Mercurial > mplayer.hg
annotate libao2/ao_pcm.c @ 18488:b23c0e341e3e
Move setting the output filename after the suboption parsing, otherwise it
cannot take the suboptions that were set into account.
author | diego |
---|---|
date | Sat, 13 May 2006 19:04:39 +0000 |
parents | 96fdfbad5b1a |
children | 9533f21d3f7e |
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 static int fast = 0; |
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31 |
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32 #define WAV_ID_RIFF 0x46464952 /* "RIFF" */ |
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33 #define WAV_ID_WAVE 0x45564157 /* "WAVE" */ |
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34 #define WAV_ID_FMT 0x20746d66 /* "fmt " */ |
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35 #define WAV_ID_DATA 0x61746164 /* "data" */ |
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36 #define WAV_ID_PCM 0x0001 |
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37 |
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38 struct WaveHeader |
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39 { |
11270 | 40 uint32_t riff; |
41 uint32_t file_length; | |
42 uint32_t wave; | |
43 uint32_t fmt; | |
44 uint32_t fmt_length; | |
45 uint16_t fmt_tag; | |
46 uint16_t channels; | |
47 uint32_t sample_rate; | |
48 uint32_t bytes_per_second; | |
49 uint16_t block_align; | |
50 uint16_t bits; | |
51 uint32_t data; | |
52 uint32_t data_length; | |
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53 }; |
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54 |
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55 /* init with default values */ |
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56 static struct WaveHeader wavhdr = { |
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57 le2me_32(WAV_ID_RIFF), |
9026 | 58 /* same conventions than in sox/wav.c/wavwritehdr() */ |
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59 0, //le2me_32(0x7ffff024), |
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60 le2me_32(WAV_ID_WAVE), |
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61 le2me_32(WAV_ID_FMT), |
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62 le2me_32(16), |
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63 le2me_16(WAV_ID_PCM), |
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64 le2me_16(2), |
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65 le2me_32(44100), |
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66 le2me_32(192000), |
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67 le2me_16(4), |
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68 le2me_16(16), |
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69 le2me_32(WAV_ID_DATA), |
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70 0, //le2me_32(0x7ffff000) |
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71 }; |
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72 |
1107 | 73 static FILE *fp = NULL; |
74 | |
75 // to set/get/query special features/parameters | |
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76 static int control(int cmd,void *arg){ |
1107 | 77 return -1; |
78 } | |
79 | |
80 // open & setup audio device | |
81 // return: 1=success 0=fail | |
82 static int init(int rate,int channels,int format,int flags){ | |
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83 int bits; |
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84 opt_t subopts[] = { |
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85 {"waveheader", OPT_ARG_BOOL, &ao_pcm_waveheader, NULL}, |
14539 | 86 {"file", OPT_ARG_MSTRZ, &ao_outputfilename, NULL}, |
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87 {"fast", OPT_ARG_BOOL, &fast, NULL}, |
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88 {NULL} |
14300 | 89 }; |
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90 // set defaults |
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91 ao_pcm_waveheader = 1; |
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92 |
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93 if (subopt_parse(ao_subdevice, subopts) != 0) { |
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94 return 0; |
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95 } |
7658 | 96 |
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97 ao_outputfilename = |
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98 strdup((ao_pcm_waveheader)?"audiodump.wav":"audiodump.pcm"); |
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99 |
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100 /* bits is only equal to format if (format == 8) or (format == 16); |
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101 this means that the following "if" is a kludge and should |
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102 really be a switch to be correct in all cases */ |
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103 |
7658 | 104 bits=8; |
105 switch(format){ | |
14245 | 106 case AF_FORMAT_S8: |
107 format=AF_FORMAT_U8; | |
108 case AF_FORMAT_U8: | |
7658 | 109 break; |
110 default: | |
14245 | 111 format=AF_FORMAT_S16_LE; |
7658 | 112 bits=16; |
113 break; | |
114 } | |
115 | |
116 ao_data.outburst = 65536; | |
117 ao_data.buffersize= 2*65536; | |
118 ao_data.channels=channels; | |
119 ao_data.samplerate=rate; | |
120 ao_data.format=format; | |
121 ao_data.bps=channels*rate*(bits/8); | |
122 | |
123 wavhdr.channels = le2me_16(ao_data.channels); | |
124 wavhdr.sample_rate = le2me_32(ao_data.samplerate); | |
