annotate libao2/ao_pcm.c @ 30882:a5c190c3ce4d

Rename SGI_MIPS architecture to plain MIPS.
author diego
date Wed, 17 Mar 2010 14:09:09 +0000
parents b20827f16dd0
children f5478d5be47e
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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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1767c271d710 Remove bswap.h, use libavutil/bswap.h instead.
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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 #ifdef __MINGW32__
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38 // for GetFileType to detect pipes
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39 #include <windows.h>
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40 #endif
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41
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42 static const ao_info_t info =
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43 {
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44 "RAW PCM/WAVE file writer audio output",
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45 "pcm",
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46 "Atmosfear",
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47 ""
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48 };
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49
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50 LIBAO_EXTERN(pcm)
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51
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52 extern int vo_pts;
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53
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54 static char *ao_outputfilename = NULL;
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55 static int ao_pcm_waveheader = 1;
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56 static int fast = 0;
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57
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58 #define WAV_ID_RIFF 0x46464952 /* "RIFF" */
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59 #define WAV_ID_WAVE 0x45564157 /* "WAVE" */
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60 #define WAV_ID_FMT 0x20746d66 /* "fmt " */
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61 #define WAV_ID_DATA 0x61746164 /* "data" */
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62 #define WAV_ID_PCM 0x0001
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63 #define WAV_ID_FLOAT_PCM 0x0003
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64 #define WAV_ID_FORMAT_EXTENSIBLE 0xfffe
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65
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66 /* init with default values */
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67 static uint64_t data_length;
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68 static FILE *fp = NULL;
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69
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70
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71 static void fput16le(uint16_t val, FILE *fp) {
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72 uint8_t bytes[2] = {val, val >> 8};
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73 fwrite(bytes, 1, 2, fp);
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74 }
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75
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76 static void fput32le(uint32_t val, FILE *fp) {
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77 uint8_t bytes[4] = {val, val >> 8, val >> 16, val >> 24};
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78 fwrite(bytes, 1, 4, fp);
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79 }
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80
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81 static void write_wave_header(FILE *fp, uint64_t data_length) {
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82 int use_waveex = (ao_data.channels >= 5 && ao_data.channels <= 8);
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83 uint16_t fmt = (ao_data.format == AF_FORMAT_FLOAT_LE) ? WAV_ID_FLOAT_PCM : WAV_ID_PCM;
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84 uint32_t fmt_chunk_size = use_waveex ? 40 : 16;
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85 int bits = af_fmt2bits(ao_data.format);
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86
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87 // Master RIFF chunk
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88 fput32le(WAV_ID_RIFF, fp);
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89 // RIFF chunk size: 'WAVE' + 'fmt ' + 4 + fmt_chunk_size + data chunk hdr (8) + data length
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90 fput32le(12 + fmt_chunk_size + 8 + data_length, fp);
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91 fput32le(WAV_ID_WAVE, fp);
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92
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93 // Format chunk
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94 fput32le(WAV_ID_FMT, fp);
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95 fput32le(fmt_chunk_size, fp);
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96 fput16le(use_waveex ? WAV_ID_FORMAT_EXTENSIBLE : fmt, fp);
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97 fput16le(ao_data.channels, fp);
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98 fput32le(ao_data.samplerate, fp);
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99 fput32le(ao_data.bps, fp);
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100 fput16le(ao_data.channels * (bits / 8), fp);
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101 fput16le(bits, fp);
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102
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103 if (use_waveex) {
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104 // Extension chunk
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105 fput16le(22, fp);
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106 fput16le(bits, fp);
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107 switch (ao_data.channels) {
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108 case 5:
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109 fput32le(0x0607, fp); // L R C Lb Rb
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110 break;
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111 case 6:
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112 fput32le(0x060f, fp); // L R C Lb Rb LFE
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113 break;
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114 case 7:
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115 fput32le(0x0727, fp); // L R C Cb Ls Rs LFE
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116 break;
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117 case 8:
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118 fput32le(0x063f, fp); // L R C Lb Rb Ls Rs LFE
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119 break;
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120 }
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121 // 2 bytes format + 14 bytes guid
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122 fput32le(fmt, fp);
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123 fput32le(0x00100000, fp);
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124 fput32le(0xAA000080, fp);
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125 fput32le(0x719B3800, fp);
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126 }
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127
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128 // Data chunk
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129 fput32le(WAV_ID_DATA, fp);
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130 fput32le(data_length, fp);
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131 }
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132
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133 // to set/get/query special features/parameters
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134 static int control(int cmd,void *arg){
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135 return -1;
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136 }
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137
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138 // open & setup audio device
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139 // return: 1=success 0=fail
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140 static int init(int rate,int channels,int format,int flags){
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141 const opt_t subopts[] = {
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142 {"waveheader", OPT_ARG_BOOL, &ao_pcm_waveheader, NULL},
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143 {"file", OPT_ARG_MSTRZ, &ao_outputfilename, NULL},
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144 {"fast", OPT_ARG_BOOL, &fast, NULL},
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145 {NULL}
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146 };
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147 // set defaults
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148 ao_pcm_waveheader = 1;
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149
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150 if (subopt_parse(ao_subdevice, subopts) != 0) {
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151 return 0;
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152 }
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153 if (!ao_outputfilename){
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154 ao_outputfilename =
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155 strdup(ao_pcm_waveheader?"audiodump.wav":"audiodump.pcm");
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156 }
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157
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158 if (ao_pcm_waveheader)
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159 {
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160 // WAV files must have one of the following formats
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161
