Thu Oct 8 00:59:39 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 or list of frames.
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_raw_write(), au_seek(), convertcap(), g722tolin_sample(), pcm_read(), and rtp_get_rate().

#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_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 440 of file frame.h.

Referenced by ast_rtp_read(), and socket_process().

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

Definition at line 439 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_readvideo_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(), cli_audio_convert(), cli_audio_convert_deprecated(), 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 __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 __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 __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(), and rpt().

#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 499 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 501 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 502 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 211 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.

00212 {
00213    if (f->frametype != AST_FRAME_VOICE) {
00214       ast_log(LOG_WARNING, "Huh?  Can't smooth a non-voice frame!\n");
00215       return -1;
00216    }
00217    if (!s->format) {
00218       s->format = f->subclass;
00219       s->samplesperbyte = (float)f->samples / (float)f->datalen;
00220    } else if (s->format != f->subclass) {
00221       ast_log(LOG_WARNING, "Smoother was working on %d format frames, now trying to feed %d?\n", s->format, f->subclass);
00222       return -1;
00223    }
00224    if (s->len + f->datalen > SMOOTHER_SIZE) {
00225       ast_log(LOG_WARNING, "Out of smoother space\n");
00226       return -1;
00227    }
00228    if (((f->datalen == s->size) ||
00229         ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) &&
00230        !s->opt &&
00231        !s->len &&
00232        (f->offset >= AST_MIN_OFFSET)) {
00233       /* Optimize by sending the frame we just got
00234          on the next read, thus eliminating the douple
00235          copy */
00236       if (swap)
00237          ast_swapcopy_samples(f->data, f->data, f->samples);
00238       s->opt = f;
00239       s->opt_needs_swap = swap ? 1 : 0;
00240       return 0;
00241    }
00242 
00243    return smoother_frame_feed(s, f, swap);
00244 }

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 656 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().

00657 {
00658    int x;
00659    char *ret = "unknown";
00660    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
00661       if(AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == codec) {
00662          ret = AST_FORMAT_LIST[x].desc;
00663          break;
00664       }
00665    }
00666    return ret;
00667 }

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 1295 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().

01296 {
01297    int x, ret = 0, slot;
01298 
01299    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
01300       slot = pref->order[x];
01301 
01302       if (!slot)
01303          break;
01304       if (formats & AST_FORMAT_LIST[slot-1].bits) {
01305          ret = AST_FORMAT_LIST[slot-1].bits;
01306          break;
01307       }
01308    }
01309    if(ret & AST_FORMAT_AUDIO_MASK)
01310       return ret;
01311 
01312    if (option_debug > 3)
01313       ast_log(LOG_DEBUG, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec");
01314 
01315       return find_best ? ast_best_codec(formats) : 0;
01316 }

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 1554 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().

01555 {
01556    int len = 0;
01557 
01558    /* XXX Still need speex, g723, and lpc10 XXX */ 
01559    switch(format) {
01560    case AST_FORMAT_ILBC:
01561       len = (samples / 240) * 50;
01562       break;
01563    case AST_FORMAT_GSM:
01564       len = (samples / 160) * 33;
01565       break;
01566    case AST_FORMAT_G729A:
01567       len = samples / 8;
01568       break;
01569    case AST_FORMAT_SLINEAR:
01570       len = samples * 2;
01571       break;
01572    case AST_FORMAT_ULAW:
01573    case AST_FORMAT_ALAW:
01574       len = samples;
01575       break;
01576    case AST_FORMAT_G722:
01577    case AST_FORMAT_ADPCM:
01578    case AST_FORMAT_G726:
01579    case AST_FORMAT_G726_AAL2:
01580       len = samples / 2;
01581       break;
01582    default:
01583       ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format));
01584    }
01585 
01586    return len;
01587 }

int ast_codec_get_samples ( struct ast_frame f  ) 

Returns the number of samples contained in the frame.

Definition at line 1511 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().

