#include "asterisk.h"
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include <errno.h>
#include <stdio.h>
#include "asterisk/lock.h"
#include "asterisk/frame.h"
#include "asterisk/logger.h"
#include "asterisk/options.h"
#include "asterisk/channel.h"
#include "asterisk/cli.h"
#include "asterisk/term.h"
#include "asterisk/utils.h"
#include "asterisk/threadstorage.h"
#include "asterisk/linkedlists.h"
#include "asterisk/translate.h"
#include "asterisk/dsp.h"
Include dependency graph for frame.c:

Go to the source code of this file.
Data Structures | |
| struct | ast_codec_alias_table |
| struct | ast_frame_cache |
| struct | ast_smoother |
Defines | |
| #define | FRAME_CACHE_MAX_SIZE 10 |
| Maximum ast_frame cache size. | |
| #define | SMOOTHER_SIZE 8000 |
| #define | TYPE_MASK 0x3 |
Enumerations | |
| enum | frame_type { TYPE_HIGH, TYPE_LOW, TYPE_SILENCE, TYPE_DONTSEND } |
Functions | |
| int | __ast_smoother_feed (struct ast_smoother *s, struct ast_frame *f, int swap) |
| char * | ast_codec2str (int codec) |
| Get a name from a format Gets a name from a format. | |
| int | ast_codec_choose (struct ast_codec_pref *pref, int formats, int find_best) |
| Select the best audio format according to preference list from supplied options. If "find_best" is non-zero then if nothing is found, the "Best" format of the format list is selected, otherwise 0 is returned. | |
| int | ast_codec_get_len (int format, int samples) |
| Returns the number of bytes for the number of samples of the given format. | |
| int | ast_codec_get_samples (struct ast_frame *f) |
| Returns the number of samples contained in the frame. | |
| int | ast_codec_pref_append (struct ast_codec_pref *pref, int format) |
| Append a audio codec to a preference list, removing it first if it was already there. | |
| void | ast_codec_pref_convert (struct ast_codec_pref *pref, char *buf, size_t size, int right) |
| Shift an audio codec preference list up or down 65 bytes so that it becomes an ASCII string. | |
| ast_format_list | ast_codec_pref_getsize (struct ast_codec_pref *pref, int format) |
| Get packet size for codec. | |
| int | ast_codec_pref_index (struct ast_codec_pref *pref, int index) |
| Codec located at a particular place in the preference index See Audio Codec Preferences. | |
| void | ast_codec_pref_prepend (struct ast_codec_pref *pref, int format, int only_if_existing) |
| Prepend an audio codec to a preference list, removing it first if it was already there. | |
| void | ast_codec_pref_remove (struct ast_codec_pref *pref, int format) |
| Remove audio a codec from a preference list. | |
| int | ast_codec_pref_setsize (struct ast_codec_pref *pref, int format, int framems) |
| Set packet size for codec. | |
| int | ast_codec_pref_string (struct ast_codec_pref *pref, char *buf, size_t size) |
| Dump audio codec preference list into a string. | |
| static const char * | ast_expand_codec_alias (const char *in) |
| int | ast_frame_adjust_volume (struct ast_frame *f, int adjustment) |
| Adjusts the volume of the audio samples contained in a frame. | |
| void | ast_frame_dump (const char *name, struct ast_frame *f, char *prefix) |
| void | ast_frame_free (struct ast_frame *fr, int cache) |
| Requests a frame to be allocated Frees a frame. | |
| static struct ast_frame * | ast_frame_header_new (void) |
| int | ast_frame_slinear_sum (struct ast_frame *f1, struct ast_frame *f2) |
| Sums two frames of audio samples. | |
| ast_frame * | ast_frdup (const struct ast_frame *f) |
| Copies a frame. | |
| ast_frame * | ast_frisolate (struct ast_frame *fr) |
| Makes a frame independent of any static storage. | |
| ast_format_list * | ast_get_format_list (size_t *size) |
| ast_format_list * | ast_get_format_list_index (int index) |
