Mercurial > mplayer.hg
annotate libao2/ao_pcm.c @ 25581:c7d66b1e270d
Give a sense to this sentence
author | ben |
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date | Sat, 05 Jan 2008 14:53:13 +0000 |
parents | dfa8a510c81c |
children | 9946e4a6e457 |
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 |
21372 | 7 #include "libavutil/common.h" |
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8 #include "mpbswap.h" |
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9 #include "subopt-helper.h" |
14245 | 10 #include "libaf/af_format.h" |
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11 #include "libaf/reorder_ch.h" |
1107 | 12 #include "audio_out.h" |
13 #include "audio_out_internal.h" | |
14123 | 14 #include "mp_msg.h" |
15 #include "help_mp.h" | |
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16 |
1107 | 17 |
18 static ao_info_t info = | |
19 { | |
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20 "RAW PCM/WAVE file writer audio output", |
1107 | 21 "pcm", |
22 "Atmosfear", | |
23 "" | |
24 }; | |
25 | |
26 LIBAO_EXTERN(pcm) | |
27 | |
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28 extern int vo_pts; |
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29 |
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30 static char *ao_outputfilename = NULL; |
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31 static int ao_pcm_waveheader = 1; |
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32 static int fast = 0; |
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33 |
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34 #define WAV_ID_RIFF 0x46464952 /* "RIFF" */ |
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35 #define WAV_ID_WAVE 0x45564157 /* "WAVE" */ |
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36 #define WAV_ID_FMT 0x20746d66 /* "fmt " */ |
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37 #define WAV_ID_DATA 0x61746164 /* "data" */ |
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38 #define WAV_ID_PCM 0x0001 |
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39 |
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40 struct WaveHeader |
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41 { |
11270 | 42 uint32_t riff; |
43 uint32_t file_length; | |
44 uint32_t wave; | |
45 uint32_t fmt; | |
46 uint32_t fmt_length; | |
47 uint16_t fmt_tag; | |
48 uint16_t channels; | |
49 uint32_t sample_rate; | |
50 uint32_t bytes_per_second; | |
51 uint16_t block_align; | |
52 uint16_t bits; | |
53 uint32_t data; | |
54 uint32_t data_length; | |
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55 }; |
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56 |
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57 /* init with default values */ |
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58 static struct WaveHeader wavhdr; |
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59 |
1107 | 60 static FILE *fp = NULL; |
61 | |
62 // to set/get/query special features/parameters | |
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63 static int control(int cmd,void *arg){ |
1107 | 64 return -1; |
65 } | |
66 | |
67 // open & setup audio device | |
68 // return: 1=success 0=fail | |
69 static int init(int rate,int channels,int format,int flags){ | |
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70 int bits; |
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71 opt_t subopts[] = { |
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72 {"waveheader", OPT_ARG_BOOL, &ao_pcm_waveheader, NULL}, |
14539 | 73 {"file", OPT_ARG_MSTRZ, &ao_outputfilename, NULL}, |
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74 {"fast", OPT_ARG_BOOL, &fast, NULL}, |
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75 {NULL} |
14300 | 76 }; |
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77 // set defaults |
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78 ao_pcm_waveheader = 1; |
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79 |
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80 if (subopt_parse(ao_subdevice, subopts) != 0) { |
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81 return 0; |
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82 } |
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83 if (!ao_outputfilename){ |
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84 ao_outputfilename = |
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85 strdup(ao_pcm_waveheader?"audiodump.wav":"audiodump.pcm"); |
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86 } |
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87 |
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88 /* bits is only equal to format if (format == 8) or (format == 16); |
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89 this means that the following "if" is a kludge and should |
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90 really be a switch to be correct in all cases */ |
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91 |
7658 | 92 bits=8; |
93 switch(format){ | |
14245 | 94 case AF_FORMAT_S8: |
95 format=AF_FORMAT_U8; | |
96 case AF_FORMAT_U8: | |
7658 | 97 break; |
25097 | 98 case AF_FORMAT_AC3: |
99 bits=16; | |
100 break; | |
7658 | 101 default: |
14245 | 102 format=AF_FORMAT_S16_LE; |
7658 | 103 bits=16; |
104 break; | |
105 } | |
106 | |
107 ao_data.outburst = 65536; | |
108 ao_data.buffersize= 2*65536; | |
109 ao_data.channels=channels; | |
110 ao_data.samplerate=rate; | |
111 ao_data.format=format; | |
112 ao_data.bps=channels*rate*(bits/8); | |
113 | |
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114 wavhdr.riff = le2me_32(WAV_ID_RIFF); |
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115 wavhdr.wave = le2me_32(WAV_ID_WAVE); |
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116 wavhdr.fmt = le2me_32(WAV_ID_FMT); |
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117 wavhdr.fmt_length = le2me_32(16); |
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118 wavhdr.fmt_tag = le2me_16(WAV_ID_PCM); |
7658 | 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 = le2me_32(WAV_ID_DATA); |
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126 wavhdr.data_length=le2me_32(0x7ffff000); |