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162 switch(format){
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163 case AF_FORMAT_U8:
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164 case AF_FORMAT_S16_LE:
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165 case AF_FORMAT_S24_LE:
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166 case AF_FORMAT_S32_LE:
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167 case AF_FORMAT_FLOAT_LE:
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168 case AF_FORMAT_AC3_BE:
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169 case AF_FORMAT_AC3_LE:
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170 break;
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171 default:
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172 format = AF_FORMAT_S16_LE;
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173 break;
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174 }
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175 }
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176
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177 ao_data.outburst = 65536;
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178 ao_data.buffersize= 2*65536;
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179 ao_data.channels=channels;
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180 ao_data.samplerate=rate;
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181 ao_data.format=format;
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182 ao_data.bps=channels*rate*(af_fmt2bits(format)/8);
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183
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184 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename,
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185 (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate,
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186 (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
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187 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_HintInfo);
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188
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189 fp = fopen(ao_outputfilename, "wb");
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190 if(fp) {
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191 if(ao_pcm_waveheader){ /* Reserve space for wave header */
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192 write_wave_header(fp, 0x7ffff000);
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193 }
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194 return 1;
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195 }
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196 mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_PCM_CantOpenOutputFile,
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197 ao_outputfilename);
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198 return 0;
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199 }
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200
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201 // close audio device
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202 static void uninit(int immed){
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203
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204 if(ao_pcm_waveheader){ /* Rewrite wave header */
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205 int broken_seek = 0;
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206 #ifdef __MINGW32__
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207 // Windows, in its usual idiocy "emulates" seeks on pipes so it always looks
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208 // like they work. So we have to detect them brute-force.
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209 broken_seek = GetFileType((HANDLE)_get_osfhandle(_fileno(fp))) != FILE_TYPE_DISK;
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210 #endif
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211 if (broken_seek || fseek(fp, 0, SEEK_SET) != 0)
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212 mp_msg(MSGT_AO, MSGL_ERR, "Could not seek to start, WAV size headers not updated!\n");
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213 else if (data_length > 0x7ffff000)
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214 mp_msg(MSGT_AO, MSGL_ERR, "File larger than allowed for WAV files, may play truncated!\n");
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215 else {
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216 write_wave_header(fp, data_length);
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217 }
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218 }
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219 fclose(fp);
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220 if (ao_outputfilename)
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221 free(ao_outputfilename);
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222 ao_outputfilename = NULL;
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223 }
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224
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225 // stop playing and empty buffers (for seeking/pause)
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226 static void reset(void){
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227
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228 }
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229
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230 // stop playing, keep buffers (for pause)
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231 static void audio_pause(void)
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232 {
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233 // for now, just call reset();
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234 reset();
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235 }
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236
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237 // resume playing, after audio_pause()
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238 static void audio_resume(void)
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239 {
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240 }
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241
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242 // return: how many bytes can be played without blocking
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243 static int get_space(void){
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244
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245 if(vo_pts)
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246 return ao_data.pts < vo_pts + fast * 30000 ? ao_data.outburst : 0;
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247 return ao_data.outburst;
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248 }
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249
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250 // plays 'len' bytes of 'data'
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251 // it should round it down to outburst*n
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252 // return: number of bytes played
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253 static int play(void* data,int len,int flags){
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254
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255 // let libaf to do the conversion...
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256 #if 0
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257 //#if HAVE_BIGENDIAN
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258 if (ao_data.format == AFMT_S16_LE) {
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259 unsigned short *buffer = (unsigned short *) data;
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260 register int i;
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261 for(i = 0; i < len/2; ++i) {
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262 buffer[i] = le2me_16(buffer[i]);
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263 }
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264 }
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265 #endif
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266
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267 if (ao_data.channels == 5 || ao_data.channels == 6 || ao_data.channels == 8) {
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268 int frame_size = af_fmt2bits(ao_data.format) / 8;
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269 len -= len % (frame_size * ao_data.channels);
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270 reorder_channel_nch(data, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
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271 AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
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272 ao_data.channels,
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273 len / frame_size, frame_size);
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274 }
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275
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276 //printf("PCM: Writing chunk!\n");
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277 fwrite(data,len,1,fp);
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278
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279 if(ao_pcm_waveheader)
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280 data_length += len;
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281
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282 return len;
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283 }
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284
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285 // return: delay in seconds between first and last sample in buffer
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286 static float get_delay(void){
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287
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288 return 0.0;
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289 }