125 wavhdr.bytes_per_second = le2me_32(ao_data.bps); | |
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126 wavhdr.bits = le2me_16(bits); |
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127 wavhdr.block_align = le2me_16(ao_data.channels * (bits / 8)); |
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128 |
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129 wavhdr.data_length=le2me_32(0x7ffff000); |
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130 wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8; |
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131 |
14264 | 132 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename, |
133 (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate, | |
134 (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format)); | |
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135 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_HintInfo); |
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136 |
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137 fp = fopen(ao_outputfilename, "wb"); |
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138 if(fp) { |
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139 if(ao_pcm_waveheader){ /* Reserve space for wave header */ |
6501 | 140 fwrite(&wavhdr,sizeof(wavhdr),1,fp); |
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141 wavhdr.file_length=wavhdr.data_length=0; |
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142 } |
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143 return 1; |
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144 } |
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145 mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_PCM_CantOpenOutputFile, |
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146 ao_outputfilename); |
1107 | 147 return 0; |
148 } | |
149 | |
150 // close audio device | |
12145 | 151 static void uninit(int immed){ |
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152 |
6501 | 153 if(ao_pcm_waveheader && fseek(fp, 0, SEEK_SET) == 0){ /* Write wave header */ |
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154 wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8; |
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155 wavhdr.file_length = le2me_32(wavhdr.file_length); |
6762 | 156 wavhdr.data_length = le2me_32(wavhdr.data_length); |
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157 fwrite(&wavhdr,sizeof(wavhdr),1,fp); |
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158 } |
1107 | 159 fclose(fp); |
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160 if (ao_outputfilename) |
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161 free(ao_outputfilename); |
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162 ao_outputfilename = NULL; |
1107 | 163 } |
164 | |
165 // stop playing and empty buffers (for seeking/pause) | |
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166 static void reset(void){ |
1107 | 167 |
168 } | |
169 | |
170 // stop playing, keep buffers (for pause) | |
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171 static void audio_pause(void) |
1107 | 172 { |
173 // for now, just call reset(); | |
174 reset(); | |
175 } | |
176 | |
177 // resume playing, after audio_pause() | |
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178 static void audio_resume(void) |
1107 | 179 { |
180 } | |
181 | |
182 // return: how many bytes can be played without blocking | |
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183 static int get_space(void){ |
1107 | 184 |
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185 if(vo_pts) |
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186 return ao_data.pts < vo_pts + fast * 30000 ? ao_data.outburst : 0; |
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187 return ao_data.outburst; |
1107 | 188 } |
189 | |
190 // plays 'len' bytes of 'data' | |
191 // it should round it down to outburst*n | |
192 // return: number of bytes played | |
193 static int play(void* data,int len,int flags){ | |
194 | |
7658 | 195 // let libaf to do the conversion... |
196 #if 0 | |
197 //#ifdef WORDS_BIGENDIAN | |
198 if (ao_data.format == AFMT_S16_LE) { | |
199 unsigned short *buffer = (unsigned short *) data; | |
200 register int i; | |
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201 for(i = 0; i < len/2; ++i) { |
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202 buffer[i] = le2me_16(buffer[i]); |
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203 } |
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204 } |
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205 #endif |
7658 | 206 |
1107 | 207 //printf("PCM: Writing chunk!\n"); |
208 fwrite(data,len,1,fp); | |
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209 |
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210 if(ao_pcm_waveheader) |
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211 wavhdr.data_length += len; |
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212 |
1107 | 213 return len; |
214 } | |
215 | |
3095 | 216 // return: delay in seconds between first and last sample in buffer |
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Patch by Stefan Huehner / stefan % huehner ! org \
rathann
parents:
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changeset
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217 static float get_delay(void){ |
1107 | 218 |
3095 | 219 return 0.0; |
1107 | 220 } |
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