Mercurial > SDL_sound_CoreAudio

annotate mixer/mixercore.c @ 591:8faf61a640f0 tip

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Resynced fixes for unit conversion bugs in the Ogg Tremor decoder from SoundDecoder/ALmixer. Ogg Vorbis uses seconds and we multiply by 1000 to convert to milliseconds. But Ogg Tremor already uses milliseconds but I was still multiplying by 1000.
author Eric Wing <ewing . public |-at-| gmail . com>
date Thu, 25 Oct 2012 16:34:18 -0700
parents 2e8907ff98e9
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rev   line source
486
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1 /*
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2 * SDL_sound -- An sound processing toolkit.
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3 * Copyright (C) 2001 Ryan C. Gordon.
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4 *
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5 * This library is free software; you can redistribute it and/or
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6 * modify it under the terms of the GNU Lesser General Public
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7 * License as published by the Free Software Foundation; either
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8 * version 2.1 of the License, or (at your option) any later version.
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9 *
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10 * This library is distributed in the hope that it will be useful,
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11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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13 * Lesser General Public License for more details.
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14 *
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15 * You should have received a copy of the GNU Lesser General Public
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16 * License along with this library; if not, write to the Free Software
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17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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18 */
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19
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20 /**
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21 * This file implements the mixer itself. Largely, this is handled in the
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22 * SDL audio callback.
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23 *
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24 * Documentation is in SDL_sound.h ... It's verbose, honest. :)
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25 *
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26 * Please see the file LICENSE.txt in the source's root directory.
486
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27 *
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28 * This file written by Ryan C. Gordon. (icculus@icculus.org)
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29 */
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30
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31 #if HAVE_CONFIG_H
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32 # include <config.h>
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33 #endif
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34
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35 #include <stdio.h>
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36 #include <stdlib.h>
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37 #include <string.h>
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38
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39 #include "SDL.h"
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40 #include "SDL_thread.h"
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41 #include "SDL_sound.h"
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42
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43 #define __SDL_SOUND_INTERNAL__
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44 #include "SDL_sound_internal.h"
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45
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46
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47
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48 typedef struct S_PlayingList
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49 {
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50 Sound_Sample *sample;
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51 struct S_PlayingList *next;
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52 } PlayingList;
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53
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54 static PlayingList *playlist = NULL;
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55
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56 static inline void mix_predecoded(Sound_Sample *samp,
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57 UInt32 *samp_frames_left,
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58 float *gains)
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59 {
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60 Sound_SampleInternal *internal = (Sound_SampleInternal *) samp->opaque;
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61 Uint32 sfl = *samp_frames_left; /* move to a local. */
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62 Uint32 max = internal->buffer_size - internal->mix_position;
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63 float *wptr; /* write pointer */
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64
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65 /* !!! FIXME: max must be converted to sample frame count... */
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66
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67 if (max > sfl) /* we have more data than mix buffer? */
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68 max = sfl;
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69
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70 assert(max > 0);
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71 *samp_frames_left -= max;
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72
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73 wptr = mixbuf + ((mixbufsize / sizeof (float)) - (max * MAX_CHANNELS));
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74 internal->mix(wptr, internal->buffer, max, gains);
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75 } /* mix_predecoded */
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77
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78 static void mix_playing_samples(Uint8 *stream, int len)
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79 {
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80 PlayingList *samples = playlist;
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81 const int frames = len / framesize;
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82 const Uint32 ticks = SDL_GetTicks(); /* used for calculating fade. */
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83
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84 while (samples) /* iterate linked list of playing samples... */
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85 {
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86 Sound_Sample *samp = samples->sample;
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87 Uint32 sample_frames_left = mixbuf_frames;
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88 float gains[MAX_CHANNELS];
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89
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90 calculate_gains(samp, ticks, gains);
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91 while (sample_frames_left)
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92 {
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93 mix_predecoded(samp, &sample_frames_left);
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94 if (!decode_more(samp))
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95 break;
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96 } /* while */
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97
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98 samples = samples->next; /* set up for next iteration. */
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99 } /* while */
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100 } /* mix_playing_samples */
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101
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102
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103 static inline void run_pre_mix(void)
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104 {
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105 if (premixer)
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106 premixer(mixbuf, mixbufsize);
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107 else /* !!! FIXME: Do memset in another thread after mix is done. */
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108 memset(mixbuf, '\0', mixbufsize * sizeof (float) * 2);
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109 } /* run_pre_mix */
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110
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111
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112 static inline void run_post_mix(void)
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113 {
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114 if (postmixer)
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115 postmixer(mixbuf, mixbufsize);
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116 } /* run_post_mix */
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117
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118
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119 /* this is where it happens: the SDL audio callback. */
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120 static void audio_callback(void *userdata, Uint8 *stream, int len)
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121 {
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122 mixer_callback_running = 1;
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123 run_pre_mix();
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124 mix_playing_samples();
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125 run_post_mix();
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126 mixer_callback_running = 0;
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127 } /* audio_callback */
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128
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129 /* end of mixercore.c ... */
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130