| int | ast_getformatbyname (const char *name) |
| Gets a format from a name. | |
| char * | ast_getformatname (int format) |
| Get the name of a format. | |
| char * | ast_getformatname_multiple (char *buf, size_t size, int format) |
| Get the names of a set of formats. | |
| AST_LIST_HEAD_NOLOCK (ast_frames, ast_frame) | |
| This is just so ast_frames, a list head struct for holding a list of ast_frame structures, is defined. | |
| void | ast_parse_allow_disallow (struct ast_codec_pref *pref, int *mask, const char *list, int allowing) |
| Parse an "allow" or "deny" line in a channel or device configuration and update the capabilities mask and pref if provided. Video codecs are not added to codec preference lists, since we can not transcode. | |
| void | ast_smoother_free (struct ast_smoother *s) |
| int | ast_smoother_get_flags (struct ast_smoother *s) |
| ast_smoother * | ast_smoother_new (int size) |
| ast_frame * | ast_smoother_read (struct ast_smoother *s) |
| void | ast_smoother_reset (struct ast_smoother *s, int size) |
| void | ast_smoother_set_flags (struct ast_smoother *s, int flags) |
| int | ast_smoother_test_flag (struct ast_smoother *s, int flag) |
| void | ast_swapcopy_samples (void *dst, const void *src, int samples) |
| AST_THREADSTORAGE_CUSTOM (frame_cache, frame_cache_init, frame_cache_cleanup) | |
| A per-thread cache of frame headers. | |
| static void | frame_cache_cleanup (void *data) |
| static int | g723_len (unsigned char buf) |
| static int | g723_samples (unsigned char *buf, int maxlen) |
| static unsigned char | get_n_bits_at (unsigned char *data, int n, int bit) |
| int | init_framer (void) |
| static int | show_codec_n (int fd, int argc, char *argv[]) |
| static int | show_codec_n_deprecated (int fd, int argc, char *argv[]) |
| static int | show_codecs (int fd, int argc, char *argv[]) |
| static int | show_codecs_deprecated (int fd, int argc, char *argv[]) |
| static int | speex_get_wb_sz_at (unsigned char *data, int len, int bit) |
| static int | speex_samples (unsigned char *data, int len) |
Variables | |
| static struct ast_format_list | AST_FORMAT_LIST [] |
| Definition of supported media formats (codecs). | |
| ast_frame | ast_null_frame = { AST_FRAME_NULL, } |
| static struct ast_cli_entry | cli_show_audio_codecs |
| static struct ast_cli_entry | cli_show_codec |
| static struct ast_cli_entry | cli_show_codecs |
| static struct ast_cli_entry | cli_show_image_codecs |
| static struct ast_cli_entry | cli_show_video_codecs |
| static char | frame_show_codec_n_usage [] |
| static char | frame_show_codecs_usage [] |
| static struct ast_cli_entry | my_clis [] |
Definition in file frame.c.
| #define FRAME_CACHE_MAX_SIZE 10 |
Maximum ast_frame cache size.
In most cases where the frame header cache will be useful, the size of the cache will stay very small. However, it is not always the case that the same thread that allocates the frame will be the one freeing them, so sometimes a thread will never have any frames in its cache, or the cache will never be pulled from. For the latter case, we limit the maximum size.
Definition at line 69 of file frame.c.
Referenced by ast_frame_free(), and iax_frame_free().
| #define SMOOTHER_SIZE 8000 |
| #define TYPE_MASK 0x3 |
| enum frame_type |
Definition at line 83 of file frame.c.
00083 { 00084 TYPE_HIGH, /* 0x0 */ 00085 TYPE_LOW, /* 0x1 */ 00086 TYPE_SILENCE, /* 0x2 */ 00087 TYPE_DONTSEND /* 0x3 */ 00088 };
| int __ast_smoother_feed | ( | struct ast_smoother * | s, | |
| struct ast_frame * | f, | |||
| int | swap | |||
| ) |
Definition at line 172 of file frame.c.
References AST_FRAME_VOICE, ast_log(), AST_MIN_OFFSET, AST_SMOOTHER_FLAG_G729, ast_swapcopy_samples(), f, LOG_NOTICE, LOG_WARNING, s, and SMOOTHER_SIZE.
