!retval->resampler_func) {
icculus@11583
  1317
            retval->resampler_state = SDL_calloc(retval->resampler_padding_samples, sizeof (float));
slouken@10773
  1318
            if (!retval->resampler_state) {
slouken@10773
  1319
                SDL_FreeAudioStream(retval);
slouken@10773
  1320
                SDL_OutOfMemory();
slouken@10773
  1321
                return NULL;
slouken@10773
  1322
            }
icculus@11508
  1323
icculus@11508
  1324
            if (SDL_PrepareResampleFilter() < 0) {
icculus@11508
  1325
                SDL_free(retval->resampler_state);
icculus@11508
  1326
                retval->resampler_state = NULL;
icculus@11508
  1327
                SDL_FreeAudioStream(retval);
icculus@11508
  1328
                return NULL;
icculus@11508
  1329
            }
icculus@11508
  1330
slouken@10773
  1331
            retval->resampler_func = SDL_ResampleAudioStream;
slouken@10773
  1332
            retval->reset_resampler_func = SDL_ResetAudioStreamResampler;
slouken@10773
  1333
            retval->cleanup_resampler_func = SDL_CleanupAudioStreamResampler;
slouken@10773
  1334
        }
slouken@10773
  1335
icculus@10757
  1336
        /* Convert us to the final format after resampling. */
slouken@10773
  1337
        if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, AUDIO_F32SYS, pre_resample_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
slouken@10773
  1338
            SDL_FreeAudioStream(retval);
icculus@10757
  1339
            return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
icculus@10757
  1340
        }
icculus@10757
  1341
    }
icculus@10757
  1342
icculus@10757
  1343
    retval->queue = SDL_NewDataQueue(packetlen, packetlen * 2);
icculus@10757
  1344
    if (!retval->queue) {
slouken@10773
  1345
        SDL_FreeAudioStream(retval);
icculus@10757
  1346
        return NULL;  /* SDL_NewDataQueue should have called SDL_SetError. */
icculus@10757
  1347
    }
icculus@10757
  1348
icculus@10757
  1349
    return retval;
icculus@10757
  1350
}
icculus@10757
  1351
icculus@10757
  1352
int
icculus@10757
  1353
SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, const Uint32 _buflen)
icculus@10757
  1354
{
icculus@10757
  1355
    int buflen = (int) _buflen;
icculus@11583
  1356
    int workbuflen;
icculus@11583
  1357
    Uint8 *workbuf;
icculus@11583
  1358
    Uint8 *resamplebuf = NULL;
icculus@11583
  1359
    int resamplebuflen = 0;
icculus@11590
  1360
    int neededpaddingbytes;
icculus@11583
  1361
    int paddingbytes;
icculus@10757
  1362
icculus@10844
  1363
    /* !!! FIXME: several converters can take advantage of SIMD, but only
icculus@10844
  1364
       !!! FIXME:  if the data is aligned to 16 bytes. EnsureStreamBufferSize()
icculus@10844
  1365
       !!! FIXME:  guarantees the buffer will align, but the
icculus@10844
  1366
       !!! FIXME:  converters will iterate over the data backwards if
icculus@10844
  1367
       !!! FIXME:  the output grows, and this means we won't align if buflen
icculus@10844
  1368
       !!! FIXME:  isn't a multiple of 16. In these cases, we should chop off
icculus@10844
  1369
       !!! FIXME:  a few samples at the end and convert them separately. */
icculus@10844
  1370
icculus@11583
  1371
    #if DEBUG_AUDIOSTREAM
icculus@11583
  1372
    printf("AUDIOSTREAM: wants to put %d preconverted bytes\n", buflen);
icculus@11583
  1373
    #endif
