Asterisk - The Open Source Telephony Project  GIT-master-8beac82
format_mp3.c
Go to the documentation of this file.
1 /*
2  * Asterisk -- An open source telephony toolkit.
3  *
4  * Anthony Minessale <anthmct@yahoo.com>
5  *
6  * Derived from other asterisk sound formats by
7  * Mark Spencer <markster@linux-support.net>
8  *
9  * Thanks to mpglib from http://www.mpg123.org/
10  * and Chris Stenton [jacs@gnome.co.uk]
11  * for coding the ability to play stereo and non-8khz files
12 
13  * See http://www.asterisk.org for more information about
14  * the Asterisk project. Please do not directly contact
15  * any of the maintainers of this project for assistance;
16  * the project provides a web site, mailing lists and IRC
17  * channels for your use.
18  *
19  * This program is free software, distributed under the terms of
20  * the GNU General Public License Version 2. See the LICENSE file
21  * at the top of the source tree.
22  */
23 
24 /*!
25  * \file
26  * \brief MP3 Format Handler
27  * \ingroup formats
28  */
29 
30 /*** MODULEINFO
31  <defaultenabled>no</defaultenabled>
32  <support_level>extended</support_level>
33  ***/
34 
35 #include "asterisk.h"
36 
37 #include "mp3/mpg123.h"
38 #include "mp3/mpglib.h"
39 
40 #include "asterisk/module.h"
41 #include "asterisk/mod_format.h"
42 #include "asterisk/logger.h"
43 #include "asterisk/format_cache.h"
44 
45 #define MP3_BUFLEN 320
46 #define MP3_SCACHE 16384
47 #define MP3_DCACHE 8192
48 
49 struct mp3_private {
50  /*! state for the mp3 decoder */
51  struct mpstr mp;
52  /*! buffer to hold mp3 data after read from disk */
54  /*! buffer for slinear audio after being decoded out of sbuf */
56  /*! how much data has been written to the output buffer in the ast_filestream */
57  int buflen;
58  /*! how much data has been written to sbuf */
59  int sbuflen;
60  /*! how much data is left to be read out of dbuf, starting at dbufoffset */
61  int dbuflen;
62  /*! current offset for reading data out of dbuf */
64  int offset;
65  long seek;
66 };
67 
68 static const char name[] = "mp3";
69 
70 #define BLOCKSIZE 160
71 #define OUTSCALE 4096
72 
73 #define GAIN -4 /* 2^GAIN is the multiple to increase the volume by */
74 
75 #if __BYTE_ORDER == __LITTLE_ENDIAN
76 #define htoll(b) (b)
77 #define htols(b) (b)
78 #define ltohl(b) (b)
79 #define ltohs(b) (b)
80 #else
81 #if __BYTE_ORDER == __BIG_ENDIAN
82 #define htoll(b) \
83  (((((b) ) & 0xFF) << 24) | \
84  ((((b) >> 8) & 0xFF) << 16) | \
85  ((((b) >> 16) & 0xFF) << 8) | \
86  ((((b) >> 24) & 0xFF) ))
87 #define htols(b) \
88  (((((b) ) & 0xFF) << 8) | \
89  ((((b) >> 8) & 0xFF) ))
90 #define ltohl(b) htoll(b)
91 #define ltohs(b) htols(b)
92 #else
93 #error "Endianess not defined"
94 #endif
95 #endif
96 
97 
98 static int mp3_open(struct ast_filestream *s)
99 {
100  struct mp3_private *p = s->_private;
101  InitMP3(&p->mp, OUTSCALE);
102  return 0;
103 }
104 
105 
106 static void mp3_close(struct ast_filestream *s)
107 {
108  struct mp3_private *p = s->_private;
109 
110  ExitMP3(&p->mp);
111  return;
112 }
113 
114 static int mp3_squeue(struct ast_filestream *s)
115 {
116  struct mp3_private *p = s->_private;
117  int res=0;
118 
119  res = ftell(s->f);
120  p->sbuflen = fread(p->sbuf, 1, MP3_SCACHE, s->f);
121  if (p->sbuflen < MP3_SCACHE) {
122  if (ferror(s->f)) {
123  ast_log(LOG_WARNING, "Error while reading MP3 file: %s\n", strerror(errno));
124  return -1;
125  }
126  }
127  res = decodeMP3(&p->mp,p->sbuf,p->sbuflen,p->dbuf,MP3_DCACHE,&p->dbuflen);
128  if(res != MP3_OK)
129  return -1;
130  p->sbuflen -= p->dbuflen;
131  p->dbufoffset = 0;
132  return 0;
133 }
134 
135 static int mp3_dqueue(struct ast_filestream *s)
136 {
137  struct mp3_private *p = s->_private;
138  int res=0;
139 
140  if((res = decodeMP3(&p->mp,NULL,0,p->dbuf,MP3_DCACHE,&p->dbuflen)) == MP3_OK) {
141  p->sbuflen -= p->dbuflen;
142  p->dbufoffset = 0;
143  }
144  return res;
145 }
146 
147 static int mp3_queue(struct ast_filestream *s)
148 {
149  struct mp3_private *p = s->_private;
150  int res = 0, bytes = 0;
151 
152  if(p->seek) {
153  ExitMP3(&p->mp);
154  InitMP3(&p->mp, OUTSCALE);
155  fseek(s->f, 0, SEEK_SET);
156  p->sbuflen = p->dbuflen = p->offset = 0;
157  while(p->offset < p->seek) {
158  if(mp3_squeue(s))
159  return -1;
160  while(p->offset < p->seek && ((res = mp3_dqueue(s))) == MP3_OK) {