00173 { 00174 if (f->frametype != AST_FRAME_VOICE) { 00175 ast_log(LOG_WARNING, "Huh? Can't smooth a non-voice frame!\n"); 00176 return -1; 00177 } 00178 if (!s->format) { 00179 s->format = f->subclass; 00180 s->samplesperbyte = (float)f->samples / (float)f->datalen; 00181 } else if (s->format != f->subclass) { 00182 ast_log(LOG_WARNING, "Smoother was working on %d format frames, now trying to feed %d?\n", s->format, f->subclass); 00183 return -1; 00184 } 00185 if (s->len + f->datalen > SMOOTHER_SIZE) { 00186 ast_log(LOG_WARNING, "Out of smoother space\n"); 00187 return -1; 00188 } 00189 if (((f->datalen == s->size) || ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) 00190 && !s->opt && (f->offset >= AST_MIN_OFFSET)) { 00191 if (!s->len) { 00192 /* Optimize by sending the frame we just got 00193 on the next read, thus eliminating the douple 00194 copy */ 00195 if (swap) 00196 ast_swapcopy_samples(f->data, f->data, f->samples); 00197 s->opt = f; 00198 return 0; 00199 } 00200 } 00201 if (s->flags & AST_SMOOTHER_FLAG_G729) { 00202 if (s->len % 10) { 00203 ast_log(LOG_NOTICE, "Dropping extra frame of G.729 since we already have a VAD frame at the end\n"); 00204 return 0; 00205 } 00206 } 00207 if (swap) 00208 ast_swapcopy_samples(s->data+s->len, f->data, f->samples); 00209 else 00210 memcpy(s->data + s->len, f->data, f->datalen); 00211 /* If either side is empty, reset the delivery time */ 00212 if (!s->len || ast_tvzero(f->delivery) || ast_tvzero(s->delivery)) /* XXX really ? */ 00213 s->delivery = f->delivery; 00214 s->len += f->datalen; 00215 return 0; 00216 }
| char* ast_codec2str | ( | int | codec | ) |
Get a name from a format Gets a name from a format.
| codec | codec number (1,2,4,8,16,etc.) |
Definition at line 600 of file frame.c.
References AST_FORMAT_LIST, and desc.
Referenced by moh_alloc(), show_codec_n(), show_codec_n_deprecated(), show_codecs(), and show_codecs_deprecated().
00601 { 00602 int x; 00603 char *ret = "unknown"; 00604 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00605 if(AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == codec) { 00606 ret = AST_FORMAT_LIST[x].desc; 00607 break; 00608 } 00609 } 00610 return ret; 00611 }
| int ast_codec_choose | ( | struct ast_codec_pref * | pref, | |
| int | formats, | |||
| int | find_best | |||
| ) |
Select the best audio format according to preference list from supplied options. If "find_best" is non-zero then if nothing is found, the "Best" format of the format list is selected, otherwise 0 is returned.
Definition at line 1263 of file frame.c.
References ast_best_codec(), AST_FORMAT_AUDIO_MASK, AST_FORMAT_LIST, ast_log(), ast_format_list::bits, LOG_DEBUG, option_debug, and ast_codec_pref::order.
Referenced by __oh323_new(), gtalk_new(), process_sdp(), sip_new(), and socket_process().
01264 { 01265 int x, ret = 0, slot; 01266 01267 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01268 slot = pref->order[x]; 01269 01270 if (!slot) 01271 break; 01272 if (formats & AST_FORMAT_LIST[slot-1].bits) { 01273 ret = AST_FORMAT_LIST[slot-1].bits; 01274 break; 01275 } 01276 } 01277 if(ret & AST_FORMAT_AUDIO_MASK) 01278 return ret; 01279 01280 if (option_debug > 3) 01281 ast_log(LOG_DEBUG, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec"); 01282 01283 return find_best ? ast_best_codec(formats) : 0; 01284 }
| int ast_codec_get_len | ( | int | format, | |
| int | samples | |||
| ) |
Returns the number of bytes for the number of samples of the given format.
