Thu May 14 15:13:32 2009

Asterisk developer's documentation


frame.h File Reference

Asterisk internal frame definitions. More...

#include <sys/types.h>
#include <sys/time.h>
#include "asterisk/compiler.h"
#include "asterisk/endian.h"
#include "asterisk/linkedlists.h"

Go to the source code of this file.

Data Structures

struct  ast_codec_pref
struct  ast_format_list
 Definition of supported media formats (codecs). More...
struct  ast_frame
 Data structure associated with a single frame of data. More...
struct  ast_option_header
struct  oprmode

Defines

#define AST_FORMAT_ADPCM   (1 << 5)
#define AST_FORMAT_ALAW   (1 << 3)
#define AST_FORMAT_AUDIO_MASK   ((1 << 16)-1)
#define AST_FORMAT_AUDIO_UNDEFINED   ((1 << 13) | (1 << 14) | (1 << 15))
#define AST_FORMAT_G722   (1 << 12)
#define AST_FORMAT_G723_1   (1 << 0)
#define AST_FORMAT_G726   (1 << 11)
#define AST_FORMAT_G726_AAL2   (1 << 4)
#define AST_FORMAT_G729A   (1 << 8)
#define AST_FORMAT_GSM   (1 << 1)
#define AST_FORMAT_H261   (1 << 18)
#define AST_FORMAT_H263   (1 << 19)
#define AST_FORMAT_H263_PLUS   (1 << 20)
#define AST_FORMAT_H264   (1 << 21)
#define AST_FORMAT_ILBC   (1 << 10)
#define AST_FORMAT_JPEG   (1 << 16)
#define AST_FORMAT_LPC10   (1 << 7)
#define AST_FORMAT_MAX_AUDIO   (1 << 15)
#define AST_FORMAT_MAX_VIDEO   (1 << 24)
#define AST_FORMAT_PNG   (1 << 17)
#define AST_FORMAT_SLINEAR   (1 << 6)
#define AST_FORMAT_SPEEX   (1 << 9)
#define AST_FORMAT_ULAW   (1 << 2)
#define AST_FORMAT_VIDEO_MASK   (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK))
#define ast_frame_byteswap_be(fr)   do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data, __f->data, __f->samples); } while(0)
#define ast_frame_byteswap_le(fr)   do { ; } while(0)
#define AST_FRAME_DTMF   AST_FRAME_DTMF_END
#define AST_FRAME_SET_BUFFER(fr, _base, _ofs, _datalen)
#define ast_frfree(fr)   ast_frame_free(fr, 1)
#define AST_FRIENDLY_OFFSET   64
#define AST_HTML_BEGIN   4
#define AST_HTML_DATA   2
#define AST_HTML_END   8
#define AST_HTML_LDCOMPLETE   16
#define AST_HTML_LINKREJECT   20
#define AST_HTML_LINKURL   18
#define AST_HTML_NOSUPPORT   17
#define AST_HTML_UNLINK   19
#define AST_HTML_URL   1
#define AST_MALLOCD_DATA   (1 << 1)
#define AST_MALLOCD_HDR   (1 << 0)
#define AST_MALLOCD_SRC   (1 << 2)
#define AST_MIN_OFFSET   32
#define AST_MODEM_T38   1
#define AST_MODEM_V150   2
#define AST_OPTION_AUDIO_MODE   4
#define AST_OPTION_ECHOCAN   8
#define AST_OPTION_FLAG_ACCEPT   1
#define AST_OPTION_FLAG_ANSWER   5
#define AST_OPTION_FLAG_QUERY   4
#define AST_OPTION_FLAG_REJECT   2
#define AST_OPTION_FLAG_REQUEST   0
#define AST_OPTION_FLAG_WTF   6
#define AST_OPTION_OPRMODE   7
#define AST_OPTION_RELAXDTMF   3
#define AST_OPTION_RXGAIN   6
#define AST_OPTION_TDD   2
#define AST_OPTION_TONE_VERIFY   1
#define AST_OPTION_TXGAIN   5
#define ast_smoother_feed(s, f)   __ast_smoother_feed(s, f, 0)
#define ast_smoother_feed_be(s, f)   __ast_smoother_feed(s, f, 1)
#define ast_smoother_feed_le(s, f)   __ast_smoother_feed(s, f, 0)
#define AST_SMOOTHER_FLAG_BE   (1 << 1)
#define AST_SMOOTHER_FLAG_G729   (1 << 0)

Enumerations

enum  { AST_FRFLAG_HAS_TIMING_INFO = (1 << 0), AST_FRFLAG_FROM_TRANSLATOR = (1 << 1), AST_FRFLAG_FROM_DSP = (1 << 2), AST_FRFLAG_FROM_FILESTREAM = (1 << 3) }
enum  ast_control_frame_type {
  AST_CONTROL_HANGUP = 1, AST_CONTROL_RING = 2, AST_CONTROL_RINGING = 3, AST_CONTROL_ANSWER = 4,
  AST_CONTROL_BUSY = 5, AST_CONTROL_TAKEOFFHOOK = 6, AST_CONTROL_OFFHOOK = 7, AST_CONTROL_CONGESTION = 8,
  AST_CONTROL_FLASH = 9, AST_CONTROL_WINK = 10, AST_CONTROL_OPTION = 11, AST_CONTROL_RADIO_KEY = 12,
  AST_CONTROL_RADIO_UNKEY = 13, AST_CONTROL_PROGRESS = 14, AST_CONTROL_PROCEEDING = 15, AST_CONTROL_HOLD = 16,
  AST_CONTROL_UNHOLD = 17, AST_CONTROL_VIDUPDATE = 18, AST_CONTROL_ATXFERCMD = 19, AST_CONTROL_SRCUPDATE = 20
}
enum  ast_frame_type {
  AST_FRAME_DTMF_END = 1, AST_FRAME_VOICE, AST_FRAME_VIDEO, AST_FRAME_CONTROL,
  AST_FRAME_NULL, AST_FRAME_IAX, AST_FRAME_TEXT, AST_FRAME_IMAGE,
  AST_FRAME_HTML, AST_FRAME_CNG, AST_FRAME_MODEM, AST_FRAME_DTMF_BEGIN
}
 Frame types. More...

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.
static int ast_codec_interp_len (int format)
 Gets duration in ms of interpolation frame for a format.
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_init (struct ast_codec_pref *pref)
 Initialize an audio codec preference to "no preference" 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 force_inline int ast_format_rate (int format)
 Get the sample rate for a given format.
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)
ast_frameast_frame_enqueue (struct ast_frame *head, struct ast_frame *f, int maxlen, int dupe)
 Appends a frame to the end of a list of frames, truncating the maximum length of the list.
void ast_frame_free (struct ast_frame *fr, int cache)
 Requests a frame to be allocated Frees a frame.
int ast_frame_slinear_sum (struct ast_frame *f1, struct ast_frame *f2)
 Sums two frames of audio samples.
ast_frameast_frdup (const struct ast_frame *fr)
 Copies a frame.
ast_frameast_frisolate (struct ast_frame *fr)
 Makes a frame independent of any static storage.
ast_format_listast_get_format_list (size_t *size)
ast_format_listast_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.
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 *smoother)
ast_smootherast_smoother_new (int bytes)
ast_frameast_smoother_read (struct ast_smoother *s)
void ast_smoother_reconfigure (struct ast_smoother *s, int bytes)
 Reconfigure an existing smoother to output a different number of bytes per frame.
void ast_smoother_reset (struct ast_smoother *s, int bytes)
void ast_smoother_set_flags (struct ast_smoother *smoother, int flags)
int ast_smoother_test_flag (struct ast_smoother *s, int flag)
void ast_swapcopy_samples (void *dst, const void *src, int samples)

Variables

ast_frame ast_null_frame


Detailed Description

Asterisk internal frame definitions.

Definition in file frame.h.


Define Documentation

#define AST_FORMAT_ADPCM   (1 << 5)

ADPCM (IMA)

Definition at line 248 of file frame.h.

Referenced by adpcmtolin_sample(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), convertcap(), vox_read(), and vox_write().

#define AST_FORMAT_ALAW   (1 << 3)

Raw A-law data (G.711)

Definition at line 244 of file frame.h.

Referenced by alawtolin_sample(), alawtoulaw_sample(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), cb_events(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_new(), dahdi_read(), dahdi_write(), find_transcoders(), is_encoder(), misdn_read(), misdn_set_opt_exec(), oh323_rtp_read(), pcm_seek(), pcm_write(), read_config(), and sms_generate().

#define AST_FORMAT_AUDIO_MASK   ((1 << 16)-1)

Maximum audio mask

Definition at line 268 of file frame.h.

Referenced by add_sdp(), ast_best_codec(), ast_codec_choose(), ast_openstream_full(), ast_parse_allow_disallow(), ast_request(), ast_translate_available_formats(), ast_translator_best_choice(), begin_dial(), func_channel_read(), generator_force(), gtalk_rtp_read(), process_sdp(), set_format(), sip_call(), sip_rtp_read(), and sip_write().

#define AST_FORMAT_AUDIO_UNDEFINED   ((1 << 13) | (1 << 14) | (1 << 15))

Unsupported audio bits

Definition at line 264 of file frame.h.

#define AST_FORMAT_G722   (1 << 12)

G.722

Definition at line 262 of file frame.h.

Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_write(), au_seek(), convertcap(), g722tolin_sample(), and pcm_read().

#define AST_FORMAT_G723_1   (1 << 0)

G.723.1 compression

Definition at line 238 of file frame.h.

Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_samples(), ast_rtp_write(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_destroy(), dahdi_translate(), g723_read(), g723_write(), load_module(), phone_request(), phone_setup(), phone_write(), and register_translator().

#define AST_FORMAT_G726   (1 << 11)

ADPCM (G.726, 32kbps, RFC3551 codeword packing)

Definition at line 260 of file frame.h.

Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_set_rtpmap_type(), g726_read(), g726_write(), and g726tolin_sample().

#define AST_FORMAT_G726_AAL2   (1 << 4)

ADPCM (G.726, 32kbps, AAL2 codeword packing)

Definition at line 246 of file frame.h.

Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_lookup_mime_subtype(), ast_rtp_set_rtpmap_type(), codec_ast2skinny(), and codec_skinny2ast().

#define AST_FORMAT_G729A   (1 << 8)

G.729A audio

Definition at line 254 of file frame.h.

Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_destroy(), dahdi_translate(), g729_read(), and g729_write().

#define AST_FORMAT_GSM   (1 << 1)

GSM compression

Definition at line 240 of file frame.h.

Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), convertcap(), gsm_read(), gsm_write(), gsmtolin_sample(), wav_read(), and wav_write().

#define AST_FORMAT_H261   (1 << 18)

H.261 Video

Definition at line 274 of file frame.h.

Referenced by codec_ast2skinny(), and codec_skinny2ast().

#define AST_FORMAT_H263   (1 << 19)

H.263 Video

Definition at line 276 of file frame.h.

Referenced by codec_ast2skinny(), codec_skinny2ast(), h263_read(), and h263_write().

#define AST_FORMAT_H263_PLUS   (1 << 20)

H.263+ Video

Definition at line 278 of file frame.h.

#define AST_FORMAT_H264   (1 << 21)

H.264 Video

Definition at line 280 of file frame.h.

Referenced by h264_read(), and h264_write().

#define AST_FORMAT_ILBC   (1 << 10)

iLBC Free Compression

Definition at line 258 of file frame.h.

Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_codec_interp_len(), convertcap(), ilbc_read(), ilbc_write(), and ilbctolin_sample().

#define AST_FORMAT_JPEG   (1 << 16)

JPEG Images

Definition at line 270 of file frame.h.

Referenced by jpeg_read_image(), and jpeg_write_image().

#define AST_FORMAT_LPC10   (1 << 7)

LPC10, 180 samples/frame

Definition at line 252 of file frame.h.

Referenced by ast_best_codec(), ast_codec_get_samples(), and lpc10tolin_sample().

#define AST_FORMAT_MAX_AUDIO   (1 << 15)

Maximum audio format

Definition at line 266 of file frame.h.

Referenced by add_sdp(), ast_filehelper(), ast_openvstream(), ast_playstream(), ast_rtp_read(), ast_translate_available_formats(), ast_writestream(), filestream_destructor(), oh323_request(), phone_read(), sip_request_call(), skinny_request(), transmit_connect_with_sdp(), and transmit_modify_with_sdp().

#define AST_FORMAT_MAX_VIDEO   (1 << 24)

Maximum video format

Definition at line 282 of file frame.h.

Referenced by add_sdp(), ast_openvstream(), and ast_translate_available_formats().

#define AST_FORMAT_PNG   (1 << 17)

PNG Images

Definition at line 272 of file frame.h.

Referenced by phone_read().

#define AST_FORMAT_SLINEAR   (1 << 6)

Raw 16-bit Signed Linear (8000 Hz) PCM

Definition at line 250 of file frame.h.

Referenced by __ast_play_and_record(), __ast_register_translator(), action_originate(), agent_new(), alsa_new(), alsa_read(), alsa_request(), ast_audiohook_read_frame(), ast_best_codec(), ast_channel_make_compatible(), ast_channel_start_silence_generator(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_call_progress(), ast_dsp_digitdetect(), ast_dsp_process(), ast_dsp_silence(), ast_frame_adjust_volume(), ast_frame_slinear_sum(), ast_rtp_read(), ast_slinfactory_feed(), attempt_reconnect(), audio_audiohook_write_list(), audiohook_read_frame_both(), audiohook_read_frame_single(), background_detect_exec(), build_conf(), chanspy_exec(), conf_run(), connect_link(), dahdi_new(), dahdi_read(), dahdi_translate(), dahdi_write(), dictate_exec(), do_waiting(), eagi_exec(), extenspy_exec(), find_transcoders(), handle_recordfile(), iax_frame_wrap(), ices_exec(), init_outgoing(), is_encoder(), isAnsweringMachine(), linear_alloc(), linear_generator(), lintoadpcm_sample(), lintoalaw_sample(), lintog722_sample(), lintog726_sample(), lintogsm_sample(), lintoilbc_sample(), lintolpc10_sample(), lintospeex_sample(), lintoulaw_sample(), load_module(), measurenoise(), misdn_set_opt_exec(), mixmonitor_thread(), moh_class_malloc(), mp3_exec(), nbs_request(), nbs_xwrite(), NBScat_exec(), ogg_vorbis_read(), ogg_vorbis_write(), oh323_rtp_read(), orig_app(), orig_exten(), oss_new(), oss_read(), oss_request(), parkandannounce_exec(), phone_new(), phone_read(), phone_request(), phone_setup(), phone_write(), playtones_alloc(), read_config(), rpt(), rpt_call(), rpt_tele_thread(), send_waveform_to_channel(), silence_generator_generate(), slinear_read(), slinear_write(), sms_generate(), socket_process(), speech_background(), speech_create(), spy_generate(), tonepair_alloc(), wav_read(), and wav_write().

#define AST_FORMAT_SPEEX   (1 << 9)

SpeeX Free Compression

Definition at line 256 of file frame.h.

Referenced by ast_best_codec(), ast_codec_get_samples(), ast_rtp_write(), convertcap(), and speextolin_sample().

#define AST_FORMAT_ULAW   (1 << 2)

Raw mu-law data (G.711)

Definition at line 242 of file frame.h.

Referenced by __adsi_transmit_messages(), adsi_careful_send(), alarmreceiver_exec(), ast_adsi_transmit_message_full(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), codec_ast2skinny(), codec_skinny2ast(), conf_run(), convertcap(), dahdi_new(), dahdi_read(), dahdi_translate(), dahdi_write(), disa_exec(), find_transcoders(), is_encoder(), load_module(), milliwatt_generate(), oh323_rtp_read(), old_milliwatt_exec(), phone_request(), phone_setup(), phone_write(), pri_dchannel(), send_tone_burst(), ulawtoalaw_sample(), and ulawtolin_sample().

#define AST_FORMAT_VIDEO_MASK   (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK))

Definition at line 283 of file frame.h.

Referenced by add_sdp(), ast_request(), ast_translate_available_formats(), check_user_full(), create_addr_from_peer(), func_channel_read(), gtalk_new(), gtalk_rtp_read(), sip_new(), and sip_rtp_read().

#define ast_frame_byteswap_be ( fr   )     do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data, __f->data, __f->samples); } while(0)

Definition at line 435 of file frame.h.

Referenced by ast_rtp_read(), and socket_process().

#define ast_frame_byteswap_le ( fr   )     do { ; } while(0)

Definition at line 434 of file frame.h.

Referenced by phone_read().

#define AST_FRAME_DTMF   AST_FRAME_DTMF_END

Definition at line 125 of file frame.h.

Referenced by __action_dialoffhook(), __adsi_transmit_messages(), __ast_play_and_record(), agent_ack_sleep(), app_exec(), ast_audiohook_write_list(), ast_bridge_call(), ast_dsp_process(), ast_feature_request_and_dial(), ast_jb_put(), background_detect_exec(), cb_events(), channel_spy(), conf_exec(), conf_run(), console_dial(), console_dial_deprecated(), dahdi_bridge(), dahdi_read(), dictate_exec(), disa_exec(), do_immediate_setup(), echo_exec(), gtalk_handle_dtmf(), handle_recordfile(), handle_request(), handle_request_info(), mgcp_rtp_read(), misdn_bridge(), mp3_exec(), NBScat_exec(), oh323_rtp_read(), phone_exception(), process_ast_dsp(), receive_dtmf_digits(), rpt(), rpt_call(), send_waveform_to_channel(), sip_rtp_read(), speech_background(), ss_thread(), wait_for_answer(), and wait_for_winner().

#define AST_FRAME_SET_BUFFER ( fr,
_base,
_ofs,
_datalen   ) 

Value:

{              \
   (fr)->data = (char *)_base + (_ofs);   \
   (fr)->offset = (_ofs);        \
   (fr)->datalen = (_datalen);      \
   }
Set the various field of a frame to point to a buffer. Typically you set the base address of the buffer, the offset as AST_FRIENDLY_OFFSET, and the datalen as the amount of bytes queued. The remaining things (to be done manually) is set the number of samples, which cannot be derived from the datalen unless you know the number of bits per sample.

Definition at line 187 of file frame.h.

Referenced by g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), ilbc_read(), ogg_vorbis_read(), pcm_read(), slinear_read(), vox_read(), and wav_read().

#define ast_frfree ( fr   )     ast_frame_free(fr, 1)

Definition at line 410 of file frame.h.

Referenced by __adsi_transmit_messages(), __ast_play_and_record(), __ast_queue_frame(), __ast_read(), __ast_request_and_dial(), adsi_careful_send(), agent_ack_sleep(), agent_read(), app_exec(), ast_audiohook_read_frame(), ast_autoservice_stop(), ast_bridge_call(), ast_channel_free(), ast_dsp_process(), ast_feature_request_and_dial(), ast_jb_destroy(), ast_jb_put(), ast_readaudio_callback(), ast_recvtext(), ast_rtp_write(), ast_safe_sleep_conditional(), ast_send_image(), ast_slinfactory_destroy(), ast_slinfactory_feed(), ast_slinfactory_flush(), ast_slinfactory_read(), ast_tonepair(), ast_translate(), ast_udptl_bridge(), ast_waitfordigit_full(), ast_write(), ast_writestream(), async_wait(), audio_audiohook_write_list(), autoservice_run(), background_detect_exec(), bridge_native_loop(), bridge_p2p_loop(), calc_cost(), channel_spy(), check_goto_on_transfer(), conf_exec(), conf_flush(), conf_free(), conf_run(), create_jb(), dahdi_bridge(), dictate_exec(), disa_exec(), do_atxfer(), do_idle_thread(), do_parking_thread(), do_waiting(), echo_exec(), find_cache(), gen_generate(), handle_invite_replaces(), handle_recordfile(), iax_park_thread(), ices_exec(), isAnsweringMachine(), jb_empty_and_reset_adaptive(), jb_empty_and_reset_fixed(), jb_get_and_deliver(), masq_park_call(), measurenoise(), moh_files_generator(), monitor_dial(), mp3_exec(), NBScat_exec(), receive_dtmf_digits(), recordthread(), rpt(), run_agi(), send_tone_burst(), send_waveform_to_channel(), sendurl_exec(), speech_background(), spy_generate(), ss_thread(), wait_for_answer(), wait_for_hangup(), wait_for_winner(), waitforring_exec(), and waitstream_core().

