Mercurial > mplayer.hg
annotate libao2/ao_pcm.c @ 4512:f52e38e03bb7
YUY2 BGR32 BGR24 support for vo_x11 if swScaler is used
author | michael |
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date | Sun, 03 Feb 2002 14:41:31 +0000 |
parents | 981a9e5118ce |
children | de4074ab4e5f |
rev | line source |
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1107 | 1 #include <stdio.h> |
2 #include <stdlib.h> | |
3 | |
4 #include "audio_out.h" | |
5 #include "audio_out_internal.h" | |
6 | |
7 static ao_info_t info = | |
8 { | |
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9 "RAW PCM/WAVE file writer audio output", |
1107 | 10 "pcm", |
11 "Atmosfear", | |
12 "" | |
13 }; | |
14 | |
15 LIBAO_EXTERN(pcm) | |
16 | |
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17 char *ao_outputfilename = NULL; |
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18 int ao_pcm_waveheader = 1; |
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19 |
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20 #define WAV_ID_RIFF 0x46464952 /* "RIFF" */ |
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21 #define WAV_ID_WAVE 0x45564157 /* "WAVE" */ |
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22 #define WAV_ID_FMT 0x20746d66 /* "fmt " */ |
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23 #define WAV_ID_DATA 0x61746164 /* "data" */ |
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24 #define WAV_ID_PCM 0x0001 |
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25 |
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26 struct WaveHeader |
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27 { |
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28 unsigned long riff; |
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29 unsigned long file_length; |
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30 unsigned long wave; |
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31 unsigned long fmt; |
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32 unsigned long fmt_length; |
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33 short fmt_tag; |
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34 short channels; |
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35 unsigned long sample_rate; |
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36 unsigned long bytes_per_second; |
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37 short block_align; |
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38 short bits; |
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39 unsigned long data; |
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40 unsigned long data_length; |
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41 }; |
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42 |
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43 |
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44 static struct WaveHeader wavhdr = { |
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45 WAV_ID_RIFF, |
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46 0x00000000, |
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47 WAV_ID_WAVE, |
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48 WAV_ID_FMT, |
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49 16, |
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50 WAV_ID_PCM, |
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51 2, |
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52 44100, |
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53 192000, |
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54 4, |
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55 16, |
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56 WAV_ID_DATA, |
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57 0x00000000 |
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58 }; |
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59 |
1107 | 60 static FILE *fp = NULL; |
61 | |
62 // to set/get/query special features/parameters | |
63 static int control(int cmd,int arg){ | |
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 if(!ao_outputfilename) { |
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71 ao_outputfilename = (char *) malloc(sizeof(char) * 14); |
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72 strcpy(ao_outputfilename,(ao_pcm_waveheader ? "audiodump.wav" : "audiodump.pcm")); |
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73 } |
1107 | 74 |
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75 wavhdr.channels = channels; |
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76 wavhdr.sample_rate = rate; |
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77 wavhdr.bytes_per_second = rate * (format / 8) * channels; |
1113 | 78 wavhdr.bits = format; |
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79 |
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80 printf("PCM: File: %s (%s) Samplerate: %iHz Channels: %s Format %s\n", ao_outputfilename, (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate, (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format)); |
1107 | 81 printf("PCM: Info - fastest dumping is achieved with -vo null -hardframedrop.\n"); |
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82 printf("PCM: Info - to write WAVE files use -waveheader (default), for RAW PCM -nowaveheader.\n"); |
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83 fp = fopen(ao_outputfilename, "wb"); |
1107 | 84 |
3095 | 85 ao_data.outburst = 4096; |
1107 | 86 |
87 | |
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88 if(fp) { |
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89 if(ao_pcm_waveheader) /* Reserve space for wave header */ |
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90 fseek(fp, sizeof(wavhdr), SEEK_SET); |
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91 return 1; |
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92 } |
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93 printf("PCM: Failed to open %s for writing!\n", ao_outputfilename); |
1107 | 94 return 0; |
95 } | |
96 | |
97 // close audio device | |
98 static void uninit(){ | |
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99 |
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100 if(ao_pcm_waveheader){ /* Write wave header */ |
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101 wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr); |
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102 fseek(fp, 0, SEEK_SET); |
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103 fwrite(&wavhdr,sizeof(wavhdr),1,fp); |
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104 } |
1107 | 105 fclose(fp); |
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106 free(ao_outputfilename); |
1107 | 107 } |
108 | |
109 // stop playing and empty buffers (for seeking/pause) | |
110 static void reset(){ | |
111 | |
112 } | |
113 | |
114 // stop playing, keep buffers (for pause) | |
115 static void audio_pause() | |
116 { | |
117 // for now, just call reset(); | |
118 reset(); | |
119 } | |
120 | |
121 // resume playing, after audio_pause() | |
122 static void audio_resume() | |
123 { | |
124 } | |
125 | |
126 // return: how many bytes can be played without blocking | |
127 static int get_space(){ | |
128 | |
3095 | 129 return ao_data.outburst; |
1107 | 130 } |
131 | |
132 // plays 'len' bytes of 'data' | |
133 // it should round it down to outburst*n | |
134 // return: number of bytes played | |
135 static int play(void* data,int len,int flags){ | |
136 | |
137 //printf("PCM: Writing chunk!\n"); | |
138 fwrite(data,len,1,fp); | |
139 | |
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140 if(ao_pcm_waveheader) |
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141 wavhdr.data_length += len; |
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142 |
1107 | 143 return len; |
144 } | |
145 | |
3095 | 146 // return: delay in seconds between first and last sample in buffer |
147 static float get_delay(){ | |
1107 | 148 |
3095 | 149 return 0.0; |
1107 | 150 } |
151 | |
152 | |
153 | |
154 | |
155 | |
156 |