Definition at line 1522 of file frame.c.
References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G722, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_ILBC, AST_FORMAT_SLINEAR, AST_FORMAT_ULAW, ast_getformatname(), ast_log(), len, and LOG_WARNING.
Referenced by moh_generate(), and monmp3thread().
01523 { 01524 int len = 0; 01525 01526 /* XXX Still need speex, g723, and lpc10 XXX */ 01527 switch(format) { 01528 case AST_FORMAT_ILBC: 01529 len = (samples / 240) * 50; 01530 break; 01531 case AST_FORMAT_GSM: 01532 len = (samples / 160) * 33; 01533 break; 01534 case AST_FORMAT_G729A: 01535 len = samples / 8; 01536 break; 01537 case AST_FORMAT_SLINEAR: 01538 len = samples * 2; 01539 break; 01540 case AST_FORMAT_ULAW: 01541 case AST_FORMAT_ALAW: 01542 len = samples; 01543 break; 01544 case AST_FORMAT_G722: 01545 case AST_FORMAT_ADPCM: 01546 case AST_FORMAT_G726: 01547 case AST_FORMAT_G726_AAL2: 01548 len = samples / 2; 01549 break; 01550 default: 01551 ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format)); 01552 } 01553 01554 return len; 01555 }
| int ast_codec_get_samples | ( | struct ast_frame * | f | ) |
Returns the number of samples contained in the frame.
Definition at line 1479 of file frame.c.
References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G722, AST_FORMAT_G723_1, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_ILBC, AST_FORMAT_LPC10, AST_FORMAT_SLINEAR, AST_FORMAT_SPEEX, AST_FORMAT_ULAW, ast_getformatname(), ast_log(), f, g723_samples(), LOG_WARNING, and speex_samples().
Referenced by ast_rtp_read(), isAnsweringMachine(), moh_generate(), schedule_delivery(), and socket_process().
01480 { 01481 int samples=0; 01482 switch(f->subclass) { 01483 case AST_FORMAT_SPEEX: 01484 samples = speex_samples(f->data, f->datalen); 01485 break; 01486 case AST_FORMAT_G723_1: 01487 samples = g723_samples(f->data, f->datalen); 01488 break; 01489 case AST_FORMAT_ILBC: 01490 samples = 240 * (f->datalen / 50); 01491 break; 01492 case AST_FORMAT_GSM: 01493 samples = 160 * (f->datalen / 33); 01494 break; 01495 case AST_FORMAT_G729A: 01496 samples = f->datalen * 8; 01497 break; 01498 case AST_FORMAT_SLINEAR: 01499 samples = f->datalen / 2; 01500 break; 01501 case AST_FORMAT_LPC10: 01502 /* assumes that the RTP packet contains one LPC10 frame */ 01503 samples = 22 * 8; 01504 samples += (((char *)(f->data))[7] & 0x1) * 8; 01505 break; 01506 case AST_FORMAT_ULAW: 01507 case AST_FORMAT_ALAW: 01508 samples = f->datalen; 01509 break; 01510 case AST_FORMAT_G722: 01511 case AST_FORMAT_ADPCM: 01512 case AST_FORMAT_G726: 01513 case AST_FORMAT_G726_AAL2: 01514 samples = f->datalen * 2; 01515 break; 01516 default: 01517 ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass)); 01518 } 01519 return samples; 01520 }
| int ast_codec_pref_append | ( | struct ast_codec_pref * | pref, | |
| int | format | |||
| ) |
Append a audio codec to a preference list, removing it first if it was already there.
Definition at line 1122 of file frame.c.
References ast_codec_pref_remove(), AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow().
