Mercurial > mplayer.hg
annotate libao2/ao_pcm.c @ 13548:661718721c92
introducing 24bit formats and make the values compliant to OSS
author | alex |
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date | Mon, 04 Oct 2004 19:05:16 +0000 |
parents | c1955840883d |
children | a92101a7eb49 |
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 "afmt.h" |
1107 | 9 #include "audio_out.h" |
10 #include "audio_out_internal.h" | |
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11 #include "../mp_msg.h" |
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12 #include "../help_mp.h" |
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13 |
1107 | 14 |
15 static ao_info_t info = | |
16 { | |
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17 "RAW PCM/WAVE file writer audio output", |
1107 | 18 "pcm", |
19 "Atmosfear", | |
20 "" | |
21 }; | |
22 | |
23 LIBAO_EXTERN(pcm) | |
24 | |
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25 extern int vo_pts; |
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26 |
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27 char *ao_outputfilename = NULL; |
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28 int ao_pcm_waveheader = 1; |
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29 |
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30 #define WAV_ID_RIFF 0x46464952 /* "RIFF" */ |
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31 #define WAV_ID_WAVE 0x45564157 /* "WAVE" */ |
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32 #define WAV_ID_FMT 0x20746d66 /* "fmt " */ |
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33 #define WAV_ID_DATA 0x61746164 /* "data" */ |
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34 #define WAV_ID_PCM 0x0001 |
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35 |
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36 struct WaveHeader |
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37 { |
11270 | 38 uint32_t riff; |
39 uint32_t file_length; | |
40 uint32_t wave; | |
41 uint32_t fmt; | |
42 uint32_t fmt_length; | |
43 uint16_t fmt_tag; | |
44 uint16_t channels; | |
45 uint32_t sample_rate; | |
46 uint32_t bytes_per_second; | |
47 uint16_t block_align; | |
48 uint16_t bits; | |
49 uint32_t data; | |
50 uint32_t data_length; | |
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51 }; |
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52 |
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53 /* init with default values */ |
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54 static struct WaveHeader wavhdr = { |
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55 le2me_32(WAV_ID_RIFF), |
9026 | 56 /* same conventions than in sox/wav.c/wavwritehdr() */ |
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57 0, //le2me_32(0x7ffff024), |
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58 le2me_32(WAV_ID_WAVE), |
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59 le2me_32(WAV_ID_FMT), |
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60 le2me_32(16), |
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61 le2me_16(WAV_ID_PCM), |
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62 le2me_16(2), |
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63 le2me_32(44100), |
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64 le2me_32(192000), |
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65 le2me_16(4), |
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66 le2me_16(16), |
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67 le2me_32(WAV_ID_DATA), |
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68 0, //le2me_32(0x7ffff000) |
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69 }; |
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70 |
1107 | 71 static FILE *fp = NULL; |
72 | |
73 // to set/get/query special features/parameters | |
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74 static int control(int cmd,void *arg){ |
1107 | 75 return -1; |
76 } | |
77 | |
78 // open & setup audio device | |
79 // return: 1=success 0=fail | |
80 static int init(int rate,int channels,int format,int flags){ | |
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81 int bits; |
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82 if(!ao_outputfilename) { |
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83 ao_outputfilename = strdup(ao_pcm_waveheader ? "audiodump.wav" : "audiodump.pcm"); |
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84 } |
7658 | 85 |
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86 /* bits is only equal to format if (format == 8) or (format == 16); |
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87 this means that the following "if" is a kludge and should |
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88 really be a switch to be correct in all cases */ |
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89 |
7658 | 90 bits=8; |
91 switch(format){ | |
92 case AFMT_S8: | |
93 format=AFMT_U8; | |
94 case AFMT_U8: | |
95 break; | |
96 default: | |
97 format=AFMT_S16_LE; | |
98 bits=16; | |
99 break; | |
100 } | |
101 | |