#define AST_FRIENDLY_OFFSET   64

Definition at line 198 of file frame.h.

Referenced by __get_from_jb(), alsa_read(), ast_frdup(), ast_frisolate(), ast_prod(), ast_rtcp_read(), ast_rtp_read(), ast_smoother_read(), ast_trans_frameout(), ast_udptl_read(), conf_run(), dahdi_decoder_frameout(), dahdi_encoder_frameout(), dahdi_read(), g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), iax_frame_wrap(), ilbc_read(), jb_get_and_deliver(), linear_generator(), milliwatt_generate(), moh_generate(), mohalloc(), mp3_exec(), NBScat_exec(), newpvt(), ogg_vorbis_read(), oss_read(), pcm_read(), phone_read(), process_rfc3389(), send_tone_burst(), send_waveform_to_channel(), slinear_read(), sms_generate(), vox_read(), and wav_read().

#define AST_HTML_BEGIN   4

Beginning frame

Definition at line 222 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_DATA   2

Data frame

Definition at line 220 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_END   8

End frame

Definition at line 224 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_LDCOMPLETE   16

Load is complete

Definition at line 226 of file frame.h.

Referenced by ast_frame_dump(), and sendurl_exec().

#define AST_HTML_LINKREJECT   20

Reject link request

Definition at line 234 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_LINKURL   18

Send URL, and track

Definition at line 230 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_NOSUPPORT   17

Peer is unable to support HTML

Definition at line 228 of file frame.h.

Referenced by ast_frame_dump(), and sendurl_exec().

#define AST_HTML_UNLINK   19

No more HTML linkage

Definition at line 232 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_URL   1

Sending a URL

Definition at line 218 of file frame.h.

Referenced by ast_channel_sendurl(), and ast_frame_dump().

#define AST_MALLOCD_DATA   (1 << 1)

Need the data be free'd?

Definition at line 206 of file frame.h.

Referenced by ast_frame_free(), and ast_frisolate().

#define AST_MALLOCD_HDR   (1 << 0)

Need the header be free'd?

Definition at line 204 of file frame.h.

Referenced by ast_frame_free(), ast_frame_header_new(), ast_frdup(), and ast_frisolate().

#define AST_MALLOCD_SRC   (1 << 2)

Need the source be free'd? (haha!)

Definition at line 208 of file frame.h.

Referenced by ast_frame_free(), and ast_frisolate().

#define AST_MIN_OFFSET   32

Definition at line 201 of file frame.h.

Referenced by __ast_smoother_feed().

#define AST_MODEM_T38   1

T.38 Fax-over-IP

Definition at line 212 of file frame.h.

Referenced by ast_frame_dump(), and udptl_rx_packet().

#define AST_MODEM_V150   2

V.150 Modem-over-IP

Definition at line 214 of file frame.h.

Referenced by ast_frame_dump().

#define AST_OPTION_AUDIO_MODE   4

Set (or clear) Audio (Not-Clear) Mode

Definition at line 330 of file frame.h.

Referenced by dahdi_hangup(), and dahdi_setoption().

#define AST_OPTION_ECHOCAN   8

Explicitly enable or disable echo cancelation for the given channel

Definition at line 352 of file frame.h.

Referenced by dahdi_setoption().

#define AST_OPTION_FLAG_ACCEPT   1

Definition at line 313 of file frame.h.

#define AST_OPTION_FLAG_ANSWER   5

Definition at line 316 of file frame.h.

#define AST_OPTION_FLAG_QUERY   4

Definition at line 315 of file frame.h.

#define AST_OPTION_FLAG_REJECT   2

Definition at line 314 of file frame.h.

#define AST_OPTION_FLAG_REQUEST   0

Definition at line 312 of file frame.h.

Referenced by ast_bridge_call(), and iax2_setoption().

#define AST_OPTION_FLAG_WTF   6

Definition at line 317 of file frame.h.

#define AST_OPTION_OPRMODE   7

Definition at line 349 of file frame.h.

Referenced by dahdi_setoption().

#define AST_OPTION_RELAXDTMF   3

Relax the parameters for DTMF reception (mainly for radio use)

Definition at line 327 of file frame.h.

Referenced by dahdi_setoption(), and rpt().

#define AST_OPTION_RXGAIN   6

Set channel receive gain Option data is a single signed char representing number of decibels (dB) to set gain to (on top of any gain specified in channel driver)

Definition at line 346 of file frame.h.

Referenced by dahdi_setoption(), func_channel_write(), iax2_setoption(), play_record_review(), reset_volumes(), set_talk_volume(), and vm_forwardoptions().

#define AST_OPTION_TDD   2

Put a compatible channel into TDD (TTY for the hearing-impared) mode

Definition at line 324 of file frame.h.

Referenced by dahdi_hangup(), dahdi_setoption(), and handle_tddmode().

#define AST_OPTION_TONE_VERIFY   1

Verify touchtones by muting audio transmission (and reception) and verify the tone is still present

Definition at line 321 of file frame.h.

Referenced by conf_run(), dahdi_hangup(), dahdi_setoption(), rpt(), and try_calling().

#define AST_OPTION_TXGAIN   5

Set channel transmit gain Option data is a single signed char representing number of decibels (dB) to set gain to (on top of any gain specified in channel driver)

Definition at line 338 of file frame.h.

Referenced by common_exec(), dahdi_setoption(), func_channel_write(), iax2_setoption(), reset_volumes(), and set_listen_volume().

#define ast_smoother_feed ( s,
f   )     __ast_smoother_feed(s, f, 0)

Definition at line 494 of file frame.h.

Referenced by ast_rtp_write().

#define ast_smoother_feed_be ( s,
f   )     __ast_smoother_feed(s, f, 1)

Definition at line 496 of file frame.h.

Referenced by ast_rtp_write().

#define ast_smoother_feed_le ( s,
f   )     __ast_smoother_feed(s, f, 0)

Definition at line 497 of file frame.h.

#define AST_SMOOTHER_FLAG_BE   (1 << 1)

Definition at line 309 of file frame.h.

Referenced by ast_rtp_write().

#define AST_SMOOTHER_FLAG_G729   (1 << 0)

Definition at line 308 of file frame.h.

Referenced by __ast_smoother_feed(), ast_smoother_read(), and smoother_frame_feed().


Enumeration Type Documentation

anonymous enum

Enumerator:
AST_FRFLAG_HAS_TIMING_INFO  This frame contains valid timing information
AST_FRFLAG_FROM_TRANSLATOR  This frame came from a translator and is still the original frame. The translator can not be free'd if the frame inside of it still has this flag set.
AST_FRFLAG_FROM_DSP  This frame came from a dsp and is still the original frame. The dsp cannot be free'd if the frame inside of it still has this flag set.
AST_FRFLAG_FROM_FILESTREAM  This frame came from a filestream and is still the original frame. The filestream cannot be free'd if the frame inside of it still has this flag set.

Definition at line 127 of file frame.h.

00127      {
00128    /*! This frame contains valid timing information */
00129    AST_FRFLAG_HAS_TIMING_INFO = (1 << 0),
00130    /*! This frame came from a translator and is still the original frame.
00131     *  The translator can not be free'd if the frame inside of it still has
00132     *  this flag set. */
00133    AST_FRFLAG_FROM_TRANSLATOR = (1 << 1),
00134    /*! This frame came from a dsp and is still the original frame.
00135     *  The dsp cannot be free'd if the frame inside of it still has
00136     *  this flag set. */
00137    AST_FRFLAG_FROM_DSP = (1 << 2),
00138    /*! This frame came from a filestream and is still the original frame.
00139     *  The filestream cannot be free'd if the frame inside of it still has
00140     *  this flag set. */
00141    AST_FRFLAG_FROM_FILESTREAM = (1 << 3),
00142 };

enum ast_control_frame_type

Enumerator:
AST_CONTROL_HANGUP  Other end has hungup
AST_CONTROL_RING  Local ring
AST_CONTROL_RINGING  Remote end is ringing
AST_CONTROL_ANSWER  Remote end has answered
AST_CONTROL_BUSY  Remote end is busy
AST_CONTROL_TAKEOFFHOOK  Make it go off hook
AST_CONTROL_OFFHOOK  Line is off hook
AST_CONTROL_CONGESTION  Congestion (circuits busy)
AST_CONTROL_FLASH  Flash hook
AST_CONTROL_WINK  Wink
AST_CONTROL_OPTION  Set a low-level option
AST_CONTROL_RADIO_KEY  Key Radio
AST_CONTROL_RADIO_UNKEY  Un-Key Radio
AST_CONTROL_PROGRESS  Indicate PROGRESS
AST_CONTROL_PROCEEDING  Indicate CALL PROCEEDING
AST_CONTROL_HOLD  Indicate call is placed on hold
AST_CONTROL_UNHOLD  Indicate call is left from hold
AST_CONTROL_VIDUPDATE  Indicate video frame update
AST_CONTROL_ATXFERCMD  AMI triggered attended transfer
AST_CONTROL_SRCUPDATE  Indicate source of media has changed

Definition at line 285 of file frame.h.

