Wed Mar 3 22:56:31 2010

Asterisk developer's documentation


frame.h File Reference

Asterisk internal frame definitions. More...

#include <sys/time.h>
#include "asterisk/endian.h"
#include "asterisk/linkedlists.h"
Include dependency graph for frame.h:
This graph shows which files directly or indirectly include this file:

Go to the source code of this file.

Data Structures

struct  ast_codec_pref
struct  ast_control_t38_parameters
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))
#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_TEXT   (1 << 28))
#define AST_FORMAT_MP4_VIDEO   (1 << 22)
#define AST_FORMAT_PNG   (1 << 17)
#define AST_FORMAT_SLINEAR   (1 << 6)
#define AST_FORMAT_SLINEAR16   (1 << 15)
#define AST_FORMAT_SPEEX   (1 << 9)
#define AST_FORMAT_T140   (1 << 27)
#define AST_FORMAT_T140RED   (1 << 26)
#define AST_FORMAT_TEXT_MASK   (((1 << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK))
#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 { ; } while(0)
#define ast_frame_byteswap_le(fr)   do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data.ptr, __f->data.ptr, __f->samples); } 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
 Offset into a frame's data buffer.
#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_T38_STATE   10
#define AST_OPTION_TDD   2
#define AST_OPTION_TONE_VERIFY   1
#define AST_OPTION_TXGAIN   5
#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, _XXX_AST_CONTROL_T38 = 19, AST_CONTROL_SRCUPDATE = 20,
  AST_CONTROL_T38_PARAMETERS = 24
}
enum  ast_control_t38 {
  AST_T38_REQUEST_NEGOTIATE = 1, AST_T38_REQUEST_TERMINATE, AST_T38_NEGOTIATED, AST_T38_TERMINATED,
  AST_T38_REFUSED
}
enum  ast_control_t38_rate {
  AST_T38_RATE_2400 = 0, AST_T38_RATE_4800, AST_T38_RATE_7200, AST_T38_RATE_9600,
  AST_T38_RATE_12000, AST_T38_RATE_14400
}
enum  ast_control_t38_rate_management { AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF = 0, AST_T38_RATE_MANAGEMENT_LOCAL_TCF }
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

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.
struct 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.
void ast_codec_pref_init (struct ast_codec_pref *pref)
 Initialize an audio codec preference to "no preference".
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)
struct 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.
int ast_frame_slinear_sum (struct ast_frame *f1, struct ast_frame *f2)
 Sums two frames of audio samples.
struct ast_frameast_frdup (const struct ast_frame *fr)
 Copies a frame.
struct ast_frameast_frisolate (struct ast_frame *fr)
 Makes a frame independent of any static storage.
struct ast_format_listast_get_format_list (size_t *size)
struct 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.
int 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_swapcopy_samples (void *dst, const void *src, int samples)

Variables

struct ast_frame ast_null_frame

AST_Smoother



#define ast_smoother_feed(s, f)   __ast_smoother_feed(s, f, 0)
#define ast_smoother_feed_be(s, f)   __ast_smoother_feed(s, f, 0)
#define ast_smoother_feed_le(s, f)   __ast_smoother_feed(s, f, 1)
int __ast_smoother_feed (struct ast_smoother *s, struct ast_frame *f, int swap)
void ast_smoother_free (struct ast_smoother *s)
int ast_smoother_get_flags (struct ast_smoother *smoother)
struct ast_smootherast_smoother_new (int bytes)
struct 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)

Detailed Description

Asterisk internal frame definitions.

Definition in file frame.h.


Define Documentation

#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))

Unsupported audio bits

Definition at line 271 of file frame.h.

#define AST_FORMAT_G722   (1 << 12)
#define AST_FORMAT_G723_1   (1 << 0)
#define AST_FORMAT_G726   (1 << 11)

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

Definition at line 267 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)
#define AST_FORMAT_G729A   (1 << 8)
#define AST_FORMAT_GSM   (1 << 1)
#define AST_FORMAT_H261   (1 << 18)

H.261 Video

Definition at line 281 of file frame.h.

Referenced by codec_ast2skinny(), codec_skinny2ast(), and h261_encap().

#define AST_FORMAT_H263   (1 << 19)

H.263 Video

Definition at line 283 of file frame.h.

Referenced by codec_ast2skinny(), codec_skinny2ast(), h263_encap(), h263_read(), and h263_write().

#define AST_FORMAT_H263_PLUS   (1 << 20)

H.263+ Video

Definition at line 285 of file frame.h.

Referenced by h263p_encap().

#define AST_FORMAT_H264   (1 << 21)

H.264 Video

Definition at line 287 of file frame.h.

Referenced by h264_encap(), h264_read(), and h264_write().

#define AST_FORMAT_ILBC   (1 << 10)
#define AST_FORMAT_JPEG   (1 << 16)

JPEG Images

Definition at line 277 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 259 of file frame.h.

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

#define AST_FORMAT_MAX_TEXT   (1 << 28))

Maximum text mask

Definition at line 296 of file frame.h.

#define AST_FORMAT_MP4_VIDEO   (1 << 22)

MPEG4 Video

Definition at line 289 of file frame.h.

Referenced by mpeg4_encap().

#define AST_FORMAT_PNG   (1 << 17)

PNG Images

Definition at line 279 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 257 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_helper(), ast_channel_start_silence_generator(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_call_progress(), ast_dsp_noise(), ast_dsp_process(), ast_dsp_silence(), ast_frame_adjust_volume(), ast_frame_slinear_sum(), ast_rtp_read(), ast_slinfactory_feed(), ast_speech_new(), 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(), fax_generator_generate(), find_transcoders(), handle_jack_audio(), handle_recordfile(), handle_speechcreate(), handle_speechrecognize(), iax_frame_wrap(), ices_exec(), init_outgoing(), is_encoder(), isAnsweringMachine(), jack_exec(), jack_hook_callback(), linear_alloc(), linear_generator(), lintoadpcm_sample(), lintoalaw_sample(), lintog722_sample(), lintog726_sample(), lintogsm_sample(), lintoilbc_sample(), lintolpc10_sample(), lintospeex_sample(), lintoulaw_sample(), load_module(), load_moh_classes(), local_ast_moh_start(), 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(), playtones_generator(), read_config(), record_exec(), rpt(), rpt_call(), rpt_exec(), rpt_tele_thread(), send_waveform_to_channel(), silence_generator_generate(), slin8_to_slin16_sample(), slinear_read(), slinear_write(), socket_process(), speech_background(), spy_generate(), tonepair_alloc(), tonepair_generator(), transmit_audio(), usbradio_new(), usbradio_read(), usbradio_request(), wav_read(), and wav_write().