161  for(bytes = 0 ; bytes < p->dbuflen ; bytes++) {
162  p->dbufoffset++;
163  p->offset++;
164  if(p->offset >= p->seek)
165  break;
166  }
167  }
168  if(res == MP3_ERR)
169  return -1;
170  }
171 
172  p->seek = 0;
173  return 0;
174  }
175  if(p->dbuflen == 0) {
176  if(p->sbuflen) {
177  res = mp3_dqueue(s);
178  if(res == MP3_ERR)
179  return -1;
180  }
181  if(! p->sbuflen || res != MP3_OK) {
182  if(mp3_squeue(s))
183  return -1;
184  }
185 
186  }
187 
188  return 0;
189 }
190 
191 static struct ast_frame *mp3_read(struct ast_filestream *s, int *whennext)
192 {
193 
194  struct mp3_private *p = s->_private;
195  int delay =0;
196  int save=0;
197 
198  /* Pre-populate the buffer that holds audio to be returned (dbuf) */
199  if (mp3_queue(s)) {
200  return NULL;
201  }
202 
203  if (p->dbuflen) {
204  /* Read out what's waiting in dbuf */
205  for (p->buflen = 0; p->buflen < MP3_BUFLEN && p->buflen < p->dbuflen; p->buflen++) {
206  s->buf[p->buflen + AST_FRIENDLY_OFFSET] = p->dbuf[p->buflen + p->dbufoffset];
207  }
208  p->dbufoffset += p->buflen;
209  p->dbuflen -= p->buflen;
210  }
211 
212  if (p->buflen < MP3_BUFLEN) {
213  /* dbuf didn't have enough, so reset dbuf, fill it back up and continue */
214  p->dbuflen = p->dbufoffset = 0;
215 
216  if (mp3_queue(s)) {
217  return NULL;
218  }
219 
220  /* Make sure dbuf has enough to complete this read attempt */
221  if (p->dbuflen >= (MP3_BUFLEN - p->buflen)) {
222  for (save = p->buflen; p->buflen < MP3_BUFLEN; p->buflen++) {
223  s->buf[p->buflen + AST_FRIENDLY_OFFSET] = p->dbuf[(p->buflen - save) + p->dbufoffset];
224  }
225  p->dbufoffset += (MP3_BUFLEN - save);
226  p->dbuflen -= (MP3_BUFLEN - save);
227  }
228 
229  }
230 
231  p->offset += p->buflen;
232  delay = p->buflen / 2;
234  s->fr.samples = delay;
235  *whennext = delay;
236  return &s->fr;
237 }
238 
239 
240 static int mp3_write(struct ast_filestream *fs, struct ast_frame *f)
241 {
242  ast_log(LOG_ERROR,"I Can't write MP3 only read them.\n");
243  return -1;
244 
245 }
246 
247 
248 static int mp3_seek(struct ast_filestream *s, off_t sample_offset, int whence)
249 {
250  struct mp3_private *p = s->_private;
251  off_t min,max,cur;
252  long offset=0,samples;
253  samples = sample_offset * 2;
254 
255  min = 0;
256  fseek(s->f, 0, SEEK_END);
257  max = ftell(s->f) * 100;
258  cur = p->offset;
259 
260  if (whence == SEEK_SET)
261  offset = samples + min;
262  else if (whence == SEEK_CUR || whence == SEEK_FORCECUR)
263  offset = samples + cur;
264  else if (whence == SEEK_END)
265  offset = max - samples;
266  if (whence != SEEK_FORCECUR) {
267  offset = (offset > max)?max:offset;
268  }
269 
270  p->seek = offset;
271  return fseek(s->f, offset, SEEK_SET);
272 
273 }
274 
275 static int mp3_rewrite(struct ast_filestream *s, const char *comment)
276 {
277  ast_log(LOG_ERROR,"I Can't write MP3 only read them.\n");
278  return -1;
279 }
280 
281 static int mp3_trunc(struct ast_filestream *s)
282 {
283 
284  ast_log(LOG_ERROR,"I Can't write MP3 only read them.\n");
285  return -1;
286 }
287 
288 static off_t mp3_tell(struct ast_filestream *s)
289 {
290  struct mp3_private *p = s->_private;
291 
292  return p->offset/2;
293 }
294 
295 static char *mp3_getcomment(struct ast_filestream *s)
296 {
297  return NULL;
298 }
299 
300 static struct ast_format_def mp3_f = {
301  .name = "mp3",
302  .exts = "mp3",
303  .open = mp3_open,
304  .write = mp3_write,
305  .rewrite = mp3_rewrite,
306  .seek = mp3_seek,
307  .trunc = mp3_trunc,
308  .tell = mp3_tell,
309  .read = mp3_read,
310  .close = mp3_close,
311  .getcomment = mp3_getcomment,
312  .buf_size = MP3_BUFLEN + AST_FRIENDLY_OFFSET,
313  .desc_size = sizeof(struct mp3_private),
314 };
315 
316 
317 static int load_module(void)
318 {
319  mp3_f.format = ast_format_slin;
320  InitMP3Constants();
321  return ast_format_def_register(&mp3_f);
322 }
323 
324 static int unload_module(void)
325 {
327 }
328 
329 AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "MP3 format [Any rate but 8000hz mono is optimal]");
static int mp3_seek(struct ast_filestream *s, off_t sample_offset, int whence)
Definition: format_mp3.c:248
static const char name[]
Definition: format_mp3.c:68
Asterisk main include file. File version handling, generic pbx functions.