01512 {
01513    int samples=0;
01514    switch(f->subclass) {
01515    case AST_FORMAT_SPEEX:
01516       samples = speex_samples(f->data, f->datalen);
01517       break;
01518    case AST_FORMAT_G723_1:
01519                 samples = g723_samples(f->data, f->datalen);
01520       break;
01521    case AST_FORMAT_ILBC:
01522       samples = 240 * (f->datalen / 50);
01523       break;
01524    case AST_FORMAT_GSM:
01525       samples = 160 * (f->datalen / 33);
01526       break;
01527    case AST_FORMAT_G729A:
01528       samples = f->datalen * 8;
01529       break;
01530    case AST_FORMAT_SLINEAR:
01531       samples = f->datalen / 2;
01532       break;
01533    case AST_FORMAT_LPC10:
01534                 /* assumes that the RTP packet contains one LPC10 frame */
01535       samples = 22 * 8;
01536       samples += (((char *)(f->data))[7] & 0x1) * 8;
01537       break;
01538    case AST_FORMAT_ULAW:
01539    case AST_FORMAT_ALAW:
01540       samples = f->datalen;
01541       break;
01542    case AST_FORMAT_G722:
01543    case AST_FORMAT_ADPCM:
01544    case AST_FORMAT_G726:
01545    case AST_FORMAT_G726_AAL2:
01546       samples = f->datalen * 2;
01547       break;
01548    default:
01549       ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass));
01550    }
01551    return samples;
01552 }

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

Gets duration in ms of interpolation frame for a format.

Definition at line 576 of file frame.h.

References AST_FORMAT_ILBC.

Referenced by __get_from_jb(), and jb_get_and_deliver().

00577 { 
00578    return (format == AST_FORMAT_ILBC) ? 30 : 20;
00579 }

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 1154 of file frame.c.

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

Referenced by ast_parse_allow_disallow().

01155 {
01156    int x, newindex = 0;
01157 
01158    ast_codec_pref_remove(pref, format);
01159 
01160    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
01161       if(AST_FORMAT_LIST[x].bits == format) {
01162          newindex = x + 1;
01163          break;
01164       }
01165    }
01166 
01167    if(newindex) {
01168       for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
01169          if(!pref->order[x]) {
01170             pref->order[x] = newindex;
01171             break;
01172          }
01173       }
01174    }
01175 
01176    return x;
01177 }

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 1056 of file frame.c.

References ast_codec_pref::order.

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

01057 {
01058    int x, differential = (int) 'A', mem;
01059    char *from, *to;
01060 
01061    if(right) {
01062       from = pref->order;
01063       to = buf;
01064       mem = size;
01065    } else {
01066       to = pref->order;
01067       from = buf;
01068       mem = 32;
01069    }
01070 
01071    memset(to, 0, mem);
01072    for (x = 0; x < 32 ; x++) {
01073       if(!from[x])
01074          break;
01075       to[x] = right ? (from[x] + differential) : (from[x] - differential);
01076    }
01077 }

struct ast_format_list ast_codec_pref_getsize ( struct ast_codec_pref pref,
int  format 
)

Get packet size for codec.

Definition at line 1256 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().

01257 {
01258    int x, index = -1, framems = 0;
01259    struct ast_format_list fmt = {0};
01260 
01261    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
01262       if(AST_FORMAT_LIST[x].bits == format) {
01263          fmt = AST_FORMAT_LIST[x];
01264          index = x;
01265          break;
01266       }
01267    }
01268 
01269    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
01270       if(pref->order[x] == (index + 1)) {
01271          framems = pref->framing[x];
01272          break;
01273       }
01274    }
01275 
01276    /* size validation */
01277    if(!framems)
01278       framems = AST_FORMAT_LIST[index].def_ms;
01279 
01280    if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */
01281       framems -= framems % AST_FORMAT_LIST[index].inc_ms;
01282 
01283    if(framems < AST_FORMAT_LIST[index].min_ms)
01284       framems = AST_FORMAT_LIST[index].min_ms;
01285 
01286    if(framems > AST_FORMAT_LIST[index].max_ms)
01287       framems = AST_FORMAT_LIST[index].max_ms;
01288 
01289    fmt.cur_ms = framems;
01290 
01291    return fmt;
01292 }

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 1114 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().

01115 {
01116    int slot = 0;
01117 
01118    
01119    if((index >= 0) && (index < sizeof(pref->order))) {
01120       slot = pref->order[index];
01121    }
01122 
01123    return slot ? AST_FORMAT_LIST[slot-1].bits : 0;
01124 }

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 1180 of file frame.c.

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

Referenced by create_addr().