#define AST_FORMAT_SLINEAR16   (1 << 15)
#define AST_FORMAT_SPEEX   (1 << 9)

SpeeX Free Compression

Definition at line 263 of file frame.h.

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

#define AST_FORMAT_T140   (1 << 27)

T.140 Text format - ITU T.140, RFC 4103

Definition at line 294 of file frame.h.

Referenced by add_tcodec_to_sdp(), ast_rtp_read(), and ast_write().

#define AST_FORMAT_T140RED   (1 << 26)

T.140 RED Text format RFC 4103

Definition at line 292 of file frame.h.

Referenced by add_tcodec_to_sdp(), ast_rtp_read(), process_sdp(), and rtp_red_init().

#define AST_FORMAT_TEXT_MASK   (((1 << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK))

Definition at line 297 of file frame.h.

Referenced by add_sdp(), ast_request(), check_peer_ok(), sip_new(), and sip_rtp_read().

#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 { ; } while(0)

Definition at line 497 of file frame.h.

Referenced by ast_rtp_read(), and socket_process().

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

Definition at line 496 of file frame.h.

Referenced by phone_read().

#define AST_FRAME_DTMF   AST_FRAME_DTMF_END
#define AST_FRAME_SET_BUFFER ( fr,
_base,
_ofs,
_datalen   ) 
Value:
{              \
   (fr)->data.ptr = (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 186 of file frame.h.

Referenced by fax_generator_generate(), g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), ilbc_read(), ogg_vorbis_read(), pcm_read(), slinear_read(), t38_tx_packet_handler(), vox_read(), and wav_read().

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

Definition at line 464 of file frame.h.

Referenced by __adsi_transmit_messages(), __ast_answer(), __ast_play_and_record(), __ast_queue_frame(), __ast_read(), __ast_request_and_dial(), adsi_careful_send(), agent_ack_sleep(), agent_read(), ast_audiohook_read_frame(), ast_autoservice_stop(), ast_bridge_call(), ast_channel_free(), ast_dsp_process(), ast_feature_request_and_dial(), ast_generic_bridge(), 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(), builtin_atxfer(), 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_idle_thread(), do_waiting(), echo_exec(), eivr_comm(), find_cache(), gen_generate(), handle_cli_file_convert(), handle_invite_replaces(), handle_recordfile(), handle_speechrecognize(), iax2_bridge(), iax_park_thread(), ices_exec(), isAnsweringMachine(), jack_exec(), jb_empty_and_reset_adaptive(), jb_empty_and_reset_fixed(), jb_get_and_deliver(), launch_asyncagi(), manage_parkinglot(), masq_park_call(), measurenoise(), moh_files_generator(), monitor_dial(), mp3_exec(), NBScat_exec(), receive_dtmf_digits(), record_exec(), recordthread(), rpt(), rpt_exec(), run_agi(), send_tone_burst(), send_waveform_to_channel(), sendurl_exec(), speech_background(), spy_generate(), ss_thread(), transmit_audio(), transmit_t38(), wait_for_answer(), wait_for_hangup(), wait_for_winner(), waitforring_exec(), and waitstream_core().

#define AST_FRIENDLY_OFFSET   64

Offset into a frame's data buffer.

By providing some "empty" space prior to the actual data of an ast_frame, this gives any consumer of the frame ample space to prepend other necessary information without having to create a new buffer.

As an example, RTP can use the data from an ast_frame and simply prepend the RTP header information into the space provided by AST_FRIENDLY_OFFSET instead of having to create a new buffer with the necessary space allocated.

Definition at line 207 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(), fax_generator_generate(), 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(), playtones_generator(), process_rfc3389(), send_tone_burst(), send_waveform_to_channel(), slinear_read(), sms_generate(), tonepair_generator(), usbradio_read(), vox_read(), and wav_read().

#define AST_HTML_BEGIN   4

Beginning frame

Definition at line 229 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_DATA   2

Data frame

Definition at line 227 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_END   8

End frame

Definition at line 231 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_LDCOMPLETE   16

Load is complete

Definition at line 233 of file frame.h.

Referenced by ast_frame_dump(), and sendurl_exec().

#define AST_HTML_LINKREJECT   20

Reject link request

Definition at line 241 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_LINKURL   18

Send URL, and track

Definition at line 237 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_NOSUPPORT   17

Peer is unable to support HTML

Definition at line 235 of file frame.h.

Referenced by ast_frame_dump(), and sendurl_exec().

#define AST_HTML_UNLINK   19

No more HTML linkage

Definition at line 239 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_URL   1

Sending a URL

Definition at line 225 of file frame.h.

Referenced by ast_channel_sendurl(), ast_frame_dump(), and sip_sendhtml().

#define AST_MALLOCD_DATA   (1 << 1)

Need the data be free'd?

Definition at line 213 of file frame.h.

Referenced by __frame_free(), ast_frisolate(), and create_video_frame().

#define AST_MALLOCD_HDR   (1 << 0)

Need the header be free'd?

Definition at line 211 of file frame.h.

Referenced by __frame_free(), ast_frame_header_new(), ast_frdup(), ast_frisolate(), and create_video_frame().

#define AST_MALLOCD_SRC   (1 << 2)

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

Definition at line 215 of file frame.h.

Referenced by __frame_free(), and ast_frisolate().

#define AST_MIN_OFFSET   32

Definition at line 208 of file frame.h.

Referenced by __ast_smoother_feed().

#define AST_MODEM_T38   1

T.38 Fax-over-IP

Definition at line 219 of file frame.h.

Referenced by ast_frame_dump(), ast_udptl_write(), t38_tx_packet_handler(), transmit_t38(), and udptl_rx_packet().

#define AST_MODEM_V150   2

V.150 Modem-over-IP

Definition at line 221 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 378 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 400 of file frame.h.

Referenced by dahdi_setoption().

#define AST_OPTION_FLAG_ACCEPT   1

Definition at line 361 of file frame.h.

#define AST_OPTION_FLAG_ANSWER   5

Definition at line 364 of file frame.h.

#define AST_OPTION_FLAG_QUERY   4

Definition at line 363 of file frame.h.

#define AST_OPTION_FLAG_REJECT   2

Definition at line 362 of file frame.h.

#define AST_OPTION_FLAG_REQUEST   0

Definition at line 360 of file frame.h.

Referenced by ast_bridge_call(), and iax2_setoption().

#define AST_OPTION_FLAG_WTF   6

Definition at line 365 of file frame.h.