icculus@11583
  1374
icculus@10757
  1375
    if (!stream) {
icculus@10757
  1376
        return SDL_InvalidParamError("stream");
icculus@10757
  1377
    } else if (!buf) {
icculus@10757
  1378
        return SDL_InvalidParamError("buf");
icculus@10757
  1379
    } else if (buflen == 0) {
icculus@10757
  1380
        return 0;  /* nothing to do. */
icculus@10757
  1381
    } else if ((buflen % stream->src_sample_frame_size) != 0) {
icculus@10757
  1382
        return SDL_SetError("Can't add partial sample frames");
icculus@11590
  1383
    } else if (buflen < ((stream->resampler_padding_samples / stream->pre_resample_channels) * stream->src_sample_frame_size)) {
icculus@11583
  1384
        return SDL_SetError("Need to put a larger buffer");
icculus@10757
  1385
    }
icculus@10757
  1386
icculus@11583
  1387
    /* no padding prepended on first run. */
icculus@11590
  1388
    neededpaddingbytes = stream->resampler_padding_samples * sizeof (float);
icculus@11583
  1389
    paddingbytes = stream->first_run ? 0 : neededpaddingbytes;
icculus@11583
  1390
    stream->first_run = SDL_FALSE;
icculus@11583
  1391
icculus@11583
  1392
    if (!stream->cvt_before_resampling.needed &&
icculus@11583
  1393
        (stream->dst_rate == stream->src_rate) &&
icculus@11583
  1394
        !stream->cvt_after_resampling.needed) {
icculus@11583
  1395
        #if DEBUG_AUDIOSTREAM
icculus@11583
  1396
        printf("AUDIOSTREAM: no conversion needed at all, queueing %d bytes.\n", buflen);
icculus@11583
  1397
        #endif
icculus@11583
  1398
        return SDL_WriteToDataQueue(stream->queue, buf, buflen);
icculus@11583
  1399
    }
icculus@11583
  1400
icculus@11583
  1401
    /* Make sure the work buffer can hold all the data we need at once... */
icculus@11583
  1402
    workbuflen = buflen;
icculus@10757
  1403
    if (stream->cvt_before_resampling.needed) {
icculus@11583
  1404
        workbuflen *= stream->cvt_before_resampling.len_mult;
icculus@11583
  1405
    }
icculus@11583
  1406
icculus@11583
  1407
    if (stream->dst_rate != stream->src_rate) {
icculus@11583
  1408
        /* resamples can't happen in place, so make space for second buf. */
icculus@11583
  1409
        const int framesize = stream->pre_resample_channels * sizeof (float);
icculus@11583
  1410
        const int frames = workbuflen / framesize;
icculus@11583
  1411
        resamplebuflen = ((int) SDL_ceil(frames * stream->rate_incr)) * framesize;
icculus@11583
  1412
        #if DEBUG_AUDIOSTREAM
icculus@11583
  1413
        printf("AUDIOSTREAM: will resample %d bytes to %d (ratio=%.6f)\n", workbuflen, resamplebuflen, stream->rate_incr);
icculus@11583
  1414
        #endif
icculus@11583
  1415
        workbuflen += resamplebuflen;
icculus@11583
  1416
    }
icculus@11583
  1417
icculus@11583
  1418
    if (stream->cvt_after_resampling.needed) {
icculus@11583
  1419
        /* !!! FIXME: buffer might be big enough already? */
icculus@11583
  1420
        workbuflen *= stream->cvt_after_resampling.len_mult;
icculus@11583
  1421
    }
icculus@11583
  1422
icculus@11583
  1423
    workbuflen += neededpaddingbytes;
icculus@11583
  1424
icculus@11583
  1425
    #if DEBUG_AUDIOSTREAM
icculus@11583
  1426
    printf("AUDIOSTREAM: Putting %d bytes of preconverted audio, need %d byte work buffer\n", buflen, workbuflen);
icculus@11583
  1427
    #endif
icculus@11583
  1428
icculus@11583
  1429
