annotate libao2/ao_openal.c @ 33926:6e3f257d7b05

Add disabled feature: noise filter for vo_gl.
author reimar
date Fri, 26 Aug 2011 18:51:51 +0000
parents 32725ca88fed
children
Ignore whitespace changes - Everywhere: Within whitespace: At end of lines:
rev   line source
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1 /*
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2 * OpenAL audio output driver for MPlayer
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3 *
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4 * Copyleft 2006 by Reimar Döffinger (Reimar.Doeffinger@stud.uni-karlsruhe.de)
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5 *
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6 * This file is part of MPlayer.
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7 *
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8 * MPlayer is free software; you can redistribute it and/or modify
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diego
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9 * it under the terms of the GNU General Public License as published by
d97a607821f1 Replace casual GPL notices by proper license headers.
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10 * the Free Software Foundation; either version 2 of the License, or
d97a607821f1 Replace casual GPL notices by proper license headers.
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11 * (at your option) any later version.
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12 *
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13 * MPlayer is distributed in the hope that it will be useful,
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14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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16 * GNU General Public License for more details.
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17 *
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18 * You should have received a copy of the GNU General Public License along
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19 * along with MPlayer; if not, write to the Free Software
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20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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21 */
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22
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23 #include "config.h"
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24
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25 #include <stdlib.h>
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26 #include <stdio.h>
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27 #include <inttypes.h>
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28 #ifdef OPENAL_AL_H
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29 #include <OpenAL/alc.h>
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30 #include <OpenAL/al.h>
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31 #else
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32 #include <AL/alc.h>
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33 #include <AL/al.h>
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34 #endif
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35
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36 #include "mp_msg.h"
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37 #include "help_mp.h"
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38
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39 #include "audio_out.h"
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40 #include "audio_out_internal.h"
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41 #include "libaf/af_format.h"
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42 #include "osdep/timer.h"
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43 #include "subopt-helper.h"
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44
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45 static const ao_info_t info =
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46 {
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47 "OpenAL audio output",
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48 "openal",
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49 "Reimar Döffinger <Reimar.Doeffinger@stud.uni-karlsruhe.de>",
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50 ""
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51 };
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52
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53 LIBAO_EXTERN(openal)
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54
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55 #define MAX_CHANS 8
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56 #define NUM_BUF 128
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57 #define CHUNK_SIZE 512
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58 static ALuint buffers[MAX_CHANS][NUM_BUF];
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59 static ALuint sources[MAX_CHANS];
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60
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61 static int cur_buf[MAX_CHANS];
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62 static int unqueue_buf[MAX_CHANS];
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63 static int16_t *tmpbuf;
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64
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65
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66 static int control(int cmd, void *arg) {
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67 switch (cmd) {
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68 case AOCONTROL_GET_VOLUME:
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69 case AOCONTROL_SET_VOLUME: {
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70 ALfloat volume;
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71 ao_control_vol_t *vol = (ao_control_vol_t *)arg;
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72 if (cmd == AOCONTROL_SET_VOLUME) {
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73 volume = (vol->left + vol->right) / 200.0;
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74 alListenerf(AL_GAIN, volume);
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75 }
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76 alGetListenerf(AL_GAIN, &volume);
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77 vol->left = vol->right = volume * 100;
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78 return CONTROL_TRUE;
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79 }
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80 }
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81 return CONTROL_UNKNOWN;
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82 }
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83
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84 /**
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85 * \brief print suboption usage help
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86 */
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87 static void print_help(void) {
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88 mp_msg(MSGT_AO, MSGL_FATAL,
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89 "\n-ao openal commandline help:\n"
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90 "Example: mplayer -ao openal\n"
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91 "\nOptions:\n"
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92 );
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93 }
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94
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95 static int init(int rate, int channels, int format, int flags) {
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96 float position[3] = {0, 0, 0};
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97 float direction[6] = {0, 0, 1, 0, -1, 0};
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98 float sppos[MAX_CHANS][3] = {
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99 {-1, 0, 0.5}, {1, 0, 0.5},
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100 {-1, 0, -1}, {1, 0, -1},
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101 {0, 0, 1}, {0, 0, 0.1},
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102 {-1, 0, 0}, {1, 0, 0},
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103 };
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104 ALCdevice *dev = NULL;
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105 ALCcontext *ctx = NULL;
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106 ALCint freq = 0;
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107 ALCint attribs[] = {ALC_FREQUENCY, rate, 0, 0};
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108 int i;
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2eff450157cd The suboption parser now takes a const options list, so mark them all const.