01181 {
01182    int x, newindex = 0;
01183 
01184    /* First step is to get the codecs "index number" */
01185    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01186       if (AST_FORMAT_LIST[x].bits == format) {
01187          newindex = x + 1;
01188          break;
01189       }
01190    }
01191    /* Done if its unknown */
01192    if (!newindex)
01193       return;
01194 
01195    /* Now find any existing occurrence, or the end */
01196    for (x = 0; x < 32; x++) {
01197       if (!pref->order[x] || pref->order[x] == newindex)
01198          break;
01199    }
01200 
01201    if (only_if_existing && !pref->order[x])
01202       return;
01203 
01204    /* Move down to make space to insert - either all the way to the end,
01205       or as far as the existing location (which will be overwritten) */
01206    for (; x > 0; x--) {
01207       pref->order[x] = pref->order[x - 1];
01208       pref->framing[x] = pref->framing[x - 1];
01209    }
01210 
01211    /* And insert the new entry */
01212    pref->order[0] = newindex;
01213    pref->framing[0] = 0; /* ? */
01214 }

void ast_codec_pref_remove ( struct ast_codec_pref pref,
int  format 
)

Remove audio a codec from a preference list.

Definition at line 1127 of file frame.c.

References AST_FORMAT_LIST, and ast_codec_pref::order.

Referenced by ast_codec_pref_append(), and ast_parse_allow_disallow().

01128 {
01129    struct ast_codec_pref oldorder;
01130    int x, y = 0;
01131    int slot;
01132    int size;
01133 
01134    if(!pref->order[0])
01135       return;
01136 
01137    memcpy(&oldorder, pref, sizeof(oldorder));
01138    memset(pref, 0, sizeof(*pref));
01139 
01140    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
01141       slot = oldorder.order[x];
01142       size = oldorder.framing[x];
01143       if(! slot)
01144          break;
01145       if(AST_FORMAT_LIST[slot-1].bits != format) {
01146          pref->order[y] = slot;
01147          pref->framing[y++] = size;
01148       }
01149    }
01150    
01151 }

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

Set packet size for codec.

Definition at line 1217 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().

01218 {
01219    int x, index = -1;
01220 
01221    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
01222       if(AST_FORMAT_LIST[x].bits == format) {
01223          index = x;
01224          break;
01225       }
01226    }
01227 
01228    if(index < 0)
01229       return -1;
01230 
01231    /* size validation */
01232    if(!framems)
01233       framems = AST_FORMAT_LIST[index].def_ms;
01234 
01235    if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */
01236       framems -= framems % AST_FORMAT_LIST[index].inc_ms;
01237 
01238    if(framems < AST_FORMAT_LIST[index].min_ms)
01239       framems = AST_FORMAT_LIST[index].min_ms;
01240 
01241    if(framems > AST_FORMAT_LIST[index].max_ms)
01242       framems = AST_FORMAT_LIST[index].max_ms;
01243 
01244 
01245    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
01246       if(pref->order[x] == (index + 1)) {
01247          pref->framing[x] = framems;
01248          break;
01249       }
01250    }
01251 
01252    return x;
01253 }

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 1079 of file frame.c.

References ast_codec_pref_index(), and ast_getformatname().

Referenced by dump_prefs(), and socket_process().

01080 {
01081    int x, codec; 
01082    size_t total_len, slen;
01083    char *formatname;
01084    
01085    memset(buf,0,size);
01086    total_len = size;
01087    buf[0] = '(';
01088    total_len--;
01089    for(x = 0; x < 32 ; x++) {
01090       if(total_len <= 0)
01091          break;
01092       if(!(codec = ast_codec_pref_index(pref,x)))
01093          break;
01094       if((formatname = ast_getformatname(codec))) {
01095          slen = strlen(formatname);
01096          if(slen > total_len)
01097             break;
01098          strncat(buf, formatname, total_len - 1); /* safe */
01099          total_len -= slen;
01100       }
01101       if(total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) {
01102          strncat(buf, "|", total_len - 1); /* safe */
01103          total_len--;
01104       }
01105    }
01106    if(total_len) {
01107       strncat(buf, ")", total_len - 1); /* safe */
01108       total_len--;
01109    }
01110 
01111    return size - total_len;
01112 }

static force_inline int ast_format_rate ( int  format  )  [static]

Get the sample rate for a given format.

Definition at line 603 of file frame.h.

References AST_FORMAT_G722.

Referenced by __get_from_jb(), ast_read_generator_actions(), ast_readaudio_callback(), ast_readvideo_callback(), ast_rtp_read(), ast_translate(), calc_cost(), calc_timestamp(), generator_force(), rtp_get_rate(), and schedule_delivery().

00604 {
00605    if (format == AST_FORMAT_G722)
00606       return 16000;
00607 
00608    return 8000;
00609 }

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 1589 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().