    workbuf = EnsureStreamBufferSize(stream, workbuflen);
icculus@11583
  1430
    if (!workbuf) {
icculus@11583
  1431
        return -1;  /* probably out of memory. */
icculus@11583
  1432
    }
icculus@11583
  1433
icculus@11583
  1434
    resamplebuf = workbuf;  /* default if not resampling. */
icculus@11583
  1435
icculus@11583
  1436
    SDL_memcpy(workbuf + paddingbytes, buf, buflen);
icculus@11583
  1437
icculus@11583
  1438
    if (stream->cvt_before_resampling.needed) {
icculus@11583
  1439
        stream->cvt_before_resampling.buf = workbuf + paddingbytes;
icculus@10757
  1440
        stream->cvt_before_resampling.len = buflen;
icculus@10757
  1441
        if (SDL_ConvertAudio(&stream->cvt_before_resampling) == -1) {
icculus@10757
  1442
            return -1;   /* uhoh! */
icculus@10757
  1443
        }
icculus@10757
  1444
        buflen = stream->cvt_before_resampling.len_cvt;
icculus@11583
  1445
icculus@11583
  1446
        #if DEBUG_AUDIOSTREAM
icculus@11583
  1447
        printf("AUDIOSTREAM: After initial conversion we have %d bytes\n", buflen);
icculus@11583
  1448
        #endif
icculus@10757
  1449
    }
icculus@10757
  1450
icculus@10757
  1451
    if (stream->dst_rate != stream->src_rate) {
icculus@11583
  1452
        /* save off some samples at the end; they are used for padding now so
icculus@11583
  1453
           the resampler is coherent and then used at the start of the next
icculus@11583
  1454
           put operation. Prepend last put operation's padding, too. */
icculus@11583
  1455
icculus@11583
  1456
        /* prepend prior put's padding. :P */
icculus@11583
  1457
        if (paddingbytes) {
icculus@11583
  1458
            SDL_memcpy(workbuf, stream->resampler_padding, paddingbytes);
icculus@11583
  1459
            buflen += paddingbytes;
icculus@10757
  1460
        }
icculus@11583
  1461
icculus@11583
  1462
        /* save off the data at the end for the next run. */
icculus@11583
  1463
        SDL_memcpy(stream->resampler_padding, workbuf + (buflen - neededpaddingbytes), neededpaddingbytes);
icculus@11583
  1464
icculus@11583
  1465
        resamplebuf = workbuf + buflen;  /* skip to second piece of workbuf. */
icculus@11591
  1466
        SDL_assert(buflen >= neededpaddingbytes);
icculus@11591
  1467
        if (buflen > neededpaddingbytes) {
icculus@11591
  1468
            buflen = stream->resampler_func(stream, workbuf, buflen - neededpaddingbytes, resamplebuf, resamplebuflen);
icculus@11591
  1469
        } else {
icculus@11591
  1470
            buflen = 0;
icculus@11591
  1471
        }
icculus@11583
  1472
icculus@11583
  1473
        #if DEBUG_AUDIOSTREAM
icculus@11583
  1474
        printf("AUDIOSTREAM: After resampling we have %d bytes\n", buflen);
icculus@11583
  1475
        #endif
icculus@10757
  1476
    }
icculus@10757
  1477
icculus@11591
  1478
    if (stream->cvt_after_resampling.needed && (buflen > 0)) {
icculus@11583
  1479
        stream->cvt_after_resampling.buf = resamplebuf;
icculus@10757
  1480
        stream->cvt_after_resampling.len = buflen;
icculus@10757
  1481
        if (SDL_ConvertAudio(&stream->cvt_after_resampling) == -1) {
icculus@10757
  1482
            return -1;   /* uhoh! */
icculus@10757
  1483
        }
icculus@10757
  1484
        buflen = stream->cvt_after_resampling.len_cvt;
icculus@11583
  1485
icculus@11583
  1486
        #if DEBUG_AUDIOSTREAM
icculus@11583
  1487