#define AST_OPTION_OPRMODE   7

Definition at line 397 of file frame.h.

Referenced by dahdi_setoption(), and dial_exec_full().

#define AST_OPTION_RELAXDTMF   3

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

Definition at line 375 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 394 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_T38_STATE   10

Definition at line 406 of file frame.h.

Referenced by ast_channel_get_t38_state(), and sip_queryoption().

#define AST_OPTION_TDD   2

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

Definition at line 372 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 369 of file frame.h.

Referenced by conf_run(), dahdi_hangup(), dahdi_setoption(), rpt(), rpt_exec(), 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 386 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 567 of file frame.h.

Referenced by ast_rtp_write().

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

Definition at line 572 of file frame.h.

Referenced by ast_rtp_write().

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

Definition at line 573 of file frame.h.

#define AST_SMOOTHER_FLAG_BE   (1 << 1)

Definition at line 357 of file frame.h.

Referenced by ast_rtp_write().

#define AST_SMOOTHER_FLAG_G729   (1 << 0)

Definition at line 356 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 126 of file frame.h.

00126      {
00127    /*! This frame contains valid timing information */
00128    AST_FRFLAG_HAS_TIMING_INFO = (1 << 0),
00129    /*! This frame came from a translator and is still the original frame.
00130     *  The translator can not be free'd if the frame inside of it still has
00131     *  this flag set. */
00132    AST_FRFLAG_FROM_TRANSLATOR = (1 << 1),
00133    /*! This frame came from a dsp and is still the original frame.
00134     *  The dsp cannot be free'd if the frame inside of it still has
00135     *  this flag set. */
00136    AST_FRFLAG_FROM_DSP = (1 << 2),
00137    /*! This frame came from a filestream and is still the original frame.
00138     *  The filestream cannot be free'd if the frame inside of it still has
00139     *  this flag set. */
00140    AST_FRFLAG_FROM_FILESTREAM = (1 << 3),
00141 };

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

_XXX_AST_CONTROL_T38 

T38 state change request/notification

Deprecated:
This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead.
AST_CONTROL_SRCUPDATE 

Indicate source of media has changed

AST_CONTROL_T38_PARAMETERS 

T38 state change request/notification with parameters

Definition at line 299 of file frame.h.

00299                             {
00300    AST_CONTROL_HANGUP = 1,    /*!< Other end has hungup */
00301    AST_CONTROL_RING = 2,      /*!< Local ring */
00302    AST_CONTROL_RINGING = 3,   /*!< Remote end is ringing */
00303    AST_CONTROL_ANSWER = 4,    /*!< Remote end has answered */
00304    AST_CONTROL_BUSY = 5,      /*!< Remote end is busy */
00305    AST_CONTROL_TAKEOFFHOOK = 6,  /*!< Make it go off hook */
00306    AST_CONTROL_OFFHOOK = 7,   /*!< Line is off hook */
00307    AST_CONTROL_CONGESTION = 8,   /*!< Congestion (circuits busy) */
00308    AST_CONTROL_FLASH = 9,     /*!< Flash hook */
00309    AST_CONTROL_WINK = 10,     /*!< Wink */
00310    AST_CONTROL_OPTION = 11,   /*!< Set a low-level option */
00311    AST_CONTROL_RADIO_KEY = 12,   /*!< Key Radio */
00312    AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */
00313    AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */
00314    AST_CONTROL_PROCEEDING = 15,  /*!< Indicate CALL PROCEEDING */
00315    AST_CONTROL_HOLD = 16,     /*!< Indicate call is placed on hold */
00316    AST_CONTROL_UNHOLD = 17,   /*!< Indicate call is left from hold */
00317    AST_CONTROL_VIDUPDATE = 18,   /*!< Indicate video frame update */
00318    _XXX_AST_CONTROL_T38 = 19, /*!< T38 state change request/notification \deprecated This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead. */
00319    AST_CONTROL_SRCUPDATE = 20,     /*!< Indicate source of media has changed */
00320    AST_CONTROL_T38_PARAMETERS = 24, /*!< T38 state change request/notification with parameters */
00321 };

Enumerator:
AST_T38_REQUEST_NEGOTIATE 

Request T38 on a channel (voice to fax)

AST_T38_REQUEST_TERMINATE 

Terminate T38 on a channel (fax to voice)

AST_T38_NEGOTIATED 

T38 negotiated (fax mode)

AST_T38_TERMINATED 

T38 terminated (back to voice)

AST_T38_REFUSED 

T38 refused for some reason (usually rejected by remote end)

Definition at line 323 of file frame.h.

00323                      {
00324    AST_T38_REQUEST_NEGOTIATE = 1,   /*!< Request T38 on a channel (voice to fax) */
00325    AST_T38_REQUEST_TERMINATE, /*!< Terminate T38 on a channel (fax to voice) */
00326    AST_T38_NEGOTIATED,     /*!< T38 negotiated (fax mode) */
00327    AST_T38_TERMINATED,     /*!< T38 terminated (back to voice) */
00328    AST_T38_REFUSED         /*!< T38 refused for some reason (usually rejected by remote end) */
00329 };

Enumerator:
AST_T38_RATE_2400 
AST_T38_RATE_4800 
AST_T38_RATE_7200 
AST_T38_RATE_9600 
AST_T38_RATE_12000 
AST_T38_RATE_14400 

Definition at line 331 of file frame.h.

Enumerator:
AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF 
AST_T38_RATE_MANAGEMENT_LOCAL_TCF 

Definition at line 340 of file frame.h.

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. This is because these constants are transmitted directly over IAX2.
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 97 of file frame.h.

00097                     {
00098    /*! DTMF end event, subclass is the digit */
00099    AST_FRAME_DTMF_END = 1,
00100    /*! Voice data, subclass is AST_FORMAT_* */
00101    AST_FRAME_VOICE,
00102    /*! Video frame, maybe?? :) */
00103    AST_FRAME_VIDEO,
00104    /*! A control frame, subclass is AST_CONTROL_* */
00105    AST_FRAME_CONTROL,
00106    /*! An empty, useless frame */
00107    AST_FRAME_NULL,
00108    /*! Inter Asterisk Exchange private frame type */
00109    AST_FRAME_IAX,
00110    /*! Text messages */
00111    AST_FRAME_TEXT,
00112    /*! Image Frames */
00113    AST_FRAME_IMAGE,
00114    /*! HTML Frame */
00115    AST_FRAME_HTML,
00116    /*! Comfort Noise frame (subclass is level of CNG in -dBov), 
00117        body may include zero or more 8-bit quantization coefficients */
00118    AST_FRAME_CNG,
00119    /*! Modem-over-IP data streams */
00120    AST_FRAME_MODEM,  
00121    /*! DTMF begin event, subclass is the digit */
00122    AST_FRAME_DTMF_BEGIN,
00123 };


Function Documentation

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

Definition at line 199 of file frame.c.