01590 {
01591    int count;
01592    short *fdata = f->data;
01593    short adjust_value = abs(adjustment);
01594 
01595    if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR))
01596       return -1;
01597 
01598    if (!adjustment)
01599       return 0;
01600 
01601    for (count = 0; count < f->samples; count++) {
01602       if (adjustment > 0) {
01603          ast_slinear_saturated_multiply(&fdata[count], &adjust_value);
01604       } else if (adjustment < 0) {
01605          ast_slinear_saturated_divide(&fdata[count], &adjust_value);
01606       }
01607    }
01608 
01609    return 0;
01610 }

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

Dump a frame for debugging purposes

Definition at line 810 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().

00811 {
00812    const char noname[] = "unknown";
00813    char ftype[40] = "Unknown Frametype";
00814    char cft[80];
00815    char subclass[40] = "Unknown Subclass";
00816    char csub[80];
00817    char moreinfo[40] = "";
00818    char cn[60];
00819    char cp[40];
00820    char cmn[40];
00821 
00822    if (!name)
00823       name = noname;
00824 
00825 
00826    if (!f) {
00827       ast_verbose("%s [ %s (NULL) ] [%s]\n", 
00828          term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00829          term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 
00830          term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00831       return;
00832    }
00833    /* XXX We should probably print one each of voice and video when the format changes XXX */
00834    if (f->frametype == AST_FRAME_VOICE)
00835       return;
00836    if (f->frametype == AST_FRAME_VIDEO)
00837       return;
00838    switch(f->frametype) {
00839    case AST_FRAME_DTMF_BEGIN:
00840       strcpy(ftype, "DTMF Begin");
00841       subclass[0] = f->subclass;
00842       subclass[1] = '\0';
00843       break;
00844    case AST_FRAME_DTMF_END:
00845       strcpy(ftype, "DTMF End");
00846       subclass[0] = f->subclass;
00847       subclass[1] = '\0';
00848       break;
00849    case AST_FRAME_CONTROL:
00850       strcpy(ftype, "Control");
00851       switch(f->subclass) {
00852       case AST_CONTROL_HANGUP:
00853          strcpy(subclass, "Hangup");
00854          break;
00855       case AST_CONTROL_RING:
00856          strcpy(subclass, "Ring");
00857          break;
00858       case AST_CONTROL_RINGING:
00859          strcpy(subclass, "Ringing");
00860          break;
00861       case AST_CONTROL_ANSWER:
00862          strcpy(subclass, "Answer");
00863          break;
00864       case AST_CONTROL_BUSY:
00865          strcpy(subclass, "Busy");
00866          break;
00867       case AST_CONTROL_TAKEOFFHOOK:
00868          strcpy(subclass, "Take Off Hook");
00869          break;
00870       case AST_CONTROL_OFFHOOK:
00871          strcpy(subclass, "Line Off Hook");
00872          break;
00873       case AST_CONTROL_CONGESTION:
00874          strcpy(subclass, "Congestion");
00875          break;
00876       case AST_CONTROL_FLASH:
00877          strcpy(subclass, "Flash");
00878          break;
00879       case AST_CONTROL_WINK:
00880          strcpy(subclass, "Wink");
00881          break;
00882       case AST_CONTROL_OPTION:
00883          strcpy(subclass, "Option");
00884          break;
00885       case AST_CONTROL_RADIO_KEY:
00886          strcpy(subclass, "Key Radio");
00887          break;
00888       case AST_CONTROL_RADIO_UNKEY:
00889          strcpy(subclass, "Unkey Radio");
00890          break;
00891       case AST_CONTROL_PROGRESS:
00892          strcpy(subclass, "Call Progress");
00893          break;
00894       case AST_CONTROL_PROCEEDING:
00895          strcpy(subclass, "Proceeding");
00896          break;
00897       case AST_CONTROL_HOLD:
00898         strcpy(subclass, "Hold");
00899         break;
00900       case AST_CONTROL_UNHOLD:
