annotate libao2/ao_pcm.c @ 28428:8df00b0c4f0d

We use libdvdcss 1.2.10, not 1.2.9.
author diego
date Wed, 04 Feb 2009 18:32:32 +0000
parents e45b08f2f5d3
children 9a5b8c2ed6de
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rev   line source
28343
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1 /*
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2 * PCM audio output driver
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3 *
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4 * This file is part of MPlayer.
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5 *
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6 * MPlayer is free software; you can redistribute it and/or modify
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7 * it under the terms of the GNU General Public License as published by
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8 * the Free Software Foundation; either version 2 of the License, or
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9 * (at your option) any later version.
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10 *
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11 * MPlayer is distributed in the hope that it will be useful,
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12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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14 * GNU General Public License for more details.
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15 *
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16 * You should have received a copy of the GNU General Public License along
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17 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
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18 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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19 */
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20
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21 #include "config.h"
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22
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23 #include <stdio.h>
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24 #include <stdlib.h>
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25 #include <string.h>
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26
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27 #include "libavutil/common.h"
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28 #include "mpbswap.h"
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29 #include "subopt-helper.h"
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30 #include "libaf/af_format.h"
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31 #include "libaf/reorder_ch.h"
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32 #include "audio_out.h"
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33 #include "audio_out_internal.h"
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34 #include "mp_msg.h"
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35 #include "help_mp.h"
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36
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37
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38 static ao_info_t info =
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39 {
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40 "RAW PCM/WAVE file writer audio output",
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41 "pcm",
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42 "Atmosfear",
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43 ""
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44 };
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45
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46 LIBAO_EXTERN(pcm)
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47
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48 extern int vo_pts;
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49
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50 static char *ao_outputfilename = NULL;
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51 static int ao_pcm_waveheader = 1;
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52 static int fast = 0;
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53
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54 #define WAV_ID_RIFF 0x46464952 /* "RIFF" */
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55 #define WAV_ID_WAVE 0x45564157 /* "WAVE" */
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56 #define WAV_ID_FMT 0x20746d66 /* "fmt " */
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57 #define WAV_ID_DATA 0x61746164 /* "data" */
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58 #define WAV_ID_PCM 0x0001
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59 #define WAV_ID_FLOAT_PCM 0x0003
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60
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61 struct WaveHeader
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62 {
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63 uint32_t riff;
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64 uint32_t file_length;
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65 uint32_t wave;
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66 uint32_t fmt;
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67 uint32_t fmt_length;
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68 uint16_t fmt_tag;
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69 uint16_t channels;
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70 uint32_t sample_rate;
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71 uint32_t bytes_per_second;
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72 uint16_t block_align;
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73 uint16_t bits;
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74 uint32_t data;
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75 uint32_t data_length;
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76 };
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77
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78 /* init with default values */
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79 static struct WaveHeader wavhdr;
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80
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81 static FILE *fp = NULL;
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82
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83 // to set/get/query special features/parameters
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84 static int control(int cmd,void *arg){
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85 return -1;
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86 }
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87
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88 // open & setup audio device
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89 // return: 1=success 0=fail
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90 static int init(int rate,int channels,int format,int flags){
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91 int bits;
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92 opt_t subopts[] = {
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93 {"waveheader", OPT_ARG_BOOL, &ao_pcm_waveheader, NULL},
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94 {"file", OPT_ARG_MSTRZ, &ao_outputfilename, NULL},
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95 {"fast", OPT_ARG_BOOL, &fast, NULL},
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96 {NULL}
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97 };
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98 // set defaults
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99 ao_pcm_waveheader = 1;
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100
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101 if (subopt_parse(ao_subdevice, subopts) != 0) {
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102 return 0;
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103 }
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104 if (!ao_outputfilename){
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105 ao_outputfilename =
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106 strdup(ao_pcm_waveheader?"audiodump.wav":"audiodump.pcm");
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107 }
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108
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109 bits=8;
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110 switch(format){
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111 case AF_FORMAT_S32_BE:
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112 format=AF_FORMAT_S32_LE;
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113 case AF_FORMAT_S32_LE:
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114 bits=32;
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115 break;
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116 case AF_FORMAT_FLOAT_BE:
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117 format=AF_FORMAT_FLOAT_LE;
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118 case AF_FORMAT_FLOAT_LE:
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119 bits=32;
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120 break;
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121 case AF_FORMAT_S8:
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122 format=AF_FORMAT_U8;
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123 case AF_FORMAT_U8:
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124 break;
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125 case AF_FORMAT_AC3:
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126 bits=16;
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127 break;
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128 default:
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129 format=AF_FORMAT_S16_LE;
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130 bits=16;
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131 break;
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132 }
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133
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134 ao_data.outburst = 65536;
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135 ao_data.buffersize= 2*65536;
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136 ao_data.channels=channels;
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137 ao_data.samplerate=rate;
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138 ao_data.format=format;
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139 ao_data.bps=channels*rate*(bits/8);
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140