01123 { 01124 int x, newindex = 0; 01125 01126 ast_codec_pref_remove(pref, format); 01127 01128 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01129 if(AST_FORMAT_LIST[x].bits == format) { 01130 newindex = x + 1; 01131 break; 01132 } 01133 } 01134 01135 if(newindex) { 01136 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01137 if(!pref->order[x]) { 01138 pref->order[x] = newindex; 01139 break; 01140 } 01141 } 01142 } 01143 01144 return x; 01145 }
| void ast_codec_pref_convert | ( | struct ast_codec_pref * | pref, | |
| char * | buf, | |||
| size_t | size, | |||
| int | right | |||
| ) |
Shift an audio codec preference list up or down 65 bytes so that it becomes an ASCII string.
Definition at line 1024 of file frame.c.
References ast_codec_pref::order.
Referenced by check_access(), create_addr(), dump_prefs(), and socket_process().
01025 { 01026 int x, differential = (int) 'A', mem; 01027 char *from, *to; 01028 01029 if(right) { 01030 from = pref->order; 01031 to = buf; 01032 mem = size; 01033 } else { 01034 to = pref->order; 01035 from = buf; 01036 mem = 32; 01037 } 01038 01039 memset(to, 0, mem); 01040 for (x = 0; x < 32 ; x++) { 01041 if(!from[x]) 01042 break; 01043 to[x] = right ? (from[x] + differential) : (from[x] - differential); 01044 } 01045 }
| struct ast_format_list ast_codec_pref_getsize | ( | struct ast_codec_pref * | pref, | |
| int | format | |||
| ) |
Get packet size for codec.
Definition at line 1224 of file frame.c.
References AST_FORMAT_LIST, ast_format_list::bits, fmt, and format.
Referenced by add_codec_to_sdp(), ast_rtp_bridge(), ast_rtp_write(), handle_open_receive_channel_ack_message(), and transmit_connect().
01225 { 01226 int x, index = -1, framems = 0; 01227 struct ast_format_list fmt = {0}; 01228 01229 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01230 if(AST_FORMAT_LIST[x].bits == format) { 01231 fmt = AST_FORMAT_LIST[x]; 01232 index = x; 01233 break; 01234 } 01235 } 01236 01237 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01238 if(pref->order[x] == (index + 1)) { 01239 framems = pref->framing[x]; 01240 break; 01241 } 01242 } 01243 01244 /* size validation */ 01245 if(!framems) 01246 framems = AST_FORMAT_LIST[index].def_ms; 01247 01248 if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */ 01249 framems -= framems % AST_FORMAT_LIST[index].inc_ms; 01250 01251 if(framems < AST_FORMAT_LIST[index].min_ms) 01252 framems = AST_FORMAT_LIST[index].min_ms; 01253 01254 if(framems > AST_FORMAT_LIST[index].max_ms) 01255 framems = AST_FORMAT_LIST[index].max_ms; 01256 01257 fmt.cur_ms = framems; 01258 01259 return fmt; 01260 }
| int ast_codec_pref_index | ( | struct ast_codec_pref * | pref, | |
| int | index | |||
| ) |
Codec located at a particular place in the preference index See Audio Codec Preferences.
Definition at line 1082 of file frame.c.
References AST_FORMAT_LIST, ast_format_list::bits, and ast_codec_pref::order.
Referenced by _sip_show_peer(), add_sdp(), ast_codec_pref_string(), function_iaxpeer(), function_sippeer(), gtalk_invite(), iax2_show_peer(), print_codec_to_cli(), and socket_process().
01083 { 01084 int slot = 0; 01085 01086 01087 if((index >= 0) && (index < sizeof(pref->order))) { 01088 slot = pref->order[index]; 01089 } 01090 01091 return slot ? AST_FORMAT_LIST[slot-1].bits : 0; 01092 }
| void ast_codec_pref_prepend | ( | struct ast_codec_pref * | pref, | |
| int | format, | |||
| int | only_if_existing | |||
| ) |
Prepend an audio codec to a preference list, removing it first if it was already there.
Definition at line 1148 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_codec_pref::framing, and ast_codec_pref::order.
Referenced by create_addr().