References AST_FRAME_VOICE, ast_log(), AST_MIN_OFFSET, AST_SMOOTHER_FLAG_G729, ast_swapcopy_samples(), ast_frame::data, ast_frame::datalen, ast_smoother::flags, ast_smoother::format, ast_frame::frametype, ast_smoother::len, LOG_WARNING, ast_frame::offset, ast_smoother::opt, ast_smoother::opt_needs_swap, ast_frame::ptr, ast_frame::samples, ast_smoother::samplesperbyte, ast_smoother::size, smoother_frame_feed(), SMOOTHER_SIZE, and ast_frame::subclass.

00200 {
00201    if (f->frametype != AST_FRAME_VOICE) {
00202       ast_log(LOG_WARNING, "Huh?  Can't smooth a non-voice frame!\n");
00203       return -1;
00204    }
00205    if (!s->format) {
00206       s->format = f->subclass;
00207       s->samplesperbyte = (float)f->samples / (float)f->datalen;
00208    } else if (s->format != f->subclass) {
00209       ast_log(LOG_WARNING, "Smoother was working on %d format frames, now trying to feed %d?\n", s->format, f->subclass);
00210       return -1;
00211    }
00212    if (s->len + f->datalen > SMOOTHER_SIZE) {
00213       ast_log(LOG_WARNING, "Out of smoother space\n");
00214       return -1;
00215    }
00216    if (((f->datalen == s->size) ||
00217         ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) &&
00218        !s->opt &&
00219        !s->len &&
00220        (f->offset >= AST_MIN_OFFSET)) {
00221       /* Optimize by sending the frame we just got
00222          on the next read, thus eliminating the douple
00223          copy */
00224       if (swap)
00225          ast_swapcopy_samples(f->data.ptr, f->data.ptr, f->samples);
00226       s->opt = f;
00227       s->opt_needs_swap = swap ? 1 : 0;
00228       return 0;
00229    }
00230 
00231    return smoother_frame_feed(s, f, swap);
00232 }

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 648 of file frame.c.

References ARRAY_LEN, and ast_format_list::desc.

Referenced by moh_alloc(), show_codec_n(), and show_codecs().

00649 {
00650    int x;
00651    char *ret = "unknown";
00652    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00653       if (AST_FORMAT_LIST[x].bits == codec) {
00654          ret = AST_FORMAT_LIST[x].desc;
00655          break;
00656       }
00657    }
00658    return ret;
00659 }

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 1205 of file frame.c.

References ARRAY_LEN, ast_best_codec(), ast_debug, AST_FORMAT_AUDIO_MASK, ast_format_list::bits, and ast_codec_pref::order.

Referenced by __oh323_new(), gtalk_new(), jingle_new(), process_sdp(), sip_new(), and socket_process().

01206 {
01207    int x, ret = 0, slot;
01208 
01209    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01210       slot = pref->order[x];
01211 
01212       if (!slot)
01213          break;
01214       if (formats & AST_FORMAT_LIST[slot-1].bits) {
01215          ret = AST_FORMAT_LIST[slot-1].bits;
01216          break;
01217       }
01218    }
01219    if (ret & AST_FORMAT_AUDIO_MASK)
01220       return ret;
01221 
01222    ast_debug(4, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec");
01223 
01224       return find_best ? ast_best_codec(formats) : 0;
01225 }

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 1469 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_SLINEAR, AST_FORMAT_SLINEAR16, AST_FORMAT_ULAW, ast_getformatname(), ast_log(), len(), and LOG_WARNING.

Referenced by moh_generate(), and monmp3thread().

01470 {
01471    int len = 0;
01472 
01473    /* XXX Still need speex, g723, and lpc10 XXX */ 
01474    switch(format) {
01475    case AST_FORMAT_G723_1:
01476       len = (samples / 240) * 20;
01477       break;
01478    case AST_FORMAT_ILBC:
01479       len = (samples / 240) * 50;
01480       break;
01481    case AST_FORMAT_GSM:
01482       len = (samples / 160) * 33;
01483       break;
01484    case AST_FORMAT_G729A:
01485       len = samples / 8;
01486       break;
01487    case AST_FORMAT_SLINEAR:
01488    case AST_FORMAT_SLINEAR16:
01489       len = samples * 2;
01490       break;
01491    case AST_FORMAT_ULAW:
01492    case AST_FORMAT_ALAW:
01493       len = samples;
01494       break;
01495    case AST_FORMAT_G722:
01496    case AST_FORMAT_ADPCM:
01497    case AST_FORMAT_G726:
01498    case AST_FORMAT_G726_AAL2:
01499       len = samples / 2;
01500       break;
01501    default:
01502       ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format));
01503    }
01504 
01505    return len;
01506 }

int ast_codec_get_samples ( struct ast_frame f  ) 

Returns the number of samples contained in the frame.

Definition at line 1425 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_SLINEAR16, AST_FORMAT_SPEEX, AST_FORMAT_ULAW, ast_getformatname(), ast_log(), ast_frame::data, ast_frame::datalen, g723_samples(), LOG_WARNING, ast_frame::ptr, speex_samples(), and ast_frame::subclass.

Referenced by ast_rtp_read(), isAnsweringMachine(), moh_generate(), schedule_delivery(), socket_process(), and socket_process_meta().

01426 {
01427    int samples=0;
01428    switch(f->subclass) {
01429    case AST_FORMAT_SPEEX:
01430       samples = speex_samples(f->data.ptr, f->datalen);
01431       break;
01432    case AST_FORMAT_G723_1:
01433       samples = g723_samples(f->data.ptr, f->datalen);
01434       break;
01435    case AST_FORMAT_ILBC:
01436       samples = 240 * (f->datalen / 50);
01437       break;
01438    case AST_FORMAT_GSM:
01439       samples = 160 * (f->datalen / 33);
01440       break;
01441    case AST_FORMAT_G729A:
01442       samples = f->datalen * 8;
01443       break;
01444    case AST_FORMAT_SLINEAR:
01445    case AST_FORMAT_SLINEAR16:
01446       samples = f->datalen / 2;
01447       break;
01448    case AST_FORMAT_LPC10:
01449       /* assumes that the RTP packet contains one LPC10 frame */
01450       samples = 22 * 8;
01451       samples += (((char *)(f->data.ptr))[7] & 0x1) * 8;
01452       break;
01453    case AST_FORMAT_ULAW:
01454    case AST_FORMAT_ALAW:
01455       samples = f->datalen;
01456       break;
01457    case AST_FORMAT_G722:
01458    case AST_FORMAT_ADPCM:
01459    case AST_FORMAT_G726:
01460    case AST_FORMAT_G726_AAL2:
01461       samples = f->datalen * 2;
01462       break;
01463    default:
01464       ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass));
01465    }
01466    return samples;
01467 }

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

Gets duration in ms of interpolation frame for a format.