00285                             {
00286    AST_CONTROL_HANGUP = 1,    /*!< Other end has hungup */
00287    AST_CONTROL_RING = 2,      /*!< Local ring */
00288    AST_CONTROL_RINGING = 3,   /*!< Remote end is ringing */
00289    AST_CONTROL_ANSWER = 4,    /*!< Remote end has answered */
00290    AST_CONTROL_BUSY = 5,      /*!< Remote end is busy */
00291    AST_CONTROL_TAKEOFFHOOK = 6,  /*!< Make it go off hook */
00292    AST_CONTROL_OFFHOOK = 7,   /*!< Line is off hook */
00293    AST_CONTROL_CONGESTION = 8,   /*!< Congestion (circuits busy) */
00294    AST_CONTROL_FLASH = 9,     /*!< Flash hook */
00295    AST_CONTROL_WINK = 10,     /*!< Wink */
00296    AST_CONTROL_OPTION = 11,   /*!< Set a low-level option */
00297    AST_CONTROL_RADIO_KEY = 12,   /*!< Key Radio */
00298    AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */
00299    AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */
00300    AST_CONTROL_PROCEEDING = 15,  /*!< Indicate CALL PROCEEDING */
00301    AST_CONTROL_HOLD = 16,     /*!< Indicate call is placed on hold */
00302    AST_CONTROL_UNHOLD = 17,   /*!< Indicate call is left from hold */
00303    AST_CONTROL_VIDUPDATE = 18,   /*!< Indicate video frame update */
00304    AST_CONTROL_ATXFERCMD = 19,   /*!< AMI triggered attended transfer */
00305    AST_CONTROL_SRCUPDATE = 20,     /*!< Indicate source of media has changed */
00306 };

enum ast_frame_type

Frame types.

Note:
It is important that the values of each frame type are never changed, because it will break backwards compatability with older versions.
Enumerator:
AST_FRAME_DTMF_END  DTMF end event, subclass is the digit
AST_FRAME_VOICE  Voice data, subclass is AST_FORMAT_*
AST_FRAME_VIDEO  Video frame, maybe?? :)
AST_FRAME_CONTROL  A control frame, subclass is AST_CONTROL_*
AST_FRAME_NULL  An empty, useless frame
AST_FRAME_IAX  Inter Asterisk Exchange private frame type
AST_FRAME_TEXT  Text messages
AST_FRAME_IMAGE  Image Frames
AST_FRAME_HTML  HTML Frame
AST_FRAME_CNG  Comfort Noise frame (subclass is level of CNG in -dBov), body may include zero or more 8-bit quantization coefficients
AST_FRAME_MODEM  Modem-over-IP data streams
AST_FRAME_DTMF_BEGIN  DTMF begin event, subclass is the digit

Definition at line 98 of file frame.h.

00098                     {
00099    /*! DTMF end event, subclass is the digit */
00100    AST_FRAME_DTMF_END = 1,
00101    /*! Voice data, subclass is AST_FORMAT_* */
00102    AST_FRAME_VOICE,
00103    /*! Video frame, maybe?? :) */
00104    AST_FRAME_VIDEO,
00105    /*! A control frame, subclass is AST_CONTROL_* */
00106    AST_FRAME_CONTROL,
00107    /*! An empty, useless frame */
00108    AST_FRAME_NULL,
00109    /*! Inter Asterisk Exchange private frame type */
00110    AST_FRAME_IAX,
00111    /*! Text messages */
00112    AST_FRAME_TEXT,
00113    /*! Image Frames */
00114    AST_FRAME_IMAGE,
00115    /*! HTML Frame */
00116    AST_FRAME_HTML,
00117    /*! Comfort Noise frame (subclass is level of CNG in -dBov), 
00118        body may include zero or more 8-bit quantization coefficients */
00119    AST_FRAME_CNG,
00120    /*! Modem-over-IP data streams */
00121    AST_FRAME_MODEM,  
00122    /*! DTMF begin event, subclass is the digit */
00123    AST_FRAME_DTMF_BEGIN,
00124 };


Function Documentation

int __ast_smoother_feed ( struct ast_smoother s,
struct ast_frame f,
int  swap 
)

Definition at line 216 of file frame.c.

References AST_FRAME_VOICE, ast_log(), AST_MIN_OFFSET, AST_SMOOTHER_FLAG_G729, ast_swapcopy_samples(), f, LOG_WARNING, s, smoother_frame_feed(), and SMOOTHER_SIZE.

00217 {
00218    if (f->frametype != AST_FRAME_VOICE) {
00219       ast_log(LOG_WARNING, "Huh?  Can't smooth a non-voice frame!\n");
00220       return -1;
00221    }
00222    if (!s->format) {
00223       s->format = f->subclass;
00224       s->samplesperbyte = (float)f->samples / (float)f->datalen;
00225    } else if (s->format != f->subclass) {
00226       ast_log(LOG_WARNING, "Smoother was working on %d format frames, now trying to feed %d?\n", s->format, f->subclass);
00227       return -1;
00228    }
00229    if (s->len + f->datalen > SMOOTHER_SIZE) {
00230       ast_log(LOG_WARNING, "Out of smoother space\n");
00231       return -1;
00232    }
00233    if (((f->datalen == s->size) ||
00234         ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) &&
00235        !s->opt &&
00236        !s->len &&
00237        (f->offset >= AST_MIN_OFFSET)) {
00238       /* Optimize by sending the frame we just got
00239          on the next read, thus eliminating the douple
00240          copy */
00241       if (swap)
00242          ast_swapcopy_samples(f->data, f->data, f->samples);
00243       s->opt = f;
00244       s->opt_needs_swap = swap ? 1 : 0;
00245       return 0;
00246    }
00247 
00248    return smoother_frame_feed(s, f, swap);
00249 }

char* ast_codec2str ( int  codec  ) 

Get a name from a format Gets a name from a format.

Parameters:
codec codec number (1,2,4,8,16,etc.)
Returns:
This returns a static string identifying the format on success, 0 on error.

Definition at line 639 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().

00640 {
00641    int x;
00642    char *ret = "unknown";
00643    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
00644       if(AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == codec) {
00645          ret = AST_FORMAT_LIST[x].desc;
00646          break;
00647       }
00648    }
00649    return ret;
00650 }

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 1314 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().

01315 {
01316    int x, ret = 0, slot;
01317 
01318    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
01319       slot = pref->order[x];
01320 
01321       if (!slot)
01322          break;
01323       if (formats & AST_FORMAT_LIST[slot-1].bits) {
01324          ret = AST_FORMAT_LIST[slot-1].bits;
01325          break;
01326       }
01327    }
01328    if(ret & AST_FORMAT_AUDIO_MASK)
01329       return ret;
01330 
01331    if (option_debug > 3)
01332       ast_log(LOG_DEBUG, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec");
01333 
01334       return find_best ? ast_best_codec(formats) : 0;
01335 }

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 1573 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().

01574 {
01575    int len = 0;
01576 
01577    /* XXX Still need speex, g723, and lpc10 XXX */ 
01578    switch(format) {
01579    case AST_FORMAT_ILBC:
01580       len = (samples / 240) * 50;
01581       break;
01582    case AST_FORMAT_GSM:
01583       len = (samples / 160) * 33;
01584       break;
01585    case AST_FORMAT_G729A:
01586       len = samples / 8;
01587       break;
01588    case AST_FORMAT_SLINEAR:
01589       len = samples * 2;
01590       break;
01591    case AST_FORMAT_ULAW:
01592    case AST_FORMAT_ALAW:
01593       len = samples;
01594       break;
01595    case AST_FORMAT_G722:
01596    case AST_FORMAT_ADPCM:
01597    case AST_FORMAT_G726:
01598    case AST_FORMAT_G726_AAL2:
01599       len = samples / 2;
01600       break;
01601    default:
01602       ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format));
01603    }
01604 
01605    return len;
01606 }

int ast_codec_get_samples ( struct ast_frame f  ) 

Returns the number of samples contained in the frame.

Definition at line 1530 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().

01531 {
01532    int samples=0;
01533    switch(f->subclass) {
01534    case AST_FORMAT_SPEEX:
01535       samples = speex_samples(f->data, f->datalen);
01536       break;
01537    case AST_FORMAT_G723_1:
01538                 samples = g723_samples(f->data, f->datalen);
01539       break;
01540    case AST_FORMAT_ILBC:
01541       samples = 240 * (f->datalen / 50);
01542       break;
01543    case AST_FORMAT_GSM:
01544       samples = 160 * (f->datalen / 33);
01545       break;
01546    case AST_FORMAT_G729A:
01547       samples = f->datalen * 8;
01548       break;
01549    case AST_FORMAT_SLINEAR:
01550       samples = f->datalen / 2;
01551       break;
01552    case AST_FORMAT_LPC10:
01553                 /* assumes that the RTP packet contains one LPC10 frame */
01554       samples = 22 * 8;
01555       samples += (((char *)(f->data))[7] & 0x1) * 8;
01556       break;
01557    case AST_FORMAT_ULAW:
01558    case AST_FORMAT_ALAW:
01559       samples = f->datalen;
01560       break;
01561    case AST_FORMAT_G722:
01562    case AST_FORMAT_ADPCM:
01563    case AST_FORMAT_G726:
01564    case AST_FORMAT_G726_AAL2:
01565       samples = f->datalen * 2;
01566       break;
01567    default:
01568       ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass));
01569    }
01570    return samples;
01571 }

static int ast_codec_interp_len ( int  format  )  [inline, static]

Gets duration in ms of interpolation frame for a format.

Definition at line 571 of file frame.h.

References AST_FORMAT_ILBC.

Referenced by __get_from_jb(), and jb_get_and_deliver().

00572 { 
00573    return (format == AST_FORMAT_ILBC) ? 30 : 20;
00574 }

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 1173 of file frame.c.