01149 { 01150 int x, newindex = 0; 01151 01152 /* First step is to get the codecs "index number" */ 01153 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01154 if (AST_FORMAT_LIST[x].bits == format) { 01155 newindex = x + 1; 01156 break; 01157 } 01158 } 01159 /* Done if its unknown */ 01160 if (!newindex) 01161 return; 01162 01163 /* Now find any existing occurrence, or the end */ 01164 for (x = 0; x < 32; x++) { 01165 if (!pref->order[x] || pref->order[x] == newindex) 01166 break; 01167 } 01168 01169 if (only_if_existing && !pref->order[x]) 01170 return; 01171 01172 /* Move down to make space to insert - either all the way to the end, 01173 or as far as the existing location (which will be overwritten) */ 01174 for (; x > 0; x--) { 01175 pref->order[x] = pref->order[x - 1]; 01176 pref->framing[x] = pref->framing[x - 1]; 01177 } 01178 01179 /* And insert the new entry */ 01180 pref->order[0] = newindex; 01181 pref->framing[0] = 0; /* ? */ 01182 }
| void ast_codec_pref_remove | ( | struct ast_codec_pref * | pref, | |
| int | format | |||
| ) |
Remove audio a codec from a preference list.
Definition at line 1095 of file frame.c.
References AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_codec_pref_append(), and ast_parse_allow_disallow().
01096 { 01097 struct ast_codec_pref oldorder; 01098 int x, y = 0; 01099 int slot; 01100 int size; 01101 01102 if(!pref->order[0]) 01103 return; 01104 01105 memcpy(&oldorder, pref, sizeof(oldorder)); 01106 memset(pref, 0, sizeof(*pref)); 01107 01108 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01109 slot = oldorder.order[x]; 01110 size = oldorder.framing[x]; 01111 if(! slot) 01112 break; 01113 if(AST_FORMAT_LIST[slot-1].bits != format) { 01114 pref->order[y] = slot; 01115 pref->framing[y++] = size; 01116 } 01117 } 01118 01119 }
| int ast_codec_pref_setsize | ( | struct ast_codec_pref * | pref, | |
| int | format, | |||
| int | framems | |||
| ) |
Set packet size for codec.
Definition at line 1185 of file frame.c.
References AST_FORMAT_LIST, ast_codec_pref::framing, and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow(), and process_sdp().
01186 { 01187 int x, index = -1; 01188 01189 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01190 if(AST_FORMAT_LIST[x].bits == format) { 01191 index = x; 01192 break; 01193 } 01194 } 01195 01196 if(index < 0) 01197 return -1; 01198 01199 /* size validation */ 01200 if(!framems) 01201 framems = AST_FORMAT_LIST[index].def_ms; 01202 01203 if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */ 01204 framems -= framems % AST_FORMAT_LIST[index].inc_ms; 01205 01206 if(framems < AST_FORMAT_LIST[index].min_ms) 01207 framems = AST_FORMAT_LIST[index].min_ms; 01208 01209 if(framems > AST_FORMAT_LIST[index].max_ms) 01210 framems = AST_FORMAT_LIST[index].max_ms; 01211 01212 01213 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01214 if(pref->order[x] == (index + 1)) { 01215 pref->framing[x] = framems; 01216 break; 01217 } 01218 } 01219 01220 return x; 01221 }
| int ast_codec_pref_string | ( | struct ast_codec_pref * | pref, | |
| char * | buf, | |||
| size_t | size | |||
| ) |
Dump audio codec preference list into a string.
Definition at line 1047 of file frame.c.
References ast_codec_pref_index(), and ast_getformatname().
Referenced by dump_prefs(), and socket_process().
01048 { 01049 int x, codec; 01050 size_t total_len, slen; 01051 char *formatname; 01052 01053 memset(buf,0,size); 01054 total_len = size; 01055 buf[0] = '('; 01056 total_len--; 01057 for(x = 0; x < 32 ; x++) { 01058 if(total_len <= 0) 01059 break; 01060 if(!(codec = ast_codec_pref_index(pref,x))) 01061 break; 01062 if((formatname = ast_getformatname(codec))) { 01063 slen = strlen(formatname); 01064 if(slen > total_len) 01065 break; 01066 strncat(buf, formatname, total_len - 1); /* safe */ 01067 total_len -= slen; 01068 } 01069 if(total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) { 01070 strncat(buf, "|", total_len - 1); /* safe */ 01071 total_len--; 01072 } 01073 } 01074 if(total_len) { 01075 strncat(buf, ")", total_len - 1); /* safe */ 01076 total_len--; 01077 } 01078 01079 return size - total_len; 01080 }
| static const char* ast_expand_codec_alias | ( | const char * | in | ) | [static] |
Definition at line 571 of file frame.c.