Definition at line 655 of file frame.h.

References AST_FORMAT_ILBC.

Referenced by __get_from_jb(), and jb_get_and_deliver().

00656 { 
00657    return (format == AST_FORMAT_ILBC) ? 30 : 20;
00658 }

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 1065 of file frame.c.

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

Referenced by ast_parse_allow_disallow().

01066 {
01067    int x, newindex = 0;
01068 
01069    ast_codec_pref_remove(pref, format);
01070 
01071    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01072       if (AST_FORMAT_LIST[x].bits == format) {
01073          newindex = x + 1;
01074          break;
01075       }
01076    }
01077 
01078    if (newindex) {
01079       for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01080          if (!pref->order[x]) {
01081             pref->order[x] = newindex;
01082             break;
01083          }
01084       }
01085    }
01086 
01087    return x;
01088 }

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 968 of file frame.c.

References ast_codec_pref::order.

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

00969 {
00970    int x, differential = (int) 'A', mem;
00971    char *from, *to;
00972 
00973    if (right) {
00974       from = pref->order;
00975       to = buf;
00976       mem = size;
00977    } else {
00978       to = pref->order;
00979       from = buf;
00980       mem = 32;
00981    }
00982 
00983    memset(to, 0, mem);
00984    for (x = 0; x < 32 ; x++) {
00985       if (!from[x])
00986          break;
00987       to[x] = right ? (from[x] + differential) : (from[x] - differential);
00988    }
00989 }

struct ast_format_list ast_codec_pref_getsize ( struct ast_codec_pref pref,
int  format 
) [read]

Get packet size for codec.

Definition at line 1166 of file frame.c.

References ARRAY_LEN, ast_format_list::bits, ast_format_list::cur_ms, ast_format_list::def_ms, format, ast_format_list::inc_ms, ast_format_list::max_ms, and ast_format_list::min_ms.

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

01167 {
01168    int x, idx = -1, framems = 0;
01169    struct ast_format_list fmt = { 0, };
01170 
01171    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01172       if (AST_FORMAT_LIST[x].bits == format) {
01173          fmt = AST_FORMAT_LIST[x];
01174          idx = x;
01175          break;
01176       }
01177    }
01178 
01179    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01180       if (pref->order[x] == (idx + 1)) {
01181          framems = pref->framing[x];
01182          break;
01183       }
01184    }
01185 
01186    /* size validation */
01187    if (!framems)
01188       framems = AST_FORMAT_LIST[idx].def_ms;
01189 
01190    if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */
01191       framems -= framems % AST_FORMAT_LIST[idx].inc_ms;
01192 
01193    if (framems < AST_FORMAT_LIST[idx].min_ms)
01194       framems = AST_FORMAT_LIST[idx].min_ms;
01195 
01196    if (framems > AST_FORMAT_LIST[idx].max_ms)
01197       framems = AST_FORMAT_LIST[idx].max_ms;
01198 
01199    fmt.cur_ms = framems;
01200 
01201    return fmt;
01202 }

int ast_codec_pref_index ( struct ast_codec_pref pref,
int  index 
)

Codec located at a particular place in the preference index.

Definition at line 1026 of file frame.c.

References 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(), handle_cli_iax2_show_peer(), jingle_accept_call(), print_codec_to_cli(), and socket_process().

01027 {
01028    int slot = 0;
01029 
01030    if ((idx >= 0) && (idx < sizeof(pref->order))) {
01031       slot = pref->order[idx];
01032    }
01033 
01034    return slot ? AST_FORMAT_LIST[slot - 1].bits : 0;
01035 }

void ast_codec_pref_init ( struct ast_codec_pref pref  ) 

Initialize an audio codec preference to "no preference".

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 1091 of file frame.c.

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

Referenced by create_addr().

01092 {
01093    int x, newindex = 0;
01094 
01095    /* First step is to get the codecs "index number" */
01096    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01097       if (AST_FORMAT_LIST[x].bits == format) {
01098          newindex = x + 1;
01099          break;
01100       }
01101    }
01102    /* Done if its unknown */
01103    if (!newindex)
01104       return;
01105 
01106    /* Now find any existing occurrence, or the end */
01107    for (x = 0; x < 32; x++) {
01108       if (!pref->order[x] || pref->order[x] == newindex)
01109          break;
01110    }
01111 
01112    if (only_if_existing && !pref->order[x])
01113       return;
01114 
01115    /* Move down to make space to insert - either all the way to the end,
01116       or as far as the existing location (which will be overwritten) */
01117    for (; x > 0; x--) {
01118       pref->order[x] = pref->order[x - 1];
01119       pref->framing[x] = pref->framing[x - 1];
01120    }
01121 
01122    /* And insert the new entry */
01123    pref->order[0] = newindex;
01124    pref->framing[0] = 0; /* ? */
01125 }

void ast_codec_pref_remove ( struct ast_codec_pref pref,
int  format 
)

Remove audio a codec from a preference list.

Definition at line 1038 of file frame.c.

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

Referenced by ast_codec_pref_append(), and ast_parse_allow_disallow().

01039 {
01040    struct ast_codec_pref oldorder;
01041    int x, y = 0;
01042    int slot;
01043    int size;
01044 
01045    if (!pref->order[0])
01046       return;
01047 
01048    memcpy(&oldorder, pref, sizeof(oldorder));
01049    memset(pref, 0, sizeof(*pref));
01050 
01051    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01052       slot = oldorder.order[x];
01053       size = oldorder.framing[x];
01054       if (! slot)
01055          break;
01056       if (AST_FORMAT_LIST[slot-1].bits != format) {
01057          pref->order[y] = slot;
01058          pref->framing[y++] = size;
01059       }
01060    }
01061    
01062 }

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

Set packet size for codec.

Definition at line 1128 of file frame.c.