References ast_codec_pref_remove(), AST_FORMAT_LIST, and ast_codec_pref::order.

Referenced by ast_parse_allow_disallow().

01174 {
01175    int x, newindex = 0;
01176 
01177    ast_codec_pref_remove(pref, format);
01178 
01179    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
01180       if(AST_FORMAT_LIST[x].bits == format) {
01181          newindex = x + 1;
01182          break;
01183       }
01184    }
01185 
01186    if(newindex) {
01187       for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
01188          if(!pref->order[x]) {
01189             pref->order[x] = newindex;
01190             break;
01191          }
01192       }
01193    }
01194 
01195    return x;
01196 }

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 1075 of file frame.c.

References ast_codec_pref::order.

Referenced by check_access(), create_addr(), dump_prefs(), and socket_process().

01076 {
01077    int x, differential = (int) 'A', mem;
01078    char *from, *to;
01079 
01080    if(right) {
01081       from = pref->order;
01082       to = buf;
01083       mem = size;
01084    } else {
01085       to = pref->order;
01086       from = buf;
01087       mem = 32;
01088    }
01089 
01090    memset(to, 0, mem);
01091    for (x = 0; x < 32 ; x++) {
01092       if(!from[x])
01093          break;
01094       to[x] = right ? (from[x] + differential) : (from[x] - differential);
01095    }
01096 }

struct ast_format_list ast_codec_pref_getsize ( struct ast_codec_pref pref,
int  format 
)

Get packet size for codec.

Definition at line 1275 of file frame.c.

References AST_FORMAT_LIST, ast_format_list::bits, and format.

Referenced by add_codec_to_sdp(), ast_rtp_bridge(), ast_rtp_codec_setpref(), ast_rtp_write(), handle_open_receive_channel_ack_message(), and transmit_connect().

01276 {
01277    int x, index = -1, framems = 0;
01278    struct ast_format_list fmt = {0};
01279 
01280    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
01281       if(AST_FORMAT_LIST[x].bits == format) {
01282          fmt = AST_FORMAT_LIST[x];
01283          index = x;
01284          break;
01285       }
01286    }
01287 
01288    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
01289       if(pref->order[x] == (index + 1)) {
01290          framems = pref->framing[x];
01291          break;
01292       }
01293    }
01294 
01295    /* size validation */
01296    if(!framems)
01297       framems = AST_FORMAT_LIST[index].def_ms;
01298 
01299    if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */
01300       framems -= framems % AST_FORMAT_LIST[index].inc_ms;
01301 
01302    if(framems < AST_FORMAT_LIST[index].min_ms)
01303       framems = AST_FORMAT_LIST[index].min_ms;
01304 
01305    if(framems > AST_FORMAT_LIST[index].max_ms)
01306       framems = AST_FORMAT_LIST[index].max_ms;
01307 
01308    fmt.cur_ms = framems;
01309 
01310    return fmt;
01311 }

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 1133 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().

01134 {
01135    int slot = 0;
01136 
01137    
01138    if((index >= 0) && (index < sizeof(pref->order))) {
01139       slot = pref->order[index];
01140    }
01141 
01142    return slot ? AST_FORMAT_LIST[slot-1].bits : 0;
01143 }

void ast_codec_pref_init ( struct ast_codec_pref pref  ) 

Initialize an audio codec preference to "no preference" 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.

Definition at line 1199 of file frame.c.

References ARRAY_LEN, AST_FORMAT_LIST, ast_codec_pref::framing, and ast_codec_pref::order.

Referenced by create_addr().

01200 {
01201    int x, newindex = 0;
01202 
01203    /* First step is to get the codecs "index number" */
01204    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01205       if (AST_FORMAT_LIST[x].bits == format) {
01206          newindex = x + 1;
01207          break;
01208       }
01209    }
01210    /* Done if its unknown */
01211    if (!newindex)
01212       return;
01213 
01214    /* Now find any existing occurrence, or the end */
01215    for (x = 0; x < 32; x++) {
01216       if (!pref->order[x] || pref->order[x] == newindex)
01217          break;
01218    }
01219 
01220    if (only_if_existing && !pref->order[x])
01221       return;
01222 
01223    /* Move down to make space to insert - either all the way to the end,
01224       or as far as the existing location (which will be overwritten) */
01225    for (; x > 0; x--) {
01226       pref->order[x] = pref->order[x - 1];
01227       pref->framing[x] = pref->framing[x - 1];
01228    }
01229 
01230    /* And insert the new entry */
01231    pref->order[0] = newindex;
01232    pref->framing[0] = 0; /* ? */
01233 }

void ast_codec_pref_remove ( struct ast_codec_pref pref,
int  format 
)

Remove audio a codec from a preference list.

Definition at line 1146 of file frame.c.

References AST_FORMAT_LIST, and ast_codec_pref::order.

Referenced by ast_codec_pref_append(), and ast_parse_allow_disallow().

01147 {
01148    struct ast_codec_pref oldorder;
01149    int x, y = 0;
01150    int slot;
01151    int size;
01152 
01153    if(!pref->order[0])
01154       return;
01155 
01156    memcpy(&oldorder, pref, sizeof(oldorder));
01157    memset(pref, 0, sizeof(*pref));
01158 
01159    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
01160       slot = oldorder.order[x];
01161       size = oldorder.framing[x];
01162       if(! slot)
01163          break;
01164       if(AST_FORMAT_LIST[slot-1].bits != format) {
01165          pref->order[y] = slot;
01166          pref->framing[y++] = size;
01167       }
01168    }
01169    
01170 }

int ast_codec_pref_setsize ( struct ast_codec_pref pref,
int  format,
int  framems 
)

Set packet size for codec.

Definition at line 1236 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().

01237 {
01238    int x, index = -1;
01239 
01240    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
01241       if(AST_FORMAT_LIST[x].bits == format) {
01242          index = x;
01243          break;
01244       }
01245    }
01246 
01247    if(index < 0)
01248       return -1;
01249 
01250    /* size validation */
01251    if(!framems)
01252       framems = AST_FORMAT_LIST[index].def_ms;
01253 
01254    if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */
01255       framems -= framems % AST_FORMAT_LIST[index].inc_ms;
01256 
01257    if(framems < AST_FORMAT_LIST[index].min_ms)
01258       framems = AST_FORMAT_LIST[index].min_ms;
01259 
01260    if(framems > AST_FORMAT_LIST[index].max_ms)
01261       framems = AST_FORMAT_LIST[index].max_ms;
01262 
01263 
01264    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
01265       if(pref->order[x] == (index + 1)) {
01266          pref->framing[x] = framems;
01267          break;
01268       }
01269    }
01270 
01271    return x;
01272 }

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 1098 of file frame.c.

References ast_codec_pref_index(), and ast_getformatname().

Referenced by dump_prefs(), and socket_process().

01099 {
01100    int x, codec; 
01101    size_t total_len, slen;
01102    char *formatname;
01103    
01104    memset(buf,0,size);
01105    total_len = size;
01106    buf[0] = '(';
01107    total_len--;
01108    for(x = 0; x < 32 ; x++) {
01109       if(total_len <= 0)
01110          break;
01111       if(!(codec = ast_codec_pref_index(pref,x)))
01112          break;
01113       if((formatname = ast_getformatname(codec))) {
01114          slen = strlen(formatname);
01115          if(slen > total_len)
01116             break;
01117          strncat(buf, formatname, total_len - 1); /* safe */
01118          total_len -= slen;
01119       }
01120       if(total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) {
01121          strncat(buf, "|", total_len - 1); /* safe */
01122          total_len--;
01123       }
01124    }
01125    if(total_len) {
01126       strncat(buf, ")", total_len - 1); /* safe */
01127       total_len--;
01128    }
01129 
01130    return size - total_len;
01131 }

static force_inline int ast_format_rate ( int  format  )  [static]

Get the sample rate for a given format.

Definition at line 598 of file frame.h.

References AST_FORMAT_G722.

Referenced by ast_read_generator_actions(), ast_readaudio_callback(), ast_readvideo_callback(), ast_rtp_read(), ast_translate(), calc_cost(), and generator_force().

00599 {
00600    if (format == AST_FORMAT_G722)
00601       return 16000;
00602 
00603    return 8000;
00604 }

int ast_frame_adjust_volume ( struct ast_frame f,
int  adjustment 
)

Adjusts the volume of the audio samples contained in a frame.

Parameters:
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.
Returns:
0 for success, non-zero for an error

Definition at line 1608 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().

01609 {
01610    int count;
01611    short *fdata = f->data;
01612    short adjust_value = abs(adjustment);
01613 
01614    if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR))
01615       return -1;
01616 
01617    if (!adjustment)
01618       return 0;
01619 
01620    for (count = 0; count < f->samples; count++) {
01621       if (adjustment > 0) {
01622          ast_slinear_saturated_multiply(&fdata[count], &adjust_value);
01623       } else if (adjustment < 0) {
01624          ast_slinear_saturated_divide(&fdata[count], &adjust_value);
01625       }
01626    }
01627 
01628    return 0;
01629 }

void ast_frame_dump ( const char *  name,
struct ast_frame f,
char *  prefix 
)

Dump a frame for debugging purposes

Definition at line 793 of file frame.c.

References AST_CONTROL_ANSWER, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_FLASH, AST_CONTROL_HANGUP, AST_CONTROL_HOLD, AST_CONTROL_OFFHOOK, AST_CONTROL_OPTION, AST_CONTROL_PROCEEDING, AST_CONTROL_PROGRESS, AST_CONTROL_RADIO_KEY, AST_CONTROL_RADIO_UNKEY, AST_CONTROL_RING, AST_CONTROL_RINGING, AST_CONTROL_TAKEOFFHOOK, AST_CONTROL_UNHOLD, AST_CONTROL_WINK, ast_copy_string(), 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().