Referenced by ast_getformatbyname().
00572 { 00573 int x; 00574 00575 for (x = 0; x < sizeof(ast_codec_alias_table) / sizeof(ast_codec_alias_table[0]); x++) { 00576 if(!strcmp(in,ast_codec_alias_table[x].alias)) 00577 return ast_codec_alias_table[x].realname; 00578 } 00579 return in; 00580 }
| int ast_frame_adjust_volume | ( | struct ast_frame * | f, | |
| int | adjustment | |||
| ) |
Adjusts the volume of the audio samples contained in a frame.
| f | The frame containing the samples (must be AST_FRAME_VOICE and AST_FORMAT_SLINEAR) | |
| adjustment | The number of dB to adjust up or down. |
Definition at line 1557 of file frame.c.
References AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_slinear_saturated_divide(), ast_slinear_saturated_multiply(), and f.
Referenced by audiohook_read_frame_single(), and conf_run().
01558 { 01559 int count; 01560 short *fdata = f->data; 01561 short adjust_value = abs(adjustment); 01562 01563 if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR)) 01564 return -1; 01565 01566 if (!adjustment) 01567 return 0; 01568 01569 for (count = 0; count < f->samples; count++) { 01570 if (adjustment > 0) { 01571 ast_slinear_saturated_multiply(&fdata[count], &adjust_value); 01572 } else if (adjustment < 0) { 01573 ast_slinear_saturated_divide(&fdata[count], &adjust_value); 01574 } 01575 } 01576 01577 return 0; 01578 }
| void ast_frame_dump | ( | const char * | name, | |
| struct ast_frame * | f, | |||
| char * | prefix | |||
| ) |
Dump a frame for debugging purposes
Definition at line 754 of file frame.c.
References AST_CONTROL_ANSWER, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_FLASH, AST_CONTROL_HANGUP, AST_CONTROL_OFFHOOK, AST_CONTROL_OPTION, AST_CONTROL_RADIO_KEY, AST_CONTROL_RADIO_UNKEY, AST_CONTROL_RING, AST_CONTROL_RINGING, AST_CONTROL_TAKEOFFHOOK, AST_CONTROL_WINK, AST_FRAME_CONTROL, AST_FRAME_DTMF_BEGIN, AST_FRAME_DTMF_END, AST_FRAME_HTML, AST_FRAME_IAX, AST_FRAME_IMAGE, AST_FRAME_MODEM, AST_FRAME_NULL, AST_FRAME_TEXT, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_getformatname(), AST_HTML_BEGIN, AST_HTML_DATA, AST_HTML_END, AST_HTML_LDCOMPLETE, AST_HTML_LINKREJECT, AST_HTML_LINKURL, AST_HTML_NOSUPPORT, AST_HTML_UNLINK, AST_HTML_URL, AST_MODEM_T38, AST_MODEM_V150, ast_strlen_zero(), ast_verbose(), COLOR_BLACK, COLOR_BRCYAN, COLOR_BRGREEN, COLOR_BRMAGENTA, COLOR_BRRED, COLOR_YELLOW, f, and term_color().
Referenced by __ast_read(), and ast_write().
