Mercurial > mplayer.hg
annotate libao2/ao_pcm.c @ 14298:3c818342a02b
Add -ao pcm suboptions and remove -aofile and -waveheader options.
Base on idea by Olivier Rolland (billl at users dot sf dot net)
author | reimar |
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date | Sun, 02 Jan 2005 14:41:47 +0000 |
parents | cb5fbade8a5c |
children | 8f1ab3da482c |
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 strarg_t file; |
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84 opt_t subopts[] = { |
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85 {"waveheader", OPT_ARG_BOOL, &ao_pcm_waveheader, NULL}, |
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86 {"file", OPT_ARG_STR, &file, NULL}, |
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87 {NULL} |
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88 } |
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89 // set defaults |
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90 ao_pcm_waveheader = 1; |
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91 file.str = "audiodump.wav"; |
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92 file.len = 13; |
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93 if (subopt_parse(ao_subdevice, subopts) != 0) { |
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94 return 0; |
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95 } |
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96 ao_outputfilename = strndup(file.str, file.len); |
7658 | 97 |
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98 /* bits is only equal to format if (format == 8) or (format == 16); |
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99 this means that the following "if" is a kludge and should |
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100 really be a switch to be correct in all cases */ |
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101 |
7658 | 102 bits=8; |
103 switch(format){ | |
14245 | 104 case AF_FORMAT_S8: |
105 format=AF_FORMAT_U8; | |
106 case AF_FORMAT_U8: | |
7658 | 107 break; |
108 default: | |
14245 | 109 format=AF_FORMAT_S16_LE; |
7658 | 110 bits=16; |
111 break; | |
112 } | |
113 | |
114 ao_data.outburst = 65536; | |
115 ao_data.buffersize= 2*65536; | |
116 ao_data.channels=channels; | |
117 ao_data.samplerate=rate; | |
118 ao_data.format=format; | |
119 ao_data.bps=channels*rate*(bits/8); | |
120 | |
121 wavhdr.channels = le2me_16(ao_data.channels); | |
122 wavhdr.sample_rate = le2me_32(ao_data.samplerate); | |
123 wavhdr.bytes_per_second = le2me_32(ao_data.bps); | |
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124 wavhdr.bits = le2me_16(bits); |
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125 |
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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) | |
163 static void reset(){ | |
164 | |
165 } | |
166 | |
167 // stop playing, keep buffers (for pause) | |
168 static void audio_pause() | |
169 { | |
170 // for now, just call reset(); | |
171 reset(); | |
172 } | |
173 | |
174 // resume playing, after audio_pause() | |
175 static void audio_resume() | |
176 { | |
177 } | |
178 | |
179 // return: how many bytes can be played without blocking | |
180 static int get_space(){ | |
181 | |
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182 if(vo_pts) |
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183 return ao_data.pts < vo_pts ? 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 |
1107 | 204 //printf("PCM: Writing chunk!\n"); |
205 fwrite(data,len,1,fp); | |
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206 |
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207 if(ao_pcm_waveheader) |
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208 wavhdr.data_length += len; |
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209 |
1107 | 210 return len; |
211 } | |
212 | |
3095 | 213 // return: delay in seconds between first and last sample in buffer |
214 static float get_delay(){ | |
1107 | 215 |
3095 | 216 return 0.0; |
1107 | 217 } |
218 | |
219 | |
220 | |
221 | |
222 | |
223 |