Mercurial > mplayer.hg
annotate libao2/ao_pcm.c @ 17997:7bb7f54b4cfa
increasing sync tag to 1.15 after update by Diego
author | gabrov |
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date | Thu, 30 Mar 2006 14:50:16 +0000 |
parents | f580a7755ac5 |
children | 96fdfbad5b1a |
rev | line source |
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6237 | 1 #include "config.h" |
2 | |
1107 | 3 #include <stdio.h> |
4 #include <stdlib.h> | |
6237 | 5 #include <string.h> |
1107 | 6 |
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7 #include "bswap.h" |
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8 #include "subopt-helper.h" |
14245 | 9 #include "libaf/af_format.h" |
1107 | 10 #include "audio_out.h" |
11 #include "audio_out_internal.h" | |
14123 | 12 #include "mp_msg.h" |
13 #include "help_mp.h" | |
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14 |
1107 | 15 |
16 static ao_info_t info = | |
17 { | |
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18 "RAW PCM/WAVE file writer audio output", |
1107 | 19 "pcm", |
20 "Atmosfear", | |
21 "" | |
22 }; | |
23 | |
24 LIBAO_EXTERN(pcm) | |
25 | |
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26 extern int vo_pts; |
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27 |
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28 static char *ao_outputfilename = NULL; |
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29 static int ao_pcm_waveheader = 1; |
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30 |
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31 #define WAV_ID_RIFF 0x46464952 /* "RIFF" */ |
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32 #define WAV_ID_WAVE 0x45564157 /* "WAVE" */ |
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33 #define WAV_ID_FMT 0x20746d66 /* "fmt " */ |
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34 #define WAV_ID_DATA 0x61746164 /* "data" */ |
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35 #define WAV_ID_PCM 0x0001 |
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36 |
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37 struct WaveHeader |
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38 { |
11270 | 39 uint32_t riff; |
40 uint32_t file_length; | |
41 uint32_t wave; | |
42 uint32_t fmt; | |
43 uint32_t fmt_length; | |
44 uint16_t fmt_tag; | |
45 uint16_t channels; | |
46 uint32_t sample_rate; | |
47 uint32_t bytes_per_second; | |
48 uint16_t block_align; | |
49 uint16_t bits; | |
50 uint32_t data; | |
51 uint32_t data_length; | |
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52 }; |
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53 |
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54 /* init with default values */ |
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55 static struct WaveHeader wavhdr = { |
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56 le2me_32(WAV_ID_RIFF), |
9026 | 57 /* same conventions than in sox/wav.c/wavwritehdr() */ |
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58 0, //le2me_32(0x7ffff024), |
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59 le2me_32(WAV_ID_WAVE), |
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60 le2me_32(WAV_ID_FMT), |
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61 le2me_32(16), |
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62 le2me_16(WAV_ID_PCM), |
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63 le2me_16(2), |
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64 le2me_32(44100), |
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65 le2me_32(192000), |
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66 le2me_16(4), |
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67 le2me_16(16), |
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68 le2me_32(WAV_ID_DATA), |
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69 0, //le2me_32(0x7ffff000) |
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70 }; |
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71 |
1107 | 72 static FILE *fp = NULL; |
73 | |
74 // to set/get/query special features/parameters | |
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75 static int control(int cmd,void *arg){ |
1107 | 76 return -1; |
77 } | |
78 | |
79 // open & setup audio device | |
80 // return: 1=success 0=fail | |
81 static int init(int rate,int channels,int format,int flags){ | |
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82 int bits; |
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83 opt_t subopts[] = { |
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84 {"waveheader", OPT_ARG_BOOL, &ao_pcm_waveheader, NULL}, |
14539 | 85 {"file", OPT_ARG_MSTRZ, &ao_outputfilename, NULL}, |
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86 {NULL} |
14300 | 87 }; |
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88 // set defaults |
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89 ao_pcm_waveheader = 1; |
14539 | 90 ao_outputfilename = |
91 strdup((ao_pcm_waveheader)?"audiodump.wav":"audiodump.pcm"); | |
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92 if (subopt_parse(ao_subdevice, subopts) != 0) { |
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93 return 0; |
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94 } |
7658 | 95 |
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96 /* bits is only equal to format if (format == 8) or (format == 16); |
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97 this means that the following "if" is a kludge and should |
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98 really be a switch to be correct in all cases */ |
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99 |
7658 | 100 bits=8; |
101 switch(format){ | |
14245 | 102 case AF_FORMAT_S8: |
103 format=AF_FORMAT_U8; | |