00755 { 00756 const char noname[] = "unknown"; 00757 char ftype[40] = "Unknown Frametype"; 00758 char cft[80]; 00759 char subclass[40] = "Unknown Subclass"; 00760 char csub[80]; 00761 char moreinfo[40] = ""; 00762 char cn[60]; 00763 char cp[40]; 00764 char cmn[40]; 00765 00766 if (!name) 00767 name = noname; 00768 00769 00770 if (!f) { 00771 ast_verbose("%s [ %s (NULL) ] [%s]\n", 00772 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00773 term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00774 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00775 return; 00776 } 00777 /* XXX We should probably print one each of voice and video when the format changes XXX */ 00778 if (f->frametype == AST_FRAME_VOICE) 00779 return; 00780 if (f->frametype == AST_FRAME_VIDEO) 00781 return; 00782 switch(f->frametype) { 00783 case AST_FRAME_DTMF_BEGIN: 00784 strcpy(ftype, "DTMF Begin"); 00785 subclass[0] = f->subclass; 00786 subclass[1] = '\0'; 00787 break; 00788 case AST_FRAME_DTMF_END: 00789 strcpy(ftype, "DTMF End"); 00790 subclass[0] = f->subclass; 00791 subclass[1] = '\0'; 00792 break; 00793 case AST_FRAME_CONTROL: 00794 strcpy(ftype, "Control"); 00795 switch(f->subclass) { 00796 case AST_CONTROL_HANGUP: 00797 strcpy(subclass, "Hangup"); 00798 break; 00799 case AST_CONTROL_RING: 00800 strcpy(subclass, "Ring"); 00801 break; 00802 case AST_CONTROL_RINGING: 00803 strcpy(subclass, "Ringing"); 00804 break; 00805 case AST_CONTROL_ANSWER: 00806 strcpy(subclass, "Answer"); 00807 break; 00808 case AST_CONTROL_BUSY: 00809 strcpy(subclass, "Busy"); 00810 break; 00811 case AST_CONTROL_TAKEOFFHOOK: 00812 strcpy(subclass, "Take Off Hook"); 00813 break; 00814 case AST_CONTROL_OFFHOOK: 00815 strcpy(subclass, "Line Off Hook"); 00816 break; 00817 case AST_CONTROL_CONGESTION: 00818 strcpy(subclass, "Congestion"); 00819 break; 00820 case AST_CONTROL_FLASH: 00821 strcpy(subclass, "Flash"); 00822 break; 00823 case AST_CONTROL_WINK: 00824 strcpy(subclass, "Wink"); 00825 break; 00826 case AST_CONTROL_OPTION: 00827 strcpy(subclass, "Option"); 00828 break; 00829 case AST_CONTROL_RADIO_KEY: 00830 strcpy(subclass, "Key Radio"); 00831 break; 00832 case AST_CONTROL_RADIO_UNKEY: 00833 strcpy(subclass, "Unkey Radio"); 00834 break; 00835 case -1: 00836 strcpy(subclass, "Stop generators"); 00837 break; 00838 default: 00839 snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass); 00840 } 00841 break; 00842 case AST_FRAME_NULL: 00843 strcpy(ftype, "Null Frame"); 00844 strcpy(subclass, "N/A"); 00845 break; 00846 case AST_FRAME_IAX: 00847 /* Should never happen */ 00848 strcpy(ftype, "IAX Specific"); 00849 snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass); 00850 break; 00851 case AST_FRAME_TEXT: 00852 strcpy(ftype, "Text"); 00853 strcpy(subclass, "N/A"); 00854 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00855 break; 00856 case AST_FRAME_IMAGE: 00857 strcpy(ftype, "Image"); 00858 snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass)); 00859 break; 00860 case AST_FRAME_HTML: 00861 strcpy(ftype, "HTML"); 00862 switch(f->subclass) { 00863 case AST_HTML_URL: 00864 strcpy(subclass, "URL"); 00865 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00866 break; 00867 case AST_HTML_DATA: 00868 strcpy(subclass, "Data"); 00869 break; 00870 case AST_HTML_BEGIN: 00871 strcpy(subclass, "Begin"); 00872 break; 00873 case AST_HTML_END: 00874 strcpy(subclass, "End"); 00875 break; 00876 case AST_HTML_LDCOMPLETE: 00877 strcpy(subclass, "Load Complete"); 00878 break; 00879 case AST_HTML_NOSUPPORT: 00880 strcpy(subclass, "No Support"); 00881 break; 00882