annotate libao2/ao_pcm.c @ 22930:8a38e7270fe7

documented -mpegopts :interleaving2
author nicodvb
date Sun, 08 Apr 2007 16:21:32 +0000
parents fa99b3d31d13
children 7a6a7f47109f
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1 #include "config.h"
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2
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3 #include <stdio.h>
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4 #include <stdlib.h>
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5 #include <string.h>
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6
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7 #include "libavutil/common.h"
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8 #include "mpbswap.h"
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9 #include "subopt-helper.h"
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10 #include "libaf/af_format.h"
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11 #include "audio_out.h"
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12 #include "audio_out_internal.h"
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13 #include "mp_msg.h"
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14 #include "help_mp.h"
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15
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16
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17 static ao_info_t info =
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18 {
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19 "RAW PCM/WAVE file writer audio output",
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20 "pcm",
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21 "Atmosfear",
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22 ""
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23 };
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24
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25 LIBAO_EXTERN(pcm)
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26
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27 extern int vo_pts;
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28
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29 static char *ao_outputfilename = NULL;
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30 static int ao_pcm_waveheader = 1;
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31 static int fast = 0;
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32
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33 #define WAV_ID_RIFF 0x46464952 /* "RIFF" */
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34 #define WAV_ID_WAVE 0x45564157 /* "WAVE" */
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35 #define WAV_ID_FMT 0x20746d66 /* "fmt " */
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36 #define WAV_ID_DATA 0x61746164 /* "data" */
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37 #define WAV_ID_PCM 0x0001
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38
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39 struct WaveHeader
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40 {
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41 uint32_t riff;
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42 uint32_t file_length;
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43 uint32_t wave;
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44 uint32_t fmt;
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45 uint32_t fmt_length;
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46 uint16_t fmt_tag;
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47 uint16_t channels;
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48 uint32_t sample_rate;
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49 uint32_t bytes_per_second;
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50 uint16_t block_align;
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51 uint16_t bits;
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52 uint32_t data;
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53 uint32_t data_length;
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54 };
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55
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56 /* init with default values */
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57 static struct WaveHeader wavhdr;
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58
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59 static FILE *fp = NULL;
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60
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61 // to set/get/query special features/parameters
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62 static int control(int cmd,void *arg){
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63 return -1;
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64 }
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65
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66 // open & setup audio device
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67 // return: 1=success 0=fail
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68 static int init(int rate,int channels,int format,int flags){
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69 int bits;
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70 opt_t subopts[] = {
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71 {"waveheader", OPT_ARG_BOOL, &ao_pcm_waveheader, NULL},
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72 {"file", OPT_ARG_MSTRZ, &ao_outputfilename, NULL},
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73 {"fast", OPT_ARG_BOOL, &fast, NULL},
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74 {NULL}
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75 };
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76 // set defaults
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77 ao_pcm_waveheader = 1;
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78
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79 if (subopt_parse(ao_subdevice, subopts) != 0) {
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80 return 0;
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81 }
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82 if (!ao_outputfilename){
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83 ao_outputfilename =
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84 strdup(ao_pcm_waveheader?"audiodump.wav":"audiodump.pcm");
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85 }
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86
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87 /* bits is only equal to format if (format == 8) or (format == 16);
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88 this means that the following "if" is a kludge and should
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89 really be a switch to be correct in all cases */
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90
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91 bits=8;
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92 switch(format){
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93 case AF_FORMAT_S8:
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94 format=AF_FORMAT_U8;
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95 case AF_FORMAT_U8:
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96 break;
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97 default:
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98 format=AF_FORMAT_S16_LE;
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99 bits=16;
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100 break;
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101 }
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102
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103 ao_data.outburst = 65536;
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104 ao_data.buffersize= 2*65536;
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105 ao_data.channels=channels;
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106 ao_data.samplerate=rate;
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107 ao_data.format=format;
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108 ao_data.bps=channels*rate*(bits/8);
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109
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110 wavhdr.riff = le2me_32(WAV_ID_RIFF);
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111 wavhdr.wave = le2me_32(WAV_ID_WAVE);
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112 wavhdr.fmt = le2me_32(WAV_ID_FMT);
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113 wavhdr.fmt_length = le2me_32(16);
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114 wavhdr.fmt_tag = le2me_16(WAV_ID_PCM);
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115 wavhdr.channels = le2me_16(ao_data.channels);
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116 wavhdr.sample_rate = le2me_32(ao_data.samplerate);
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117 wavhdr.bytes_per_second = le2me_32(ao_data.bps);
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118 wavhdr.bits = le2me_16(bits);
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119 wavhdr.block_align = le2me_16(ao_data.channels * (bits / 8));
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120
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121 wavhdr.data = le2me_32(WAV_ID_DATA);
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122 wavhdr.data_length=le2me_32(0x7ffff000);
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123 wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8;
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124
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125 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename,
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126 (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate,
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127 (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
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128 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_HintInfo);
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129
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130 fp = fopen(ao_outputfilename, "wb");
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131 if(fp) {
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132 if(ao_pcm_waveheader){ /* Reserve space for wave header */
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133 fwrite(&wavhdr,sizeof(wavhdr),1,fp);
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134 wavhdr.file_length=wavhdr.data_length=0;
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135 }
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136 return 1;
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137 }
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138 mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_PCM_CantOpenOutputFile,
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139 ao_outputfilename);
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140 return 0;
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141 }
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142
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143 // close audio device
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144 static void uninit(int immed){
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145
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146 if(ao_pcm_waveheader && fseek(fp, 0, SEEK_SET) == 0){ /* Write wave header */
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147 wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8;
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148 wavhdr.file_length = le2me_32(wavhdr.file_length);
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149 wavhdr.data_length = le2me_32(wavhdr.data_length);
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150 fwrite(&wavhdr,sizeof(wavhdr),1,fp);
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151 }
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152 fclose(fp);
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153 if (ao_outputfilename)
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154 free(ao_outputfilename);
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155 ao_outputfilename = NULL;
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156 }
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157
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158 // stop playing and empty buffers (for seeking/pause)
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159 static void reset(void){
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160
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161 }
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162
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163 // stop playing, keep buffers (for pause)
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164 static void audio_pause(void)
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165 {
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166 // for now, just call reset();
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167 reset();
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168 }
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169
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170 // resume playing, after audio_pause()
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171 static void audio_resume(void)
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172 {
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173 }
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174
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175 // return: how many bytes can be played without blocking
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176 static int get_space(void){
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177
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178 if(vo_pts)
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179 return ao_data.pts < vo_pts + fast * 30000 ? ao_data.outburst : 0;
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180 return ao_data.outburst;
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181 }
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182
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183 // plays 'len' bytes of 'data'
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184 // it should round it down to outburst*n
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185 // return: number of bytes played
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186 static int play(void* data,int len,int flags){
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187
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188 // let libaf to do the conversion...
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189 #if 0
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190 //#ifdef WORDS_BIGENDIAN
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191 if (ao_data.format == AFMT_S16_LE) {
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192 unsigned short *buffer = (unsigned short *) data;
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193 register int i;
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194 for(i = 0; i < len/2; ++i) {
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195 buffer[i] = le2me_16(buffer[i]);
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196 }
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197 }
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198 #endif
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199
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200 //printf("PCM: Writing chunk!\n");
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201 fwrite(data,len,1,fp);
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202
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203 if(ao_pcm_waveheader)
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204 wavhdr.data_length += len;
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205
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206 return len;
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207 }
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208
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209 // return: delay in seconds between first and last sample in buffer
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210 static float get_delay(void){
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211
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212 return 0.0;
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213 }
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214
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216
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218
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219