Mercurial > mplayer.hg
comparison stream/audio_in.c @ 19271:64d82a45a05d
introduce new 'stream' directory for all stream layer related components and split them from libmpdemux
author | ben |
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date | Mon, 31 Jul 2006 17:39:17 +0000 |
parents | libmpdemux/audio_in.c@d2d9d011203f |
children | e7c989f7a7c9 |
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19270:7d39b911f0bd | 19271:64d82a45a05d |
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1 #include <stdio.h> | |
2 #include <stdlib.h> | |
3 #include <unistd.h> | |
4 | |
5 #include "config.h" | |
6 | |
7 #include "audio_in.h" | |
8 #include "mp_msg.h" | |
9 #include "help_mp.h" | |
10 #include <string.h> | |
11 #include <errno.h> | |
12 | |
13 // sanitizes ai structure before calling other functions | |
14 int audio_in_init(audio_in_t *ai, int type) | |
15 { | |
16 ai->type = type; | |
17 ai->setup = 0; | |
18 | |
19 ai->channels = -1; | |
20 ai->samplerate = -1; | |
21 ai->blocksize = -1; | |
22 ai->bytes_per_sample = -1; | |
23 ai->samplesize = -1; | |
24 | |
25 switch (ai->type) { | |
26 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) | |
27 case AUDIO_IN_ALSA: | |
28 ai->alsa.handle = NULL; | |
29 ai->alsa.log = NULL; | |
30 ai->alsa.device = strdup("default"); | |
31 return 0; | |
32 #endif | |
33 #ifdef USE_OSS_AUDIO | |
34 case AUDIO_IN_OSS: | |
35 ai->oss.audio_fd = -1; | |
36 ai->oss.device = strdup("/dev/dsp"); | |
37 return 0; | |
38 #endif | |
39 default: | |
40 return -1; | |
41 } | |
42 } | |
43 | |
44 int audio_in_setup(audio_in_t *ai) | |
45 { | |
46 | |
47 switch (ai->type) { | |
48 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) | |
49 case AUDIO_IN_ALSA: | |
50 if (ai_alsa_init(ai) < 0) return -1; | |
51 ai->setup = 1; | |
52 return 0; | |
53 #endif | |
54 #ifdef USE_OSS_AUDIO | |
55 case AUDIO_IN_OSS: | |
56 if (ai_oss_init(ai) < 0) return -1; | |
57 ai->setup = 1; | |
58 return 0; | |
59 #endif | |
60 default: | |
61 return -1; | |
62 } | |
63 } | |
64 | |
65 int audio_in_set_samplerate(audio_in_t *ai, int rate) | |
66 { | |
67 switch (ai->type) { | |
68 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) | |
69 case AUDIO_IN_ALSA: | |
70 ai->req_samplerate = rate; | |
71 if (!ai->setup) return 0; | |
72 if (ai_alsa_setup(ai) < 0) return -1; | |
73 return ai->samplerate; | |
74 #endif | |
75 #ifdef USE_OSS_AUDIO | |
76 case AUDIO_IN_OSS: | |
77 ai->req_samplerate = rate; | |
78 if (!ai->setup) return 0; | |
79 if (ai_oss_set_samplerate(ai) < 0) return -1; | |
80 return ai->samplerate; | |
81 #endif | |
82 default: | |
83 return -1; | |
84 } | |
85 } | |
86 | |
87 int audio_in_set_channels(audio_in_t *ai, int channels) | |
88 { | |
89 switch (ai->type) { | |
90 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) | |
91 case AUDIO_IN_ALSA: | |
92 ai->req_channels = channels; | |
93 if (!ai->setup) return 0; | |
94 if (ai_alsa_setup(ai) < 0) return -1; | |
95 return ai->channels; | |
96 #endif | |
97 #ifdef USE_OSS_AUDIO | |
98 case AUDIO_IN_OSS: | |
99 ai->req_channels = channels; | |
100 if (!ai->setup) return 0; | |
101 if (ai_oss_set_channels(ai) < 0) return -1; | |
102 return ai->channels; | |
103 #endif | |
104 default: | |
105 return -1; | |
106 } | |