References ARRAY_LEN, ast_format_list::def_ms, ast_codec_pref::framing, ast_format_list::inc_ms, ast_format_list::max_ms, ast_format_list::min_ms, and ast_codec_pref::order.

Referenced by ast_parse_allow_disallow(), and process_sdp().

01129 {
01130    int x, idx = -1;
01131 
01132    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01133       if (AST_FORMAT_LIST[x].bits == format) {
01134          idx = x;
01135          break;
01136       }
01137    }
01138 
01139    if (idx < 0)
01140       return -1;
01141 
01142    /* size validation */
01143    if (!framems)
01144       framems = AST_FORMAT_LIST[idx].def_ms;
01145 
01146    if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */
01147       framems -= framems % AST_FORMAT_LIST[idx].inc_ms;
01148 
01149    if (framems < AST_FORMAT_LIST[idx].min_ms)
01150       framems = AST_FORMAT_LIST[idx].min_ms;
01151 
01152    if (framems > AST_FORMAT_LIST[idx].max_ms)
01153       framems = AST_FORMAT_LIST[idx].max_ms;
01154 
01155    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01156       if (pref->order[x] == (idx + 1)) {
01157          pref->framing[x] = framems;
01158          break;
01159       }
01160    }
01161 
01162    return x;
01163 }

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 991 of file frame.c.

References ast_codec_pref_index(), and ast_getformatname().

Referenced by dump_prefs(), and socket_process().

00992 {
00993    int x, codec; 
00994    size_t total_len, slen;
00995    char *formatname;
00996    
00997    memset(buf,0,size);
00998    total_len = size;
00999    buf[0] = '(';
01000    total_len--;
01001    for(x = 0; x < 32 ; x++) {
01002       if (total_len <= 0)
01003          break;
01004       if (!(codec = ast_codec_pref_index(pref,x)))
01005          break;
01006       if ((formatname = ast_getformatname(codec))) {
01007          slen = strlen(formatname);
01008          if (slen > total_len)
01009             break;
01010          strncat(buf, formatname, total_len - 1); /* safe */
01011          total_len -= slen;
01012       }
01013       if (total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) {
01014          strncat(buf, "|", total_len - 1); /* safe */
01015          total_len--;
01016       }
01017    }
01018    if (total_len) {
01019       strncat(buf, ")", total_len - 1); /* safe */
01020       total_len--;
01021    }
01022 
01023    return size - total_len;
01024 }

static force_inline int ast_format_rate ( int  format  )  [static]

Get the sample rate for a given format.

Definition at line 682 of file frame.h.

References AST_FORMAT_G722, and AST_FORMAT_SLINEAR16.

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

00683 {
00684    if (format == AST_FORMAT_G722 || format == AST_FORMAT_SLINEAR16)
00685       return 16000;
00686 
00687    return 8000;
00688 }

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 1508 of file frame.c.

References AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_slinear_saturated_divide(), ast_slinear_saturated_multiply(), ast_frame::data, ast_frame::frametype, ast_frame::ptr, ast_frame::samples, and ast_frame::subclass.

Referenced by audiohook_read_frame_single(), audiohook_volume_callback(), conf_run(), and volume_callback().

01509 {
01510    int count;
01511    short *fdata = f->data.ptr;
01512    short adjust_value = abs(adjustment);
01513 
01514    if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR))
01515       return -1;
01516 
01517    if (!adjustment)
01518       return 0;
01519 
01520    for (count = 0; count < f->samples; count++) {
01521       if (adjustment > 0) {
01522          ast_slinear_saturated_multiply(&fdata[count], &adjust_value);
01523       } else if (adjustment < 0) {
01524          ast_slinear_saturated_divide(&fdata[count], &adjust_value);
01525       }
01526    }
01527 
01528    return 0;
01529 }

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

Dump a frame for debugging purposes

Definition at line 750 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_RADIO_KEY, AST_CONTROL_RADIO_UNKEY, AST_CONTROL_RING, AST_CONTROL_RINGING, AST_CONTROL_T38_PARAMETERS, 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_T38_NEGOTIATED, AST_T38_REFUSED, AST_T38_REQUEST_NEGOTIATE, AST_T38_REQUEST_TERMINATE, AST_T38_TERMINATED, ast_verbose, COLOR_BLACK, COLOR_BRCYAN, COLOR_BRGREEN, COLOR_BRMAGENTA, COLOR_BRRED, COLOR_YELLOW, ast_frame::data, ast_frame::datalen, ast_frame::frametype, ast_frame::ptr, ast_control_t38_parameters::request_response, ast_frame::subclass, and term_color().

Referenced by __ast_read(), and ast_write().