102 ao_data.outburst = 65536; | |
103 ao_data.buffersize= 2*65536; | |
104 ao_data.channels=channels; | |
105 ao_data.samplerate=rate; | |
106 ao_data.format=format; | |
107 ao_data.bps=channels*rate*(bits/8); | |
108 | |
109 wavhdr.channels = le2me_16(ao_data.channels); | |
110 wavhdr.sample_rate = le2me_32(ao_data.samplerate); | |
111 wavhdr.bytes_per_second = le2me_32(ao_data.bps); | |
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112 wavhdr.bits = le2me_16(bits); |
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113 |
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114 wavhdr.data_length=le2me_32(0x7ffff000); |
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115 wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8; |
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116 |
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117 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename, |
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118 (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate, |
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119 (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format)); |
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120 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_HintInfo); |
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121 |
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122 fp = fopen(ao_outputfilename, "wb"); |
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123 if(fp) { |
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124 if(ao_pcm_waveheader){ /* Reserve space for wave header */ |
6501 | 125 fwrite(&wavhdr,sizeof(wavhdr),1,fp); |
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126 wavhdr.file_length=wavhdr.data_length=0; |
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127 } |
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128 return 1; |
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129 } |
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130 mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_PCM_CantOpenOutputFile, |
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131 ao_outputfilename); |
1107 | 132 return 0; |
133 } | |
134 | |
135 // close audio device | |
12145 | 136 static void uninit(int immed){ |
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137 |
6501 | 138 if(ao_pcm_waveheader && fseek(fp, 0, SEEK_SET) == 0){ /* Write wave header */ |
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139 wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8; |
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140 wavhdr.file_length = le2me_32(wavhdr.file_length); |
6762 | 141 wavhdr.data_length = le2me_32(wavhdr.data_length); |
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142 fwrite(&wavhdr,sizeof(wavhdr),1,fp); |
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143 } |
1107 | 144 fclose(fp); |
145 } | |
146 | |
147 // stop playing and empty buffers (for seeking/pause) | |
148 static void reset(){ | |
149 | |
150 } | |
151 | |
152 // stop playing, keep buffers (for pause) | |
153 static void audio_pause() | |
154 { | |
155 // for now, just call reset(); | |
156 reset(); | |
157 } | |
158 | |
159 // resume playing, after audio_pause() | |
160 static void audio_resume() | |
161 { | |
162 } | |
163 | |
164 // return: how many bytes can be played without blocking | |
165 static int get_space(){ | |
166 | |
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167 if(vo_pts) |
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168 return ao_data.pts < vo_pts ? ao_data.outburst : 0; |
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169 return ao_data.outburst; |
1107 | 170 } |
171 | |
172 // plays 'len' bytes of 'data' | |
173 // it should round it down to outburst*n | |
174 // return: number of bytes played | |
175 static int play(void* data,int len,int flags){ | |
176 | |
7658 | 177 // let libaf to do the conversion... |
178 #if 0 | |
179 //#ifdef WORDS_BIGENDIAN | |
180 if (ao_data.format == AFMT_S16_LE) { | |
181 unsigned short *buffer = (unsigned short *) data; | |
182 register int i; | |
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183 for(i = 0; i < len/2; ++i) { |
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184 buffer[i] = le2me_16(buffer[i]); |
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185 } |
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186 } |
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187 #endif |
7658 | 188 |
1107 | 189 //printf("PCM: Writing chunk!\n"); |
190 fwrite(data,len,1,fp); | |
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191 |
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192 if(ao_pcm_waveheader) |
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193 wavhdr.data_length += len; |
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194 |
1107 | 195 return len; |
196 } | |
197 | |
3095 | 198 // return: delay in seconds between first and last sample in buffer |
199 static float get_delay(){ | |
1107 | 200 |
3095 | 201 return 0.0; |
1107 | 202 } |
203 | |
204 | |
205 | |
206 | |
207 | |
208 |