00901         strcpy(subclass, "UnHold");
00902         break;
00903       case -1:
00904          strcpy(subclass, "Stop generators");
00905          break;
00906       default:
00907          snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass);
00908       }
00909       break;
00910    case AST_FRAME_NULL:
00911       strcpy(ftype, "Null Frame");
00912       strcpy(subclass, "N/A");
00913       break;
00914    case AST_FRAME_IAX:
00915       /* Should never happen */
00916       strcpy(ftype, "IAX Specific");
00917       snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass);
00918       break;
00919    case AST_FRAME_TEXT:
00920       strcpy(ftype, "Text");
00921       strcpy(subclass, "N/A");
00922       ast_copy_string(moreinfo, f->data, sizeof(moreinfo));
00923       break;
00924    case AST_FRAME_IMAGE:
00925       strcpy(ftype, "Image");
00926       snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass));
00927       break;
00928    case AST_FRAME_HTML:
00929       strcpy(ftype, "HTML");
00930       switch(f->subclass) {
00931       case AST_HTML_URL:
00932          strcpy(subclass, "URL");
00933          ast_copy_string(moreinfo, f->data, sizeof(moreinfo));
00934          break;
00935       case AST_HTML_DATA:
00936          strcpy(subclass, "Data");
00937          break;
00938       case AST_HTML_BEGIN:
00939          strcpy(subclass, "Begin");
00940          break;
00941       case AST_HTML_END:
00942          strcpy(subclass, "End");
00943          break;
00944       case AST_HTML_LDCOMPLETE:
00945          strcpy(subclass, "Load Complete");
00946          break;
00947       case AST_HTML_NOSUPPORT:
00948          strcpy(subclass, "No Support");
00949          break;
00950       case AST_HTML_LINKURL:
00951          strcpy(subclass, "Link URL");
00952          ast_copy_string(moreinfo, f->data, sizeof(moreinfo));
00953          break;
00954       case AST_HTML_UNLINK:
00955          strcpy(subclass, "Unlink");
00956          break;
00957       case AST_HTML_LINKREJECT:
00958          strcpy(subclass, "Link Reject");
00959          break;
00960       default:
00961          snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass);
00962          break;
00963       }
00964       break;
00965    case AST_FRAME_MODEM:
00966       strcpy(ftype, "Modem");
00967       switch (f->subclass) {
00968       case AST_MODEM_T38:
00969          strcpy(subclass, "T.38");
00970          break;
00971       case AST_MODEM_V150:
00972          strcpy(subclass, "V.150");
00973          break;
00974       default:
00975          snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass);
00976          break;
00977       }
00978       break;
00979    default:
00980       snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype);
00981    }
00982    if (!ast_strlen_zero(moreinfo))
00983       ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n",  
00984              term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00985              term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)),
00986              f->frametype, 
00987              term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)),
00988              f->subclass, 
00989              term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)),
00990              term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00991    else
00992       ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n",  
00993              term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00994              term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)),
00995              f->frametype, 
00996              term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)),
00997              f->subclass, 
00998              term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00999 }