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109 const opt_t subopts[] = {
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110 {NULL}
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111 };
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112 if (subopt_parse(ao_subdevice, subopts) != 0) {
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113 print_help();
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114 return 0;
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115 }
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116 if (channels > MAX_CHANS) {
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117 mp_msg(MSGT_AO, MSGL_FATAL, "[OpenAL] Invalid number of channels: %i\n", channels);
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118 goto err_out;
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119 }
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120 dev = alcOpenDevice(NULL);
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121 if (!dev) {
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122 mp_msg(MSGT_AO, MSGL_FATAL, "[OpenAL] could not open device\n");
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123 goto err_out;
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124 }
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125 ctx = alcCreateContext(dev, attribs);
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126 alcMakeContextCurrent(ctx);
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127 alListenerfv(AL_POSITION, position);
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128 alListenerfv(AL_ORIENTATION, direction);
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129 alGenSources(channels, sources);
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130 for (i = 0; i < channels; i++) {
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131 cur_buf[i] = 0;
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132 unqueue_buf[i] = 0;
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133 alGenBuffers(NUM_BUF, buffers[i]);
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134 alSourcefv(sources[i], AL_POSITION, sppos[i]);
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135 alSource3f(sources[i], AL_VELOCITY, 0, 0, 0);
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136 }
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137 if (channels == 1)
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138 alSource3f(sources[0], AL_POSITION, 0, 0, 1);
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139 ao_data.channels = channels;
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140 alcGetIntegerv(dev, ALC_FREQUENCY, 1, &freq);
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141 if (alcGetError(dev) == ALC_NO_ERROR && freq)
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142 rate = freq;
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143 ao_data.samplerate = rate;
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144 ao_data.format = AF_FORMAT_S16_NE;
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145 ao_data.bps = channels * rate * 2;
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146 ao_data.buffersize = CHUNK_SIZE * NUM_BUF;
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147 ao_data.outburst = channels * CHUNK_SIZE;
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148 tmpbuf = malloc(CHUNK_SIZE);
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149 return 1;
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150
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151 err_out:
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152 return 0;
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153 }
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154
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155 // close audio device
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156 static void uninit(int immed) {
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157 ALCcontext *ctx = alcGetCurrentContext();
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158 ALCdevice *dev = alcGetContextsDevice(ctx);
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159 free(tmpbuf);
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160 if (!immed) {
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161 ALint state;
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162 alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
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163 while (state == AL_PLAYING) {
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164 usec_sleep(10000);
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165 alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
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166 }
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167 }
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168 reset();
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169 alcMakeContextCurrent(NULL);
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170 alcDestroyContext(ctx);
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171 alcCloseDevice(dev);
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172 }
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173
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174 static void unqueue_buffers(void) {
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175 ALint p;
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74db93fee74f Remove unused variable.
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176 int s;
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177 for (s = 0; s < ao_data.channels; s++) {
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178 int till_wrap = NUM_BUF - unqueue_buf[s];
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179 alGetSourcei(sources[s], AL_BUFFERS_PROCESSED, &p);
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180 if (p >= till_wrap) {
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181 alSourceUnqueueBuffers(sources[s], till_wrap, &buffers[s][unqueue_buf[s]]);
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182 unqueue_buf[s] = 0;
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183 p -= till_wrap;
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184 }
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185 if (p) {
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186 alSourceUnqueueBuffers(sources[s], p, &buffers[s][unqueue_buf[s]]);
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187 unqueue_buf[s] += p;
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188 }
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189 }
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190 }
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191
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192 /**
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193 * \brief stop playing and empty buffers (for seeking/pause)
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194 */
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195 static void reset(void) {
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196 alSourceStopv(ao_data.channels, sources);
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197 unqueue_buffers();
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198 }
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199
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200 /**
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201 * \brief stop playing, keep buffers (for pause)
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202 */
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203 static void audio_pause(void) {
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204 alSourcePausev(ao_data.channels, sources);
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205 }
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206
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207 /**
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208 * \brief resume playing, after audio_pause()
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209 */
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210 static void audio_resume(void) {
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211 alSourcePlayv(ao_data.channels, sources);
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212 }
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213
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214 static int get_space(void) {
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215 ALint queued;
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216 unqueue_buffers();
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217 alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
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218 queued = NUM_BUF - queued - 3;
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219 if (queued < 0) return 0;
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220 return queued * CHUNK_SIZE * ao_data.channels;
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221 }
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222
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223 /**
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224 * \brief write data into buffer and reset underrun flag
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225 */
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226 static int play(void *data, int len, int flags) {
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227 ALint state;
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228 int i, j, k;
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229 int ch;
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230 int16_t *d = data;
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231 len /= ao_data.channels * CHUNK_SIZE;
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232 for (i = 0; i < len; i++) {
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233 for (ch = 0; ch < ao_data.channels; ch++) {
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234 for (j = 0, k = ch; j < CHUNK_SIZE / 2; j++, k += ao_data.channels)
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235 tmpbuf[j] = d[k];
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236 alBufferData(buffers[ch][cur_buf[ch]], AL_FORMAT_MONO16, tmpbuf,
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237 CHUNK_SIZE, ao_data.samplerate);
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238 alSourceQueueBuffers(sources[ch], 1, &buffers[ch][cur_buf[ch]]);
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239 cur_buf[ch] = (cur_buf[ch] + 1) % NUM_BUF;
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240 }
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241 d += ao_data.channels * CHUNK_SIZE / 2;
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242 }
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243 alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
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244 if (state != AL_PLAYING) // checked here in case of an underrun
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245 alSourcePlayv(ao_data.channels, sources);
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246 return len * ao_data.channels * CHUNK_SIZE;
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247 }
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248
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249 static float get_delay(void) {
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250 ALint queued;
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251 unqueue_buffers();
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252 alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
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253 return queued * CHUNK_SIZE / 2 / (float)ao_data.samplerate;
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254 }