00794 {
00795    const char noname[] = "unknown";
00796    char ftype[40] = "Unknown Frametype";
00797    char cft[80];
00798    char subclass[40] = "Unknown Subclass";
00799    char csub[80];
00800    char moreinfo[40] = "";
00801    char cn[60];
00802    char cp[40];
00803    char cmn[40];
00804 
00805    if (!name)
00806       name = noname;
00807 
00808 
00809    if (!f) {
00810       ast_verbose("%s [ %s (NULL) ] [%s]\n", 
00811          term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00812          term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 
00813          term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00814       return;
00815    }
00816    /* XXX We should probably print one each of voice and video when the format changes XXX */
00817    if (f->frametype == AST_FRAME_VOICE)
00818       return;
00819    if (f->frametype == AST_FRAME_VIDEO)
00820       return;
00821    switch(f->frametype) {
00822    case AST_FRAME_DTMF_BEGIN:
00823       strcpy(ftype, "DTMF Begin");
00824       subclass[0] = f->subclass;
00825       subclass[1] = '\0';
00826       break;
00827    case AST_FRAME_DTMF_END:
00828       strcpy(ftype, "DTMF End");
00829       subclass[0] = f->subclass;
00830       subclass[1] = '\0';
00831       break;
00832    case AST_FRAME_CONTROL:
00833       strcpy(ftype, "Control");
00834       switch(f->subclass) {
00835       case AST_CONTROL_HANGUP:
00836          strcpy(subclass, "Hangup");
00837          break;
00838       case AST_CONTROL_RING:
00839          strcpy(subclass, "Ring");
00840          break;
00841       case AST_CONTROL_RINGING:
00842          strcpy(subclass, "Ringing");
00843          break;
00844       case AST_CONTROL_ANSWER:
00845          strcpy(subclass, "Answer");
00846          break;
00847       case AST_CONTROL_BUSY:
00848          strcpy(subclass, "Busy");
00849          break;
00850       case AST_CONTROL_TAKEOFFHOOK:
00851          strcpy(subclass, "Take Off Hook");
00852          break;
00853       case AST_CONTROL_OFFHOOK:
00854          strcpy(subclass, "Line Off Hook");
00855          break;
00856       case AST_CONTROL_CONGESTION:
00857          strcpy(subclass, "Congestion");
00858          break;
00859       case AST_CONTROL_FLASH:
00860          strcpy(subclass, "Flash");
00861          break;
00862       case AST_CONTROL_WINK:
00863          strcpy(subclass, "Wink");
00864          break;
00865       case AST_CONTROL_OPTION:
00866          strcpy(subclass, "Option");
00867          break;
00868       case AST_CONTROL_RADIO_KEY:
00869          strcpy(subclass, "Key Radio");
00870          break;
00871       case AST_CONTROL_RADIO_UNKEY:
00872          strcpy(subclass, "Unkey Radio");
00873          break;
00874       case AST_CONTROL_PROGRESS:
00875          strcpy(subclass, "Call Progress");
00876          break;
00877       case AST_CONTROL_PROCEEDING:
00878          strcpy(subclass, "Proceeding");
00879          break;
00880       case AST_CONTROL_HOLD:
00881         strcpy(subclass, "Hold");
00882         break;
00883       case AST_CONTROL_UNHOLD:
00884         strcpy(subclass, "UnHold");
00885         break;
00886       case -1:
00887          strcpy(subclass, "Stop generators");
00888          break;
00889       default:
00890          snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass);
00891       }
00892       break;
00893    case AST_FRAME_NULL:
00894       strcpy(ftype, "Null Frame");
00895       strcpy(subclass, "N/A");
00896       break;
00897    case AST_FRAME_IAX:
00898       /* Should never happen */
00899       strcpy(ftype, "IAX Specific");
00900       snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass);
00901       break;
00902    case AST_FRAME_TEXT:
00903       strcpy(ftype, "Text");
00904       strcpy(subclass, "N/A");
00905       ast_copy_string(moreinfo, f->data, sizeof(moreinfo));
00906       break;
00907    case AST_FRAME_IMAGE:
00908       strcpy(ftype, "Image");
00909       snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass));
00910       break;
00911    case AST_FRAME_HTML:
00912       strcpy(ftype, "HTML");
00913       switch(f->subclass) {
00914       case AST_HTML_URL:
00915          strcpy(subclass, "URL");
00916          ast_copy_string(moreinfo, f->data, sizeof(moreinfo));
00917          break;
00918       case AST_HTML_DATA:
00919          strcpy(subclass, "Data");
00920          break;
00921       case AST_HTML_BEGIN:
00922          strcpy(subclass, "Begin");
00923          break;
00924       case AST_HTML_END:
00925          strcpy(subclass, "End");
00926          break;
00927       case AST_HTML_LDCOMPLETE:
00928          strcpy(subclass, "Load Complete");
00929          break;
00930       case AST_HTML_NOSUPPORT:
00931          strcpy(subclass, "No Support");
00932          break;
00933       case AST_HTML_LINKURL:
00934          strcpy(subclass, "Link URL");
00935          ast_copy_string(moreinfo, f->data, sizeof(moreinfo));
00936          break;
00937       case AST_HTML_UNLINK:
00938          strcpy(subclass, "Unlink");
00939          break;
00940       case AST_HTML_LINKREJECT:
00941          strcpy(subclass, "Link Reject");
00942          break;
00943       default:
00944          snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass);
00945          break;
00946       }
00947       break;
00948    case AST_FRAME_MODEM:
00949       strcpy(ftype, "Modem");
00950       switch (f->subclass) {
00951       case AST_MODEM_T38:
00952          strcpy(subclass, "T.38");
00953          break;
00954       case AST_MODEM_V150:
00955          strcpy(subclass, "V.150");
00956          break;
00957       default:
00958          snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass);
00959          break;
00960       }
00961       break;
00962    default:
00963       snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype);
00964    }
00965    if (!ast_strlen_zero(moreinfo))
00966       ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n",  
00967              term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00968              term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)),
00969              f->frametype, 
00970              term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)),
00971              f->subclass, 
00972              term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)),
00973              term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00974    else
00975       ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n",  
00976              term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00977              term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)),
00978              f->frametype, 
00979              term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)),
00980              f->subclass, 
00981              term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00982 }

struct ast_frame* ast_frame_enqueue ( struct ast_frame head,
struct ast_frame f,
int  maxlen,
int  dupe 
)

Appends a frame to the end of a list of frames, truncating the maximum length of the list.

void ast_frame_free ( struct ast_frame fr,
int  cache 
)

Requests a frame to be allocated Frees a frame.

Parameters:
fr Frame to free
cache Whether to consider this frame for frame caching

Definition at line 354 of file frame.c.

References ast_dsp_frame_freed(), ast_filestream_frame_freed(), AST_FRFLAG_FROM_DSP, AST_FRFLAG_FROM_FILESTREAM, AST_FRFLAG_FROM_TRANSLATOR, AST_LIST_INSERT_HEAD, AST_LIST_LOCK, AST_LIST_REMOVE, AST_LIST_UNLOCK, AST_MALLOCD_DATA, AST_MALLOCD_HDR, AST_MALLOCD_SRC, ast_test_flag, ast_threadstorage_get(), ast_translate_frame_freed(), ast_frame::data, frame_cache, FRAME_CACHE_MAX_SIZE, frames, free, ast_frame::mallocd, ast_frame::offset, and ast_frame::src.

Referenced by mixmonitor_thread().

00355 {
00356    if (ast_test_flag(fr, AST_FRFLAG_FROM_TRANSLATOR)) {
00357       ast_translate_frame_freed(fr);
00358    } else if (ast_test_flag(fr, AST_FRFLAG_FROM_DSP)) {
00359       ast_dsp_frame_freed(fr);
00360    } else if (ast_test_flag(fr, AST_FRFLAG_FROM_FILESTREAM)) {
00361       ast_filestream_frame_freed(fr);
00362    }
00363 
00364    if (!fr->mallocd)
00365       return;
00366 
00367 #if !defined(LOW_MEMORY)
00368    if (cache && fr->mallocd == AST_MALLOCD_HDR) {
00369       /* Cool, only the header is malloc'd, let's just cache those for now 
00370        * to keep things simple... */
00371       struct ast_frame_cache *frames;
00372 
00373       if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames))) 
00374           && frames->size < FRAME_CACHE_MAX_SIZE) {
00375          AST_LIST_INSERT_HEAD(&frames->list, fr, frame_list);
00376          frames->size++;
00377          return;
00378       }
00379    }
00380 #endif
00381    
00382    if (fr->mallocd & AST_MALLOCD_DATA) {
00383       if (fr->data) 
00384          free(fr->data - fr->offset);
00385    }
00386    if (fr->mallocd & AST_MALLOCD_SRC) {
00387       if (fr->src)
00388          free((char *)fr->src);
00389    }
00390    if (fr->mallocd & AST_MALLOCD_HDR) {
00391 #ifdef TRACE_FRAMES
00392       AST_LIST_LOCK(&headerlist);
00393       headers--;
00394       AST_LIST_REMOVE(&headerlist, fr, frame_list);
00395       AST_LIST_UNLOCK(&headerlist);
00396 #endif         
00397       free(fr);
00398    }
00399 }

int ast_frame_slinear_sum ( struct ast_frame f1,
struct ast_frame f2 
)

Sums two frames of audio samples.

Parameters:
f1 The first frame (which will contain the result)
f2 The second frame
Returns:
0 for success, non-zero for an error
The frames must be AST_FRAME_VOICE and must contain AST_FORMAT_SLINEAR samples, and must contain the same number of samples.

Definition at line 1631 of file frame.c.

References AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_slinear_saturated_add(), ast_frame::data, ast_frame::frametype, ast_frame::samples, and ast_frame::subclass.