00751 {
00752    const char noname[] = "unknown";
00753    char ftype[40] = "Unknown Frametype";
00754    char cft[80];
00755    char subclass[40] = "Unknown Subclass";
00756    char csub[80];
00757    char moreinfo[40] = "";
00758    char cn[60];
00759    char cp[40];
00760    char cmn[40];
00761    const char *message = "Unknown";
00762 
00763    if (!name)
00764       name = noname;
00765 
00766 
00767    if (!f) {
00768       ast_verbose("%s [ %s (NULL) ] [%s]\n", 
00769          term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00770          term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 
00771          term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00772       return;
00773    }
00774    /* XXX We should probably print one each of voice and video when the format changes XXX */
00775    if (f->frametype == AST_FRAME_VOICE)
00776       return;
00777    if (f->frametype == AST_FRAME_VIDEO)
00778       return;
00779    switch(f->frametype) {
00780    case AST_FRAME_DTMF_BEGIN:
00781       strcpy(ftype, "DTMF Begin");
00782       subclass[0] = f->subclass;
00783       subclass[1] = '\0';
00784       break;
00785    case AST_FRAME_DTMF_END:
00786       strcpy(ftype, "DTMF End");
00787       subclass[0] = f->subclass;
00788       subclass[1] = '\0';
00789       break;
00790    case AST_FRAME_CONTROL:
00791       strcpy(ftype, "Control");
00792       switch(f->subclass) {
00793       case AST_CONTROL_HANGUP:
00794          strcpy(subclass, "Hangup");
00795          break;
00796       case AST_CONTROL_RING:
00797          strcpy(subclass, "Ring");
00798          break;
00799       case AST_CONTROL_RINGING:
00800          strcpy(subclass, "Ringing");
00801          break;
00802       case AST_CONTROL_ANSWER:
00803          strcpy(subclass, "Answer");
00804          break;
00805       case AST_CONTROL_BUSY:
00806          strcpy(subclass, "Busy");
00807          break;
00808       case AST_CONTROL_TAKEOFFHOOK:
00809          strcpy(subclass, "Take Off Hook");
00810          break;
00811       case AST_CONTROL_OFFHOOK:
00812          strcpy(subclass, "Line Off Hook");
00813          break;
00814       case AST_CONTROL_CONGESTION:
00815          strcpy(subclass, "Congestion");
00816          break;
00817       case AST_CONTROL_FLASH:
00818          strcpy(subclass, "Flash");
00819          break;
00820       case AST_CONTROL_WINK:
00821          strcpy(subclass, "Wink");
00822          break;
00823       case AST_CONTROL_OPTION:
00824          strcpy(subclass, "Option");
00825          break;
00826       case AST_CONTROL_RADIO_KEY:
00827          strcpy(subclass, "Key Radio");
00828          break;
00829       case AST_CONTROL_RADIO_UNKEY:
00830          strcpy(subclass, "Unkey Radio");
00831          break;
00832       case AST_CONTROL_HOLD:
00833          strcpy(subclass, "Hold");
00834          break;
00835       case AST_CONTROL_UNHOLD:
00836          strcpy(subclass, "Unhold");
00837          break;
00838       case AST_CONTROL_T38_PARAMETERS:
00839          if (f->datalen != sizeof(struct ast_control_t38_parameters *)) {
00840             message = "Invalid";
00841          } else {
00842             struct ast_control_t38_parameters *parameters = f->data.ptr;
00843             enum ast_control_t38 state = parameters->request_response;
00844             if (state == AST_T38_REQUEST_NEGOTIATE)
00845                message = "Negotiation Requested";
00846             else if (state == AST_T38_REQUEST_TERMINATE)
00847                message = "Negotiation Request Terminated";
00848             else if (state == AST_T38_NEGOTIATED)
00849                message = "Negotiated";
00850             else if (state == AST_T38_TERMINATED)
00851                message = "Terminated";
00852             else if (state == AST_T38_REFUSED)
00853                message = "Refused";
00854          }
00855          snprintf(subclass, sizeof(subclass), "T38_Parameters/%s", message);
00856          break;
00857       case -1:
00858          strcpy(subclass, "Stop generators");
00859          break;
00860       default:
00861          snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass);
00862       }
00863       break;
00864    case AST_FRAME_NULL:
00865       strcpy(ftype, "Null Frame");
00866       strcpy(subclass, "N/A");
00867       break;
00868    case AST_FRAME_IAX:
00869       /* Should never happen */
00870       strcpy(ftype, "IAX Specific");
00871       snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass);
00872       break;
00873    case AST_FRAME_TEXT:
00874       strcpy(ftype, "Text");
00875       strcpy(subclass, "N/A");
00876       ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo));
00877       break;
00878    case AST_FRAME_IMAGE:
00879       strcpy(ftype, "Image");
00880       snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass));
00881       break;
00882    case AST_FRAME_HTML:
00883       strcpy(ftype, "HTML");
00884       switch(f->subclass) {
00885       case AST_HTML_URL:
00886          strcpy(subclass, "URL");
00887          ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo));
00888          break;
00889       case AST_HTML_DATA:
00890          strcpy(subclass, "Data");
00891          break;
00892       case AST_HTML_BEGIN:
00893          strcpy(subclass, "Begin");
00894          break;
00895       case AST_HTML_END:
00896          strcpy(subclass, "End");
00897          break;
00898       case AST_HTML_LDCOMPLETE:
00899          strcpy(subclass, "Load Complete");
00900          break;
00901       case AST_HTML_NOSUPPORT:
00902          strcpy(subclass, "No Support");
00903          break;
00904       case AST_HTML_LINKURL:
00905          strcpy(subclass, "Link URL");
00906          ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo));
00907          break;
00908       case AST_HTML_UNLINK:
00909          strcpy(subclass, "Unlink");
00910          break;
00911       case AST_HTML_LINKREJECT:
00912          strcpy(subclass, "Link Reject");
00913          break;
00914       default:
00915          snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass);
00916          break;
00917       }
00918       break;
00919    case AST_FRAME_MODEM:
00920       strcpy(ftype, "Modem");
00921       switch (f->subclass) {
00922       case AST_MODEM_T38:
00923          strcpy(subclass, "T.38");
00924          break;
00925       case AST_MODEM_V150:
00926          strcpy(subclass, "V.150");
00927          break;
00928       default:
00929          snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass);
00930          break;
00931       }
00932       break;
00933    default:
00934       snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype);
00935    }
00936    if (!ast_strlen_zero(moreinfo))
00937       ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n",  
00938              term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00939              term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)),
00940              f->frametype, 
00941              term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)),
00942              f->subclass, 
00943              term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)),
00944              term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00945    else
00946       ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n",  
00947              term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00948              term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)),
00949              f->frametype, 
00950              term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)),
00951              f->subclass, 
00952              term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00953 }

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

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.

Parameters:
source Request a frame be allocated. source is an optional source of the frame, len is the requested length, or "0" if the caller will supply the buffer

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 373 of file frame.c.

References __frame_free(), and AST_LIST_NEXT.

Referenced by mixmonitor_thread().

00374 {
00375    struct ast_frame *next;
00376 
00377    for (next = AST_LIST_NEXT(frame, frame_list);
00378         frame;
00379         frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) {
00380       __frame_free(frame, cache);
00381    }
00382 }

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 1531 of file frame.c.

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

01532 {
01533    int count;
01534    short *data1, *data2;
01535 
01536    if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR))
01537       return -1;
01538 
01539    if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR))
01540       return -1;
01541 
01542    if (f1->samples != f2->samples)
01543       return -1;
01544 
01545    for (count = 0, data1 = f1->data.ptr, data2 = f2->data.ptr;
01546         count < f1->samples;
01547         count++, data1++, data2++)
01548       ast_slinear_saturated_add(data1, data2);
01549 
01550    return 0;
01551 }

struct ast_frame* ast_frdup ( const struct ast_frame fr  )  [read]

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 470 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(), buf, ast_frame::data, ast_frame::datalen, ast_frame::delivery, frame_cache, frames, ast_frame::frametype, ast_frame::len, len(), ast_frame_cache::list, ast_frame::mallocd, ast_frame::mallocd_hdr_len, ast_frame::offset, ast_frame::ptr, ast_frame::samples, ast_frame::seqno, ast_frame_cache::size, ast_frame::src, ast_frame::subclass, ast_frame::ts, and ast_frame::uint32.