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127 wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8; |
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128 |
14264 | 129 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename, |
130 (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate, | |
131 (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format)); | |
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132 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_HintInfo); |
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133 |
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134 fp = fopen(ao_outputfilename, "wb"); |
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135 if(fp) { |
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136 if(ao_pcm_waveheader){ /* Reserve space for wave header */ |
6501 | 137 fwrite(&wavhdr,sizeof(wavhdr),1,fp); |
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138 wavhdr.file_length=wavhdr.data_length=0; |
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139 } |
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140 return 1; |
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141 } |
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142 mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_PCM_CantOpenOutputFile, |
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143 ao_outputfilename); |
1107 | 144 return 0; |
145 } | |
146 | |
147 // close audio device | |
12145 | 148 static void uninit(int immed){ |
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149 |
6501 | 150 if(ao_pcm_waveheader && fseek(fp, 0, SEEK_SET) == 0){ /* Write wave header */ |
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151 wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8; |
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152 wavhdr.file_length = le2me_32(wavhdr.file_length); |
6762 | 153 wavhdr.data_length = le2me_32(wavhdr.data_length); |
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154 fwrite(&wavhdr,sizeof(wavhdr),1,fp); |
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155 } |
1107 | 156 fclose(fp); |
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157 if (ao_outputfilename) |
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158 free(ao_outputfilename); |
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159 ao_outputfilename = NULL; |
1107 | 160 } |
161 | |
162 // stop playing and empty buffers (for seeking/pause) | |
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163 static void reset(void){ |
1107 | 164 |
165 } | |
166 | |
167 // stop playing, keep buffers (for pause) | |
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168 static void audio_pause(void) |
1107 | 169 { |
170 // for now, just call reset(); | |
171 reset(); | |
172 } | |
173 | |
174 // resume playing, after audio_pause() | |
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175 static void audio_resume(void) |
1107 | 176 { |
177 } | |
178 | |
179 // return: how many bytes can be played without blocking | |
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180 static int get_space(void){ |
1107 | 181 |
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182 if(vo_pts) |
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183 return ao_data.pts < vo_pts + fast * 30000 ? ao_data.outburst : 0; |
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184 return ao_data.outburst; |
1107 | 185 } |
186 | |
187 // plays 'len' bytes of 'data' | |
188 // it should round it down to outburst*n | |
189 // return: number of bytes played | |
190 static int play(void* data,int len,int flags){ | |
191 | |
7658 | 192 // let libaf to do the conversion... |
193 #if 0 | |
194 //#ifdef WORDS_BIGENDIAN | |
195 if (ao_data.format == AFMT_S16_LE) { | |
196 unsigned short *buffer = (unsigned short *) data; | |
197 register int i; | |
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198 for(i = 0; i < len/2; ++i) { |
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199 buffer[i] = le2me_16(buffer[i]); |
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200 } |
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201 } |
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202 #endif |
7658 | 203 |
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204 if (ao_data.channels == 6 || ao_data.channels == 5) { |
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205 int frame_size = le2me_16(wavhdr.bits) / 8; |
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206 len -= len % (frame_size * ao_data.channels); |
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207 reorder_channel_nch(data, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT, |
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208 AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT, |
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209 ao_data.channels, |
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210 len / frame_size, frame_size); |
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211 } |
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212 |
1107 | 213 //printf("PCM: Writing chunk!\n"); |
214 fwrite(data,len,1,fp); | |
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215 |
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Added support for writing wave files and specifying filename to write to.
atmosfear
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216 if(ao_pcm_waveheader) |
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Added support for writing wave files and specifying filename to write to.
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217 wavhdr.data_length += len; |
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Added support for writing wave files and specifying filename to write to.
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218 |
1107 | 219 return len; |
220 } | |
221 | |
3095 | 222 // return: delay in seconds between first and last sample in buffer |
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Patch by Stefan Huehner / stefan % huehner ! org \
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223 static float get_delay(void){ |
1107 | 224 |
3095 | 225 return 0.0; |
1107 | 226 } |
227 | |
228 | |
229 | |
230 | |
231 | |
232 |