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141 wavhdr.riff = le2me_32(WAV_ID_RIFF);
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142 wavhdr.wave = le2me_32(WAV_ID_WAVE);
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143 wavhdr.fmt = le2me_32(WAV_ID_FMT);
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144 wavhdr.fmt_length = le2me_32(16);
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145 wavhdr.fmt_tag = le2me_16(format == AF_FORMAT_FLOAT_LE ? WAV_ID_FLOAT_PCM : WAV_ID_PCM);
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146 wavhdr.channels = le2me_16(ao_data.channels);
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147 wavhdr.sample_rate = le2me_32(ao_data.samplerate);
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148 wavhdr.bytes_per_second = le2me_32(ao_data.bps);
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149 wavhdr.bits = le2me_16(bits);
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150 wavhdr.block_align = le2me_16(ao_data.channels * (bits / 8));
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151
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152 wavhdr.data = le2me_32(WAV_ID_DATA);
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153 wavhdr.data_length=le2me_32(0x7ffff000);
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154 wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8;
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155
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156 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename,
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157 (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate,
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158 (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
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159 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_HintInfo);
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160
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161 fp = fopen(ao_outputfilename, "wb");
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162 if(fp) {
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163 if(ao_pcm_waveheader){ /* Reserve space for wave header */
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164 fwrite(&wavhdr,sizeof(wavhdr),1,fp);
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d824f1a3c341 The "initialize wav header with infinite lenght" broke the
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165 wavhdr.file_length=wavhdr.data_length=0;
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166 }
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167 return 1;
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168 }
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169 mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_PCM_CantOpenOutputFile,
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170 ao_outputfilename);
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171 return 0;
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172 }
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173
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174 // close audio device
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175 static void uninit(int immed){
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176
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177 if(ao_pcm_waveheader && fseek(fp, 0, SEEK_SET) == 0){ /* Write wave header */
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178 wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8;
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179 wavhdr.file_length = le2me_32(wavhdr.file_length);
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180 wavhdr.data_length = le2me_32(wavhdr.data_length);
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181 fwrite(&wavhdr,sizeof(wavhdr),1,fp);
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182 }
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183 fclose(fp);
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184 if (ao_outputfilename)
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185 free(ao_outputfilename);
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186 ao_outputfilename = NULL;
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187 }
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188
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189 // stop playing and empty buffers (for seeking/pause)
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190 static void reset(void){
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191
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192 }
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193
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194 // stop playing, keep buffers (for pause)
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195 static void audio_pause(void)
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196 {
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197 // for now, just call reset();
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198 reset();
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199 }
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200
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201 // resume playing, after audio_pause()
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202 static void audio_resume(void)
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203 {
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204 }
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205
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206 // return: how many bytes can be played without blocking
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207 static int get_space(void){
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208
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209 if(vo_pts)
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210 return ao_data.pts < vo_pts + fast * 30000 ? ao_data.outburst : 0;
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211 return ao_data.outburst;
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212 }
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213
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214 // plays 'len' bytes of 'data'
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215 // it should round it down to outburst*n
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216 // return: number of bytes played
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217 static int play(void* data,int len,int flags){
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218
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219 // let libaf to do the conversion...
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220 #if 0
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221 //#ifdef WORDS_BIGENDIAN
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222 if (ao_data.format == AFMT_S16_LE) {
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223 unsigned short *buffer = (unsigned short *) data;
65037d309de3 updated for libaf rules
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224 register int i;
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225 for(i = 0; i < len/2; ++i) {
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226 buffer[i] = le2me_16(buffer[i]);
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227 }
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atmos4
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228 }
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229 #endif
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230
25315
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ulion
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231 if (ao_data.channels == 6 || ao_data.channels == 5) {
dfa8a510c81c Fix all current known multi-channel wrong order problems by adding
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232 int frame_size = le2me_16(wavhdr.bits) / 8;
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233 len -= len % (frame_size * ao_data.channels);
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ulion
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234 reorder_channel_nch(data, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
dfa8a510c81c Fix all current known multi-channel wrong order problems by adding
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235 AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
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236 ao_data.channels,
dfa8a510c81c Fix all current known multi-channel wrong order problems by adding
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237 len / frame_size, frame_size);
dfa8a510c81c Fix all current known multi-channel wrong order problems by adding
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238 }
dfa8a510c81c Fix all current known multi-channel wrong order problems by adding
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239
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240 //printf("PCM: Writing chunk!\n");
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241 fwrite(data,len,1,fp);
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242
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243 if(ao_pcm_waveheader)
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244 wavhdr.data_length += len;
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245
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246 return len;
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247 }
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248
3095
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anders
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249 // return: delay in seconds between first and last sample in buffer
17566
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rathann
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diff changeset
250 static float get_delay(void){
1107
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251
3095
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anders
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252 return 0.0;
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253 }
9d764880d989 Added raw PCM writer ao driver.
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254
9d764880d989 Added raw PCM writer ao driver.
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255
9d764880d989 Added raw PCM writer ao driver.
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256
9d764880d989 Added raw PCM writer ao driver.
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257
9d764880d989 Added raw PCM writer ao driver.
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258
9d764880d989 Added raw PCM writer ao driver.
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259