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 or list of frames.

Parameters:
fr Frame to free, or head of list to free
cache Whether to consider this frame for frame caching

Definition at line 385 of file frame.c.

References __frame_free(), AST_LIST_NEXT, ast_frame::frame_list, and ast_frame::next.

Referenced by mixmonitor_thread().

00386 {
00387    struct ast_frame *next;
00388 
00389    for (next = AST_LIST_NEXT(frame, frame_list);
00390         frame;
00391         frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) {
00392       __frame_free(frame, cache);
00393    }
00394 }

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 1612 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.

01613 {
01614    int count;
01615    short *data1, *data2;
01616 
01617    if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR))
01618       return -1;
01619 
01620    if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR))
01621       return -1;
01622 
01623    if (f1->samples != f2->samples)
01624       return -1;
01625 
01626    for (count = 0, data1 = f1->data, data2 = f2->data;
01627         count < f1->samples;
01628         count++, data1++, data2++)
01629       ast_slinear_saturated_add(data1, data2);
01630 
01631    return 0;
01632 }

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 482 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_frisolate(), ast_jb_put(), ast_rtp_write(), ast_slinfactory_feed(), audiohook_read_frame_single(), autoservice_run(), recordthread(), and rpt().

00483 {
00484    struct ast_frame *out = NULL;
00485    int len, srclen = 0;
00486    void *buf = NULL;
00487 
00488 #if !defined(LOW_MEMORY)
00489    struct ast_frame_cache *frames;
00490 #endif
00491 
00492    /* Start with standard stuff */
00493    len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen;
00494    /* If we have a source, add space for it */
00495    /*
00496     * XXX Watch out here - if we receive a src which is not terminated
00497     * properly, we can be easily attacked. Should limit the size we deal with.
00498     */
00499    if (f->src)
00500       srclen = strlen(f->src);
00501    if (srclen > 0)
00502       len += srclen + 1;
00503    
00504 #if !defined(LOW_MEMORY)
00505    if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) {
00506       AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) {
00507          if (out->mallocd_hdr_len >= len) {
00508             size_t mallocd_len = out->mallocd_hdr_len;
00509             AST_LIST_REMOVE_CURRENT(&frames->list, frame_list);
00510             memset(out, 0, sizeof(*out));
00511             out->mallocd_hdr_len = mallocd_len;
00512             buf = out;
00513             frames->size--;
00514             break;
00515          }
00516       }
00517       AST_LIST_TRAVERSE_SAFE_END;
00518    }
00519 #endif
00520 
00521    if (!buf) {
00522       if (!(buf = ast_calloc_cache(1, len)))
00523          return NULL;
00524       out = buf;
00525       out->mallocd_hdr_len = len;
00526    }
00527 
00528    out->frametype = f->frametype;
00529    out->subclass = f->subclass;
00530    out->datalen = f->datalen;
00531    out->samples = f->samples;
00532    out->delivery = f->delivery;
00533    /* Set us as having malloc'd header only, so it will eventually
00534       get freed. */
00535    out->mallocd = AST_MALLOCD_HDR;
00536    out->offset = AST_FRIENDLY_OFFSET;
00537    if (out->datalen) {
00538       out->data = buf + sizeof(*out) + AST_FRIENDLY_OFFSET;
00539       memcpy(out->data, f->data, out->datalen); 
00540    }
00541    if (srclen > 0) {
00542       /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */
00543       char *src;
00544       out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen;
00545       src = (char *) out->src;
00546       /* Must have space since we allocated for it */
00547       strcpy(src, f->src);
00548    }
00549    ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO);
00550    out->ts = f->ts;
00551    out->len = f->len;
00552    out->seqno = f->seqno;
00553    return out;
00554 }

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
Note:
This function may modify the frame passed to it, so you must not assume the frame will be intact after the isolated frame has been produced. In other words, calling this function on a frame should be the last operation you do with that frame before freeing it (or exiting the block, if the frame is on the stack.)

Definition at line 401 of file frame.c.

References ast_clear_flag, ast_copy_flags, ast_frame_header_new(), ast_frdup(), AST_FRFLAG_FROM_DSP, AST_FRFLAG_FROM_FILESTREAM, 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 ast_slinfactory_feed(), autoservice_run(), and jpeg_read_image().

00402 {
00403    struct ast_frame *out;
00404    void *newdata;
00405 
00406    /* if none of the existing frame is malloc'd, let ast_frdup() do it
00407       since it is more efficient
00408    */
00409    if (fr->mallocd == 0) {
00410       return ast_frdup(fr);
00411    }
00412 
00413    /* if everything is already malloc'd, we are done */
00414    if ((fr->mallocd & (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) ==
00415        (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) {
00416       return fr;
00417    }
00418 
00419    if (!(fr->mallocd & AST_MALLOCD_HDR)) {
00420       /* Allocate a new header if needed */
00421       if (!(out = ast_frame_header_new())) {
00422          return NULL;
00423       }
00424       out->frametype = fr->frametype;
00425       out->subclass = fr->subclass;
00426       out->datalen = fr->datalen;
00427       out->samples = fr->samples;
00428       out->offset = fr->offset;
00429       /* Copy the timing data */
00430       ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO);
00431       if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) {
00432          out->ts = fr->ts;
00433          out->len = fr->len;
00434          out->seqno = fr->seqno;
00435       }
00436    } else {
00437       ast_clear_flag(fr, AST_FRFLAG_FROM_TRANSLATOR);
00438       ast_clear_flag(fr, AST_FRFLAG_FROM_DSP);
00439       ast_clear_flag(fr, AST_FRFLAG_FROM_FILESTREAM);
00440       out = fr;
00441    }
00442    
00443    if (!(fr->mallocd & AST_MALLOCD_SRC) && fr->src) {
00444       if (!(out->src = ast_strdup(fr->src))) {
00445          if (out != fr) {
00446             free(out);
00447          }
00448          return NULL;
00449       }
00450    } else {
00451       out->src = fr->src;
00452       fr->src = NULL;
00453       fr->mallocd &= ~AST_MALLOCD_SRC;
00454    }
00455    
00456    if (!(fr->mallocd & AST_MALLOCD_DATA))  {
00457       if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) {
00458          if (out->src != fr->src) {
00459             free((void *) out->src);
00460          }
00461          if (out != fr) {
00462             free(out);
00463          }
00464          return NULL;
00465       }
00466       newdata += AST_FRIENDLY_OFFSET;
00467       out->offset = AST_FRIENDLY_OFFSET;
00468       out->datalen = fr->datalen;
00469       memcpy(newdata, fr->data, fr->datalen);
00470       out->data = newdata;
00471    } else {
00472       out->data = fr->data;
00473       fr->data = NULL;
00474       fr->mallocd &= ~AST_MALLOCD_DATA;
00475    }
00476 
00477    out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA;
00478    
00479    return out;
00480 }

struct ast_format_list* ast_get_format_list ( size_t *  size  ) 