        printf("AUDIOSTREAM: After final conversion we have %d bytes\n", buflen);
icculus@11583
  1488
        #endif
icculus@10757
  1489
    }
icculus@10757
  1490
icculus@11583
  1491
    #if DEBUG_AUDIOSTREAM
icculus@11583
  1492
    printf("AUDIOSTREAM: Final output is %d bytes\n", buflen);
icculus@11583
  1493
    #endif
icculus@11583
  1494
icculus@11583
  1495
    /* resamplebuf holds the final output, even if we didn't resample. */
icculus@11591
  1496
    return buflen ? SDL_WriteToDataQueue(stream->queue, resamplebuf, buflen) : 0;
icculus@10757
  1497
}
icculus@10757
  1498
icculus@10757
  1499
void
icculus@10757
  1500
SDL_AudioStreamClear(SDL_AudioStream *stream)
icculus@10757
  1501
{
icculus@10757
  1502
    if (!stream) {
icculus@10757
  1503
        SDL_InvalidParamError("stream");
icculus@10757
  1504
    } else {
icculus@10757
  1505
        SDL_ClearDataQueue(stream->queue, stream->packetlen * 2);
icculus@10776
  1506
        if (stream->reset_resampler_func) {
icculus@10776
  1507
            stream->reset_resampler_func(stream);
icculus@10776
  1508
        }
icculus@11583
  1509
        stream->first_run = SDL_TRUE;
icculus@10757
  1510
    }
icculus@10757
  1511
}
icculus@10757
  1512
icculus@10757
  1513
icculus@10757
  1514
/* get converted/resampled data from the stream */
icculus@10757
  1515
int
icculus@10764
  1516
SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, const Uint32 len)
icculus@10757
  1517
{
icculus@11583
  1518
    #if DEBUG_AUDIOSTREAM
icculus@11583
  1519
    printf("AUDIOSTREAM: want to get %u converted bytes\n", (unsigned int) len);
icculus@11583
  1520
    #endif
icculus@11583
  1521
icculus@10757
  1522
    if (!stream) {
icculus@10757
  1523
        return SDL_InvalidParamError("stream");
icculus@10757
  1524
    } else if (!buf) {
icculus@10757
  1525
        return SDL_InvalidParamError("buf");
icculus@10757
  1526
    } else if (len == 0) {
icculus@10757
  1527
        return 0;  /* nothing to do. */
icculus@10757
  1528
    } else if ((len % stream->dst_sample_frame_size) != 0) {
icculus@10757
  1529
        return SDL_SetError("Can't request partial sample frames");
icculus@10757
  1530
    }
icculus@10757
  1531
icculus@10764
  1532
    return (int) SDL_ReadFromDataQueue(stream->queue, buf, len);
icculus@10757
  1533
}
icculus@10757
  1534
icculus@10757
  1535
/* number of converted/resampled bytes available */
icculus@10757
  1536
int
icculus@10757
  1537
SDL_AudioStreamAvailable(SDL_AudioStream *stream)
icculus@10757
  1538
{
icculus@10757
  1539
    return stream ? (int) SDL_CountDataQueue(stream->queue) : 0;
icculus@10757
  1540
}
icculus@10757
  1541
icculus@10757
  1542
/* dispose of a stream */
icculus@10757
  1543
void
icculus@10757
  1544
SDL_FreeAudioStream(SDL_AudioStream *stream)
icculus@10757
  1545
{
icculus@10757
  1546
    if (stream) {
slouken@10773
  1547
        if (stream->cleanup_resampler_func) {
slouken@10773
  1548
            stream->cleanup_resampler_func(stream);
slouken@10773
  1549
        }
icculus@10757
  1550
        SDL_FreeDataQueue(stream->queue);
icculus@10844
  1551
        SDL_free(stream->work_buffer_base);
icculus@11583
  1552
        SDL_free(stream->resampler_padding);
icculus@10757
  1553
        SDL_free(stream);
icculus@10757
  1554
    }
icculus@10757
  1555
}
icculus@10757
  1556
icculus@10575
  1557
/* vi: set ts=4 sw=4 expandtab: */
slouken@2716
  1558