struct mpstr mp
Definition: format_mp3.c:51
#define MP3_BUFLEN
Definition: format_mp3.c:45
AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "MP3 format [Any rate but 8000hz mono is optimal]")
static int mp3_rewrite(struct ast_filestream *s, const char *comment)
Definition: format_mp3.c:275
#define OUTSCALE
Definition: format_mp3.c:71
#define LOG_WARNING
Definition: logger.h:274
int dbufoffset
Definition: format_mp3.c:63
#define MP3_DCACHE
Definition: format_mp3.c:47
static struct ast_frame * mp3_read(struct ast_filestream *s, int *whennext)
Definition: format_mp3.c:191
char dbuf[MP3_DCACHE]
Definition: format_mp3.c:55
static int mp3_squeue(struct ast_filestream *s)
Definition: format_mp3.c:114
static int unload_module(void)
Definition: format_mp3.c:324
static struct ast_format_def mp3_f
Definition: format_mp3.c:300
static int load_module(void)
Definition: format_mp3.c:317
#define NULL
Definition: resample.c:96
Each supported file format is described by the following structure.
Definition: mod_format.h:43
Header for providers of file and format handling routines. Clients of these routines should include "...
static void mp3_close(struct ast_filestream *s)
Definition: format_mp3.c:106
int ast_format_def_unregister(const char *name)
Unregisters a file format.
Definition: file.c:162
#define ast_log
Definition: astobj2.c:42
static char * mp3_getcomment(struct ast_filestream *s)
Definition: format_mp3.c:295
#define AST_FRIENDLY_OFFSET
Offset into a frame&#39;s data buffer.
#define ast_format_def_register(f)
Definition: mod_format.h:136
struct ast_frame fr
frame produced by read, typically
Definition: mod_format.h:122
struct ast_format * format
Definition: mod_format.h:48
static int mp3_dqueue(struct ast_filestream *s)
Definition: format_mp3.c:135
void * _private
Definition: mod_format.h:124
#define LOG_ERROR
Definition: logger.h:285
#define AST_FRAME_SET_BUFFER(fr, _base, _ofs, _datalen)
char sbuf[MP3_SCACHE]
Definition: format_mp3.c:53
int errno
#define comment
Definition: ael_lex.c:976
char name[80]
Definition: mod_format.h:44
static int mp3_open(struct ast_filestream *s)
Definition: format_mp3.c:98
static off_t mp3_tell(struct ast_filestream *s)
Definition: format_mp3.c:288
#define SEEK_FORCECUR
Definition: file.h:51
Support for logging to various files, console and syslog Configuration in file logger.conf.
static int mp3_write(struct ast_filestream *fs, struct ast_frame *f)
Definition: format_mp3.c:240
static int mp3_trunc(struct ast_filestream *s)
Definition: format_mp3.c:281
This structure is allocated by file.c in one chunk, together with buf_size and desc_size bytes of mem...
Definition: mod_format.h:101
static int mp3_queue(struct ast_filestream *s)
Definition: format_mp3.c:147
#define MP3_SCACHE
Definition: format_mp3.c:46
Data structure associated with a single frame of data.
#define ASTERISK_GPL_KEY
The text the key() function should return.
Definition: module.h:46
struct ast_format * ast_format_slin
Built-in cached signed linear 8kHz format.
Definition: format_cache.c:41
Asterisk module definitions.
#define min(a, b)
Definition: f2c.h:197
Media Format Cache API.
#define max(a, b)
Definition: f2c.h:198