01632 {
01633    int count;
01634    short *data1, *data2;
01635 
01636    if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR))
01637       return -1;
01638 
01639    if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR))
01640       return -1;
01641 
01642    if (f1->samples != f2->samples)
01643       return -1;
01644 
01645    for (count = 0, data1 = f1->data, data2 = f2->data;
01646         count < f1->samples;
01647         count++, data1++, data2++)
01648       ast_slinear_saturated_add(data1, data2);
01649 
01650    return 0;
01651 }

struct ast_frame* ast_frdup ( const struct ast_frame fr  ) 

Copies a frame.

Parameters:
fr frame to copy Duplicates a frame -- should only rarely be used, typically frisolate is good enough
Returns:
Returns a frame on success, NULL on error

Definition at line 465 of file frame.c.

References ast_calloc_cache, ast_copy_flags, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, AST_LIST_REMOVE_CURRENT, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, AST_MALLOCD_HDR, ast_threadstorage_get(), ast_frame::data, ast_frame::datalen, ast_frame::delivery, f, frame_cache, frames, ast_frame::frametype, ast_frame::len, len(), ast_frame::mallocd, ast_frame::mallocd_hdr_len, ast_frame::offset, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, and ast_frame::ts.

Referenced by __ast_queue_frame(), ast_jb_put(), ast_rtp_write(), ast_slinfactory_feed(), audiohook_read_frame_single(), autoservice_run(), recordthread(), and rpt().

00466 {
00467    struct ast_frame *out = NULL;
00468    int len, srclen = 0;
00469    void *buf = NULL;
00470 
00471 #if !defined(LOW_MEMORY)
00472    struct ast_frame_cache *frames;
00473 #endif
00474 
00475    /* Start with standard stuff */
00476    len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen;
00477    /* If we have a source, add space for it */
00478    /*
00479     * XXX Watch out here - if we receive a src which is not terminated
00480     * properly, we can be easily attacked. Should limit the size we deal with.
00481     */
00482    if (f->src)
00483       srclen = strlen(f->src);
00484    if (srclen > 0)
00485       len += srclen + 1;
00486    
00487 #if !defined(LOW_MEMORY)
00488    if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) {
00489       AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) {
00490          if (out->mallocd_hdr_len >= len) {
00491             size_t mallocd_len = out->mallocd_hdr_len;
00492             AST_LIST_REMOVE_CURRENT(&frames->list, frame_list);
00493             memset(out, 0, sizeof(*out));
00494             out->mallocd_hdr_len = mallocd_len;
00495             buf = out;
00496             frames->size--;
00497             break;
00498          }
00499       }
00500       AST_LIST_TRAVERSE_SAFE_END
00501    }
00502 #endif
00503 
00504    if (!buf) {
00505       if (!(buf = ast_calloc_cache(1, len)))
00506          return NULL;
00507       out = buf;
00508       out->mallocd_hdr_len = len;
00509    }
00510 
00511    out->frametype = f->frametype;
00512    out->subclass = f->subclass;
00513    out->datalen = f->datalen;
00514    out->samples = f->samples;
00515    out->delivery = f->delivery;
00516    /* Set us as having malloc'd header only, so it will eventually
00517       get freed. */
00518    out->mallocd = AST_MALLOCD_HDR;
00519    out->offset = AST_FRIENDLY_OFFSET;
00520    if (out->datalen) {
00521       out->data = buf + sizeof(*out) + AST_FRIENDLY_OFFSET;
00522       memcpy(out->data, f->data, out->datalen); 
00523    }
00524    if (srclen > 0) {
00525       /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */
00526       char *src;
00527       out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen;
00528       src = (char *) out->src;
00529       /* Must have space since we allocated for it */
00530       strcpy(src, f->src);
00531    }
00532    ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO);
00533    out->ts = f->ts;
00534    out->len = f->len;
00535    out->seqno = f->seqno;
00536    return out;
00537 }

struct ast_frame* ast_frisolate ( struct ast_frame fr  ) 

Makes a frame independent of any static storage.

Parameters:
fr frame to act upon Take a frame, and if it's not been malloc'd, make a malloc'd copy and if the data hasn't been malloced then make the data malloc'd. If you need to store frames, say for queueing, then you should call this function.
Returns:
Returns a frame on success, NULL on error

Definition at line 406 of file frame.c.

References ast_clear_flag, ast_copy_flags, ast_frame_header_new(), AST_FRFLAG_FROM_DSP, AST_FRFLAG_FROM_TRANSLATOR, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, ast_malloc, AST_MALLOCD_DATA, AST_MALLOCD_HDR, AST_MALLOCD_SRC, ast_strdup, ast_test_flag, ast_frame::data, ast_frame::datalen, ast_frame::frametype, free, ast_frame::len, ast_frame::mallocd, ast_frame::offset, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, and ast_frame::ts.

Referenced by jpeg_read_image().

00407 {
00408    struct ast_frame *out;
00409    void *newdata;
00410 
00411    ast_clear_flag(fr, AST_FRFLAG_FROM_TRANSLATOR);
00412    ast_clear_flag(fr, AST_FRFLAG_FROM_DSP);
00413 
00414    if (!(fr->mallocd & AST_MALLOCD_HDR)) {
00415       /* Allocate a new header if needed */
00416       if (!(out = ast_frame_header_new()))
00417          return NULL;
00418       out->frametype = fr->frametype;
00419       out->subclass = fr->subclass;
00420       out->datalen = fr->datalen;
00421       out->samples = fr->samples;
00422       out->offset = fr->offset;
00423       out->data = fr->data;
00424       /* Copy the timing data */
00425       ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO);
00426       if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) {
00427          out->ts = fr->ts;
00428          out->len = fr->len;
00429          out->seqno = fr->seqno;
00430       }
00431    } else
00432       out = fr;
00433    
00434    if (!(fr->mallocd & AST_MALLOCD_SRC)) {
00435       if (fr->src) {
00436          if (!(out->src = ast_strdup(fr->src))) {
00437             if (out != fr)
00438                free(out);
00439             return NULL;
00440          }
00441       }
00442    } else
00443       out->src = fr->src;
00444    
00445    if (!(fr->mallocd & AST_MALLOCD_DATA))  {
00446       if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) {
00447          if (out->src != fr->src)
00448             free((void *) out->src);
00449          if (out != fr)
00450             free(out);
00451          return NULL;
00452       }
00453       newdata += AST_FRIENDLY_OFFSET;
00454       out->offset = AST_FRIENDLY_OFFSET;
00455       out->datalen = fr->datalen;
00456       memcpy(newdata, fr->data, fr->datalen);
00457       out->data = newdata;
00458    }
00459 
00460    out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA;
00461    
00462    return out;
00463 }

struct ast_format_list* ast_get_format_list ( size_t *  size  ) 

Definition at line 555 of file frame.c.

References AST_FORMAT_LIST.

00556 {
00557    *size = (sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]));
00558    return AST_FORMAT_LIST;
00559 }

struct ast_format_list* ast_get_format_list_index ( int  index  ) 

Definition at line 550 of file frame.c.

References AST_FORMAT_LIST.

00551 {
00552    return &AST_FORMAT_LIST[index];
00553 }

int ast_getformatbyname ( const char *  name  ) 

Gets a format from a name.

Parameters:
name string of format
Returns:
This returns the form of the format in binary on success, 0 on error.

Definition at line 621 of file frame.c.

References ast_expand_codec_alias(), AST_FORMAT_LIST, and format.

Referenced by ast_parse_allow_disallow(), iax_template_parse(), reload_config(), and try_suggested_sip_codec().

00622 {
00623    int x, all, format = 0;
00624 
00625    all = strcasecmp(name, "all") ? 0 : 1;
00626    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
00627       if(AST_FORMAT_LIST[x].visible && (all || 
00628            !strcasecmp(AST_FORMAT_LIST[x].name,name) ||
00629            !strcasecmp(AST_FORMAT_LIST[x].name,ast_expand_codec_alias(name)))) {
00630          format |= AST_FORMAT_LIST[x].bits;
00631          if(!all)
00632             break;
00633       }
00634    }
00635 
00636    return format;
00637 }

char* ast_getformatname ( int  format  ) 

Get the name of a format.

Parameters:
format id of format
Returns:
A static string containing the name of the format or "unknown" if unknown.

Definition at line 561 of file frame.c.

References AST_FORMAT_LIST, ast_format_list::bits, name, and ast_format_list::visible.

Referenced by __ast_play_and_record(), __ast_read(), __ast_register_translator(), __login_exec(), _sip_show_peer(), add_codec_to_answer(), add_codec_to_sdp(), agent_call(), ast_codec_get_len(), ast_codec_get_samples(), ast_codec_pref_string(), ast_dsp_process(), ast_frame_dump(), ast_openvstream(), ast_rtp_write(), ast_slinfactory_feed(), ast_streamfile(), ast_translator_build_path(), ast_unregister_translator(), ast_writestream(), background_detect_exec(), dahdi_read(), do_waiting(), eagi_exec(), func_channel_read(), function_iaxpeer(), function_sippeer(), gtalk_show_channels(), iax2_request(), iax2_show_channels(), iax2_show_peer(), iax_show_provisioning(), moh_classes_show(), moh_release(), oh323_rtp_read(), phone_setup(), print_codec_to_cli(), rebuild_matrix(), register_translator(), set_format(), set_peer_capabilities(), show_codecs(), show_codecs_deprecated(), show_file_formats(), show_file_formats_deprecated(), show_image_formats(), show_image_formats_deprecated(), show_translation(), show_translation_deprecated(), sip_request_call(), sip_rtp_read(), and socket_process().

00562 {
00563    int x;
00564    char *ret = "unknown";
00565    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
00566       if(AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == format) {
00567          ret = AST_FORMAT_LIST[x].name;
00568          break;
00569       }
00570    }
00571    return ret;
00572 }

char* ast_getformatname_multiple ( char *  buf,
size_t  size,
int  format 
)

Get the names of a set of formats.