Definition at line 572 of file frame.c.

References AST_FORMAT_LIST.

00573 {
00574    *size = (sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]));
00575    return AST_FORMAT_LIST;
00576 }

struct ast_format_list* ast_get_format_list_index ( int  index  ) 

Definition at line 567 of file frame.c.

References AST_FORMAT_LIST.

00568 {
00569    return &AST_FORMAT_LIST[index];
00570 }

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 638 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().

00639 {
00640    int x, all, format = 0;
00641 
00642    all = strcasecmp(name, "all") ? 0 : 1;
00643    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
00644       if(AST_FORMAT_LIST[x].visible && (all || 
00645            !strcasecmp(AST_FORMAT_LIST[x].name,name) ||
00646            !strcasecmp(AST_FORMAT_LIST[x].name,ast_expand_codec_alias(name)))) {
00647          format |= AST_FORMAT_LIST[x].bits;
00648          if(!all)
00649             break;
00650       }
00651    }
00652 
00653    return format;
00654 }

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 578 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_local_capabilities(), 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().

00579 {
00580    int x;
00581    char *ret = "unknown";
00582    for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
00583       if(AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == format) {
00584          ret = AST_FORMAT_LIST[x].name;
00585          break;
00586       }
00587    }
00588    return ret;
00589 }

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 591 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().

00592 {
00593    int x;
00594    unsigned len;
00595    char *start, *end = buf;
00596 
00597    if (!size)
00598       return buf;
00599    snprintf(end, size, "0x%x (", format);
00600    len = strlen(end);
00601    end += len;
00602    size -= len;
00603    start = end;
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 & format)) {
00606          snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name);
00607          len = strlen(end);
00608          end += len;
00609          size -= len;
00610       }
00611    }
00612    if (start == end)
00613       snprintf(start, size, "nothing)");
00614    else if (size > 1)
00615       *(end -1) = ')';
00616    return buf;
00617 }

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 1318 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().

01319 {
01320    char *parse = NULL, *this = NULL, *psize = NULL;
01321    int format = 0, framems = 0;
01322 
01323    parse = ast_strdupa(list);
01324    while ((this = strsep(&parse, ","))) {
01325       framems = 0;
01326       if ((psize = strrchr(this, ':'))) {
01327          *psize++ = '\0';
01328          if (option_debug)
01329             ast_log(LOG_DEBUG,"Packetization for codec: %s is %s\n", this, psize);
01330          framems = atoi(psize);
01331          if (framems < 0)
01332             framems = 0;
01333       }
01334       if (!(format = ast_getformatbyname(this))) {
01335          ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this);
01336          continue;
01337       }
01338 
01339       if (mask) {
01340          if (allowing)
01341             *mask |= format;
01342          else
01343             *mask &= ~format;
01344       }
01345 
01346       /* Set up a preference list for audio. Do not include video in preferences 
01347          since we can not transcode video and have to use whatever is offered
01348        */
01349       if (pref && (format & AST_FORMAT_AUDIO_MASK)) {
01350          if (strcasecmp(this, "all")) {
01351             if (allowing) {
01352                ast_codec_pref_append(pref, format);
01353                ast_codec_pref_setsize(pref, format, framems);
01354             }
01355             else
01356                ast_codec_pref_remove(pref, format);
01357          } else if (!allowing) {
01358             memset(pref, 0, sizeof(*pref));
01359          }
01360       }
01361    }
01362 }

void ast_smoother_free ( struct ast_smoother s  ) 

Definition at line 296 of file frame.c.

References free, and s.

Referenced by ast_rtp_destroy(), and ast_rtp_write().

00297 {
00298    free(s);
00299 }

int ast_smoother_get_flags ( struct ast_smoother smoother  ) 

Definition at line 196 of file frame.c.

References s.