104 case AF_FORMAT_U8: | |
7658 | 105 break; |
106 default: | |
14245 | 107 format=AF_FORMAT_S16_LE; |
7658 | 108 bits=16; |
109 break; | |
110 } | |
111 | |
112 ao_data.outburst = 65536; | |
113 ao_data.buffersize= 2*65536; | |
114 ao_data.channels=channels; | |
115 ao_data.samplerate=rate; | |
116 ao_data.format=format; | |
117 ao_data.bps=channels*rate*(bits/8); | |
118 | |
119 wavhdr.channels = le2me_16(ao_data.channels); | |
120 wavhdr.sample_rate = le2me_32(ao_data.samplerate); | |
121 wavhdr.bytes_per_second = le2me_32(ao_data.bps); | |
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122 wavhdr.bits = le2me_16(bits); |
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123 wavhdr.block_align = le2me_16(ao_data.channels * (bits / 8)); |
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124 |
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125 wavhdr.data_length=le2me_32(0x7ffff000); |
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126 wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8; |
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127 |
14264 | 128 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename, |
129 (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate, | |
130 (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format)); | |
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131 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_HintInfo); |
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132 |
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133 fp = fopen(ao_outputfilename, "wb"); |
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134 if(fp) { |
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135 if(ao_pcm_waveheader){ /* Reserve space for wave header */ |
6501 | 136 fwrite(&wavhdr,sizeof(wavhdr),1,fp); |
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137 wavhdr.file_length=wavhdr.data_length=0; |
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138 } |
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139 return 1; |
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140 } |
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141 mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_PCM_CantOpenOutputFile, |
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142 ao_outputfilename); |
1107 | 143 return 0; |
144 } | |
145 | |
146 // close audio device | |
12145 | 147 static void uninit(int immed){ |
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148 |
6501 | 149 if(ao_pcm_waveheader && fseek(fp, 0, SEEK_SET) == 0){ /* Write wave header */ |
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150 wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8; |
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151 wavhdr.file_length = le2me_32(wavhdr.file_length); |
6762 | 152 wavhdr.data_length = le2me_32(wavhdr.data_length); |
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153 fwrite(&wavhdr,sizeof(wavhdr),1,fp); |
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154 } |
1107 | 155 fclose(fp); |
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156 if (ao_outputfilename) |
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157 free(ao_outputfilename); |
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158 ao_outputfilename = NULL; |
1107 | 159 } |
160 | |
161 // stop playing and empty buffers (for seeking/pause) | |
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162 static void reset(void){ |
1107 | 163 |
164 } | |
165 | |
166 // stop playing, keep buffers (for pause) | |
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167 static void audio_pause(void) |
1107 | 168 { |
169 // for now, just call reset(); | |
170 reset(); | |
171 } | |
172 | |
173 // resume playing, after audio_pause() | |
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174 static void audio_resume(void) |
1107 | 175 { |
176 } | |
177 | |
178 // return: how many bytes can be played without blocking | |
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179 static int get_space(void){ |
1107 | 180 |
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181 if(vo_pts) |
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182 return ao_data.pts < vo_pts ? ao_data.outburst : 0; |
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183 return ao_data.outburst; |
1107 | 184 } |
185 | |
186 // plays 'len' bytes of 'data' | |
187 // it should round it down to outburst*n | |
188 // return: number of bytes played | |
189 static int play(void* data,int len,int flags){ | |
190 | |
7658 | 191 // let libaf to do the conversion... |
192 #if 0 | |
193 //#ifdef WORDS_BIGENDIAN | |
194 if (ao_data.format == AFMT_S16_LE) { | |
195 unsigned short *buffer = (unsigned short *) data; | |
196 register int i; | |
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197 for(i = 0; i < len/2; ++i) { |
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198 buffer[i] = le2me_16(buffer[i]); |
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199 } |
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200 } |
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201 #endif |
7658 | 202 |
1107 | 203 //printf("PCM: Writing chunk!\n"); |
204 fwrite(data,len,1,fp); | |
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205 |
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206 if(ao_pcm_waveheader) |
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207 wavhdr.data_length += len; |
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208 |
1107 | 209 return len; |
210 } | |
211 | |
3095 | 212 // return: delay in seconds between first and last sample in buffer |
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213 static float get_delay(void){ |
1107 | 214 |
3095 | 215 return 0.0; |
1107 | 216 } |
217 | |
218 | |
219 | |
220 | |
221 | |
222 |