107 } | |
108 | |
109 int audio_in_set_device(audio_in_t *ai, char *device) | |
110 { | |
111 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) | |
112 int i; | |
113 #endif | |
114 if (ai->setup) return -1; | |
115 switch (ai->type) { | |
116 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) | |
117 case AUDIO_IN_ALSA: | |
118 if (ai->alsa.device) free(ai->alsa.device); | |
119 ai->alsa.device = strdup(device); | |
120 /* mplayer cannot handle colons in arguments */ | |
121 for (i = 0; i < (int)strlen(ai->alsa.device); i++) { | |
122 if (ai->alsa.device[i] == '.') ai->alsa.device[i] = ':'; | |
123 } | |
124 return 0; | |
125 #endif | |
126 #ifdef USE_OSS_AUDIO | |
127 case AUDIO_IN_OSS: | |
128 if (ai->oss.device) free(ai->oss.device); | |
129 ai->oss.device = strdup(device); | |
130 return 0; | |
131 #endif | |
132 default: | |
133 return -1; | |
134 } | |
135 } | |
136 | |
137 int audio_in_uninit(audio_in_t *ai) | |
138 { | |
139 if (ai->setup) { | |
140 switch (ai->type) { | |
141 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) | |
142 case AUDIO_IN_ALSA: | |
143 if (ai->alsa.log) | |
144 snd_output_close(ai->alsa.log); | |
145 if (ai->alsa.handle) { | |
146 snd_pcm_close(ai->alsa.handle); | |
147 } | |
148 ai->setup = 0; | |
149 return 0; | |
150 #endif | |
151 #ifdef USE_OSS_AUDIO | |
152 case AUDIO_IN_OSS: | |
153 close(ai->oss.audio_fd); | |
154 ai->setup = 0; | |
155 return 0; | |
156 #endif | |
157 } | |
158 } | |
159 return -1; | |
160 } | |
161 | |
162 int audio_in_start_capture(audio_in_t *ai) | |
163 { | |
164 switch (ai->type) { | |
165 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) | |
166 case AUDIO_IN_ALSA: | |
167 return snd_pcm_start(ai->alsa.handle); | |
168 #endif | |
169 #ifdef USE_OSS_AUDIO | |
170 case AUDIO_IN_OSS: | |
171 return 0; | |
172 #endif | |
173 default: | |
174 return -1; | |
175 } | |
176 } | |
177 | |
178 int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer) | |
179 { | |
180 int ret; | |
181 | |
182 switch (ai->type) { | |
183 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) | |
184 case AUDIO_IN_ALSA: | |
185 ret = snd_pcm_readi(ai->alsa.handle, buffer, ai->alsa.chunk_size); | |
186 if (ret != ai->alsa.chunk_size) { | |
187 if (ret < 0) { | |
188 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, snd_strerror(ret)); | |
189 if (ret == -EPIPE) { | |
190 if (ai_alsa_xrun(ai) == 0) { | |
191 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_XRUNSomeFramesMayBeLeftOut); | |
192 } else { | |
193 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrFatalCannotRecover); | |
194 } | |
195 } | |
196 } else { | |
197 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_NotEnoughSamples); | |
198 } | |
199 return -1; | |
200 } | |
201 return ret; | |
202 #endif | |
203 #ifdef USE_OSS_AUDIO | |
204 case AUDIO_IN_OSS: | |
205 ret = read(ai->oss.audio_fd, buffer, ai->blocksize); | |
206 if (ret != ai->blocksize) { | |
207 if (ret < 0) { | |
208 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, strerror(errno)); | |
209 } else { | |
210 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_NotEnoughSamples); | |
211 } | |
212 return -1; | |
213 } | |
214 return ret; | |
215 #endif | |
216 default: | |
217 return -1; | |
218 } | |
219 } |