Parameters:
buf a buffer for the output string
size size of buf (bytes)
format the format (combined IDs of codecs) Prints a list of readable codec names corresponding to "format". ex: for format=AST_FORMAT_GSM|AST_FORMAT_SPEEX|AST_FORMAT_ILBC it will return "0x602 (GSM|SPEEX|ILBC)"
Returns:
The return value is buf.

Definition at line 574 of file frame.c.

References AST_FORMAT_LIST, ast_format_list::bits, len(), name, and ast_format_list::visible.

Referenced by __ast_read(), __sip_show_channels(), _sip_show_peer(), add_sdp(), ast_streamfile(), function_iaxpeer(), function_sippeer(), gtalk_is_answered(), gtalk_newcall(), handle_showchan(), handle_showchan_deprecated(), iax2_show_peer(), process_sdp(), serialize_showchan(), set_format(), sip_new(), sip_request_call(), sip_show_channel(), sip_show_settings(), and sip_write().

00575 {
00576    int x;
00577    unsigned len;
00578    char *start, *end = buf;
00579 
00580    if (!size)
00581       return buf;
00582    snprintf(end, size, "0x%x (", format);
00583    len = strlen(end);
00584    end += len;
00585    size -= len;
00586    start = end;
00587    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
00588       if (AST_FORMAT_LIST[x].visible && (AST_FORMAT_LIST[x].bits & format)) {
00589          snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name);
00590          len = strlen(end);
00591          end += len;
00592          size -= len;
00593       }
00594    }
00595    if (start == end)
00596       snprintf(start, size, "nothing)");
00597    else if (size > 1)
00598       *(end -1) = ')';
00599    return buf;
00600 }

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.

Definition at line 1337 of file frame.c.

References ast_codec_pref_append(), ast_codec_pref_remove(), ast_codec_pref_setsize(), AST_FORMAT_AUDIO_MASK, ast_getformatbyname(), ast_log(), ast_strdupa, format, LOG_DEBUG, LOG_WARNING, option_debug, and parse().

Referenced by action_originate(), apply_outgoing(), build_device(), build_peer(), build_user(), gtalk_create_member(), gtalk_load_config(), reload_config(), set_config(), and update_common_options().

01338 {
01339    char *parse = NULL, *this = NULL, *psize = NULL;
01340    int format = 0, framems = 0;
01341 
01342    parse = ast_strdupa(list);
01343    while ((this = strsep(&parse, ","))) {
01344       framems = 0;
01345       if ((psize = strrchr(this, ':'))) {
01346          *psize++ = '\0';
01347          if (option_debug)
01348             ast_log(LOG_DEBUG,"Packetization for codec: %s is %s\n", this, psize);
01349          framems = atoi(psize);
01350          if (framems < 0)
01351             framems = 0;
01352       }
01353       if (!(format = ast_getformatbyname(this))) {
01354          ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this);
01355          continue;
01356       }
01357 
01358       if (mask) {
01359          if (allowing)
01360             *mask |= format;
01361          else
01362             *mask &= ~format;
01363       }
01364 
01365       /* Set up a preference list for audio. Do not include video in preferences 
01366          since we can not transcode video and have to use whatever is offered
01367        */
01368       if (pref && (format & AST_FORMAT_AUDIO_MASK)) {
01369          if (strcasecmp(this, "all")) {
01370             if (allowing) {
01371                ast_codec_pref_append(pref, format);
01372                ast_codec_pref_setsize(pref, format, framems);
01373             }
01374             else
01375                ast_codec_pref_remove(pref, format);
01376          } else if (!allowing) {
01377             memset(pref, 0, sizeof(*pref));
01378          }
01379       }
01380    }
01381 }

void ast_smoother_free ( struct ast_smoother s  ) 

Definition at line 301 of file frame.c.

References free, and s.

Referenced by ast_rtp_destroy(), and ast_rtp_write().

00302 {
00303    free(s);
00304 }

int ast_smoother_get_flags ( struct ast_smoother smoother  ) 

Definition at line 201 of file frame.c.

References s.

00202 {
00203    return s->flags;
00204 }

struct ast_smoother* ast_smoother_new ( int  bytes  ) 

Definition at line 191 of file frame.c.

References ast_malloc, ast_smoother_reset(), and s.

Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().

00192 {
00193    struct ast_smoother *s;
00194    if (size < 1)
00195       return NULL;
00196    if ((s = ast_malloc(sizeof(*s))))
00197       ast_smoother_reset(s, size);
00198    return s;
00199 }

struct ast_frame* ast_smoother_read ( struct ast_smoother s  ) 

Definition at line 251 of file frame.c.

References AST_FRAME_VOICE, AST_FRIENDLY_OFFSET, ast_log(), ast_samp2tv(), AST_SMOOTHER_FLAG_G729, ast_tvadd(), ast_tvzero(), len(), LOG_WARNING, and s.

Referenced by ast_rtp_write().

00252 {
00253    struct ast_frame *opt;
00254    int len;
00255 
00256    /* IF we have an optimization frame, send it */
00257    if (s->opt) {
00258       if (s->opt->offset < AST_FRIENDLY_OFFSET)
00259          ast_log(LOG_WARNING, "Returning a frame of inappropriate offset (%d).\n",
00260                      s->opt->offset);
00261       opt = s->opt;
00262       s->opt = NULL;
00263       return opt;
00264    }
00265 
00266    /* Make sure we have enough data */
00267    if (s->len < s->size) {
00268       /* Or, if this is a G.729 frame with VAD on it, send it immediately anyway */
00269       if (!((s->flags & AST_SMOOTHER_FLAG_G729) && (s->size % 10)))
00270          return NULL;
00271    }
00272    len = s->size;
00273    if (len > s->len)
00274       len = s->len;
00275    /* Make frame */
00276    s->f.frametype = AST_FRAME_VOICE;
00277    s->f.subclass = s->format;
00278    s->f.data = s->framedata + AST_FRIENDLY_OFFSET;
00279    s->f.offset = AST_FRIENDLY_OFFSET;
00280    s->f.datalen = len;
00281    /* Samples will be improper given VAD, but with VAD the concept really doesn't even exist */
00282    s->f.samples = len * s->samplesperbyte;   /* XXX rounding */
00283    s->f.delivery = s->delivery;
00284    /* Fill Data */
00285    memcpy(s->f.data, s->data, len);
00286    s->len -= len;
00287    /* Move remaining data to the front if applicable */
00288    if (s->len) {
00289       /* In principle this should all be fine because if we are sending
00290          G.729 VAD, the next timestamp will take over anyawy */
00291       memmove(s->data, s->data + len, s->len);
00292       if (!ast_tvzero(s->delivery)) {
00293          /* If we have delivery time, increment it, otherwise, leave it at 0 */
00294          s->delivery = ast_tvadd(s->delivery, ast_samp2tv(s->f.samples, 8000));
00295       }
00296    }
00297    /* Return frame */
00298    return &s->f;
00299 }

void ast_smoother_reconfigure ( struct ast_smoother s,
int  bytes 
)

Reconfigure an existing smoother to output a different number of bytes per frame.

Parameters:
s the smoother to reconfigure
bytes the desired number of bytes per output frame
Returns:
nothing

Definition at line 169 of file frame.c.

References s, and smoother_frame_feed().

Referenced by ast_rtp_codec_setpref().

00170 {
00171    /* if there is no change, then nothing to do */
00172    if (s->size == bytes) {
00173       return;
00174    }
00175    /* set the new desired output size */
00176    s->size = bytes;
00177    /* if there is no 'optimized' frame in the smoother,
00178     *   then there is nothing left to do
00179     */
00180    if (!s->opt) {
00181       return;
00182    }
00183    /* there is an 'optimized' frame here at the old size,
00184     * but it must now be put into the buffer so the data
00185     * can be extracted at the new size
00186     */
00187    smoother_frame_feed(s, s->opt, s->opt_needs_swap);
00188    s->opt = NULL;
00189 }

void ast_smoother_reset ( struct ast_smoother s,
int  bytes 
)

Definition at line 163 of file frame.c.

References s.

Referenced by ast_smoother_new().

00164 {
00165    memset(s, 0, sizeof(*s));
00166    s->size = bytes;
00167 }

void ast_smoother_set_flags ( struct ast_smoother smoother,
int  flags 
)

Definition at line 206 of file frame.c.

References s.

Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().

00207 {
00208    s->flags = flags;
00209 }

int ast_smoother_test_flag ( struct ast_smoother s,
int  flag 
)

Definition at line 211 of file frame.c.

References s.

Referenced by ast_rtp_write().

00212 {
00213    return (s->flags & flag);
00214 }

void ast_swapcopy_samples ( void *  dst,
const void *  src,
int  samples 
)

Definition at line 539 of file frame.c.

Referenced by __ast_smoother_feed(), iax_frame_wrap(), phone_write_buf(), and smoother_frame_feed().

00540 {
00541    int i;
00542    unsigned short *dst_s = dst;
00543    const unsigned short *src_s = src;
00544 
00545    for (i = 0; i < samples; i++)
00546       dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8);
00547 }


Variable Documentation

struct ast_frame ast_null_frame

Queueing a null frame is fairly common, so we declare a global null frame object for this purpose instead of having to declare one on the stack

Definition at line 139 of file frame.c.

Referenced by __ast_read(), __oh323_rtp_create(), __oh323_update_info(), agent_new(), agent_read(), ast_channel_masquerade(), ast_channel_setwhentohangup(), ast_do_masquerade(), ast_rtcp_read(), ast_rtp_read(), ast_softhangup_nolock(), ast_udptl_read(), conf_run(), features_read(), gtalk_rtp_read(), handle_request_invite(), handle_response_invite(), local_read(), mgcp_rtp_read(), oh323_read(), oh323_rtp_read(), process_rfc2833(), process_sdp(), send_dtmf(), sip_read(), sip_rtp_read(), skinny_rtp_read(), and wakeup_sub().


Generated on Thu May 14 15:13:32 2009 for Asterisk - the Open Source PBX by  doxygen 1.4.7