00197 {
00198    return s->flags;
00199 }

struct ast_smoother* ast_smoother_new ( int  bytes  ) 

Definition at line 186 of file frame.c.

References ast_malloc, ast_smoother_reset(), and s.

Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().

00187 {
00188    struct ast_smoother *s;
00189    if (size < 1)
00190       return NULL;
00191    if ((s = ast_malloc(sizeof(*s))))
00192       ast_smoother_reset(s, size);
00193    return s;
00194 }

struct ast_frame* ast_smoother_read ( struct ast_smoother s  ) 

Definition at line 246 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().

00247 {
00248    struct ast_frame *opt;
00249    int len;
00250 
00251    /* IF we have an optimization frame, send it */
00252    if (s->opt) {
00253       if (s->opt->offset < AST_FRIENDLY_OFFSET)
00254          ast_log(LOG_WARNING, "Returning a frame of inappropriate offset (%d).\n",
00255                      s->opt->offset);
00256       opt = s->opt;
00257       s->opt = NULL;
00258       return opt;
00259    }
00260 
00261    /* Make sure we have enough data */
00262    if (s->len < s->size) {
00263       /* Or, if this is a G.729 frame with VAD on it, send it immediately anyway */
00264       if (!((s->flags & AST_SMOOTHER_FLAG_G729) && (s->len % 10)))
00265          return NULL;
00266    }
00267    len = s->size;
00268    if (len > s->len)
00269       len = s->len;
00270    /* Make frame */
00271    s->f.frametype = AST_FRAME_VOICE;
00272    s->f.subclass = s->format;
00273    s->f.data = s->framedata + AST_FRIENDLY_OFFSET;
00274    s->f.offset = AST_FRIENDLY_OFFSET;
00275    s->f.datalen = len;
00276    /* Samples will be improper given VAD, but with VAD the concept really doesn't even exist */
00277    s->f.samples = len * s->samplesperbyte;   /* XXX rounding */
00278    s->f.delivery = s->delivery;
00279    /* Fill Data */
00280    memcpy(s->f.data, s->data, len);
00281    s->len -= len;
00282    /* Move remaining data to the front if applicable */
00283    if (s->len) {
00284       /* In principle this should all be fine because if we are sending
00285          G.729 VAD, the next timestamp will take over anyawy */
00286       memmove(s->data, s->data + len, s->len);
00287       if (!ast_tvzero(s->delivery)) {
00288          /* If we have delivery time, increment it, otherwise, leave it at 0 */
00289          s->delivery = ast_tvadd(s->delivery, ast_samp2tv(s->f.samples, 8000));
00290       }
00291    }
00292    /* Return frame */
00293    return &s->f;
00294 }

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 164 of file frame.c.

References s, and smoother_frame_feed().

Referenced by ast_rtp_codec_setpref().

00165 {
00166    /* if there is no change, then nothing to do */
00167    if (s->size == bytes) {
00168       return;
00169    }
00170    /* set the new desired output size */
00171    s->size = bytes;
00172    /* if there is no 'optimized' frame in the smoother,
00173     *   then there is nothing left to do
00174     */
00175    if (!s->opt) {
00176       return;
00177    }
00178    /* there is an 'optimized' frame here at the old size,
00179     * but it must now be put into the buffer so the data
00180     * can be extracted at the new size
00181     */
00182    smoother_frame_feed(s, s->opt, s->opt_needs_swap);
00183    s->opt = NULL;
00184 }

void ast_smoother_reset ( struct ast_smoother s,
int  bytes 
)

Definition at line 158 of file frame.c.

References s.

Referenced by ast_smoother_new().

00159 {
00160    memset(s, 0, sizeof(*s));
00161    s->size = bytes;
00162 }

void ast_smoother_set_flags ( struct ast_smoother smoother,
int  flags 
)

Definition at line 201 of file frame.c.

References s.

Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().

00202 {
00203    s->flags = flags;
00204 }

int ast_smoother_test_flag ( struct ast_smoother s,
int  flag 
)

Definition at line 206 of file frame.c.

References s.

Referenced by ast_rtp_write().

00207 {
00208    return (s->flags & flag);
00209 }

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

Definition at line 556 of file frame.c.

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

00557 {
00558    int i;
00559    unsigned short *dst_s = dst;
00560    const unsigned short *src_s = src;
00561 
00562    for (i = 0; i < samples; i++)
00563       dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8);
00564 }


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 134 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 Oct 8 00:59:39 2009 for Asterisk - the Open Source PBX by  doxygen 1.4.7