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

00471 {
00472    struct ast_frame *out = NULL;
00473    int len, srclen = 0;
00474    void *buf = NULL;
00475 
00476 #if !defined(LOW_MEMORY)
00477    struct ast_frame_cache *frames;
00478 #endif
00479 
00480    /* Start with standard stuff */
00481    len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen;
00482    /* If we have a source, add space for it */
00483    /*
00484     * XXX Watch out here - if we receive a src which is not terminated
00485     * properly, we can be easily attacked. Should limit the size we deal with.
00486     */
00487    if (f->src)
00488       srclen = strlen(f->src);
00489    if (srclen > 0)
00490       len += srclen + 1;
00491    
00492 #if !defined(LOW_MEMORY)
00493    if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) {
00494       AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) {
00495          if (out->mallocd_hdr_len >= len) {
00496             size_t mallocd_len = out->mallocd_hdr_len;
00497 
00498             AST_LIST_REMOVE_CURRENT(frame_list);
00499             memset(out, 0, sizeof(*out));
00500             out->mallocd_hdr_len = mallocd_len;
00501             buf = out;
00502             frames->size--;
00503             break;
00504          }
00505       }
00506       AST_LIST_TRAVERSE_SAFE_END;
00507    }
00508 #endif
00509 
00510    if (!buf) {
00511       if (!(buf = ast_calloc_cache(1, len)))
00512          return NULL;
00513       out = buf;
00514       out->mallocd_hdr_len = len;
00515    }
00516 
00517    out->frametype = f->frametype;
00518    out->subclass = f->subclass;
00519    out->datalen = f->datalen;
00520    out->samples = f->samples;
00521    out->delivery = f->delivery;
00522    /* Set us as having malloc'd header only, so it will eventually
00523       get freed. */
00524    out->mallocd = AST_MALLOCD_HDR;
00525    out->offset = AST_FRIENDLY_OFFSET;
00526    if (out->datalen) {
00527       out->data.ptr = buf + sizeof(*out) + AST_FRIENDLY_OFFSET;
00528       memcpy(out->data.ptr, f->data.ptr, out->datalen);  
00529    } else {
00530       out->data.uint32 = f->data.uint32;
00531    }
00532    if (srclen > 0) {
00533       /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */
00534       char *src;
00535       out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen;
00536       src = (char *) out->src;
00537       /* Must have space since we allocated for it */
00538       strcpy(src, f->src);
00539    }
00540    ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO);
00541    out->ts = f->ts;
00542    out->len = f->len;
00543    out->seqno = f->seqno;
00544    return out;
00545 }

struct ast_frame* ast_frisolate ( struct ast_frame fr  )  [read]

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 389 of file frame.c.

References ast_clear_flag, ast_copy_flags, ast_frame_header_new(), ast_frdup(), ast_free, 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, ast_frame::len, ast_frame::mallocd, ast_frame::offset, ast_frame::ptr, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, and ast_frame::ts.

Referenced by __ast_answer(), ast_slinfactory_feed(), autoservice_run(), and jpeg_read_image().

00390 {
00391    struct ast_frame *out;
00392    void *newdata;
00393 
00394    /* if none of the existing frame is malloc'd, let ast_frdup() do it
00395       since it is more efficient
00396    */
00397    if (fr->mallocd == 0) {
00398       return ast_frdup(fr);
00399    }
00400 
00401    /* if everything is already malloc'd, we are done */
00402    if ((fr->mallocd & (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) ==
00403        (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) {
00404       return fr;
00405    }
00406 
00407    if (!(fr->mallocd & AST_MALLOCD_HDR)) {
00408       /* Allocate a new header if needed */
00409       if (!(out = ast_frame_header_new())) {
00410          return NULL;
00411       }
00412       out->frametype = fr->frametype;
00413       out->subclass = fr->subclass;
00414       out->datalen = fr->datalen;
00415       out->samples = fr->samples;
00416       out->offset = fr->offset;
00417       /* Copy the timing data */
00418       ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO);
00419       if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) {
00420          out->ts = fr->ts;
00421          out->len = fr->len;
00422          out->seqno = fr->seqno;
00423       }
00424    } else {
00425       ast_clear_flag(fr, AST_FRFLAG_FROM_TRANSLATOR);
00426       ast_clear_flag(fr, AST_FRFLAG_FROM_DSP);
00427       ast_clear_flag(fr, AST_FRFLAG_FROM_FILESTREAM);
00428       out = fr;
00429    }
00430    
00431    if (!(fr->mallocd & AST_MALLOCD_SRC) && fr->src) {
00432       if (!(out->src = ast_strdup(fr->src))) {
00433          if (out != fr) {
00434             ast_free(out);
00435          }
00436          return NULL;
00437       }
00438    } else {
00439       out->src = fr->src;
00440       fr->src = NULL;
00441       fr->mallocd &= ~AST_MALLOCD_SRC;
00442    }
00443    
00444    if (!(fr->mallocd & AST_MALLOCD_DATA))  {
00445       if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) {
00446          if (out->src != fr->src) {
00447             ast_free((void *) out->src);
00448          }
00449          if (out != fr) {
00450             ast_free(out);
00451          }
00452          return NULL;
00453       }
00454       newdata += AST_FRIENDLY_OFFSET;
00455       out->offset = AST_FRIENDLY_OFFSET;
00456       out->datalen = fr->datalen;
00457       memcpy(newdata, fr->data.ptr, fr->datalen);
00458       out->data.ptr = newdata;
00459    } else {
00460       out->data = fr->data;
00461       memset(&fr->data, 0, sizeof(fr->data));
00462       fr->mallocd &= ~AST_MALLOCD_DATA;
00463    }
00464 
00465    out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA;
00466    
00467    return out;
00468 }

struct ast_format_list* ast_get_format_list ( size_t *  size  )  [read]

Definition at line 563 of file frame.c.

References ARRAY_LEN.

00564 {
00565    *size = ARRAY_LEN(AST_FORMAT_LIST);
00566    return AST_FORMAT_LIST;
00567 }

struct ast_format_list* ast_get_format_list_index ( int  index  )  [read]

Definition at line 558 of file frame.c.

00559 {
00560    return &AST_FORMAT_LIST[idx];
00561 }

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 630 of file frame.c.

References ARRAY_LEN, ast_expand_codec_alias(), ast_format_list::bits, and format.

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

00631 {
00632    int x, all, format = 0;
00633 
00634    all = strcasecmp(name, "all") ? 0 : 1;
00635    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00636       if (all || 
00637            !strcasecmp(AST_FORMAT_LIST[x].name,name) ||
00638            !strcasecmp(AST_FORMAT_LIST[x].name, ast_expand_codec_alias(name))) {
00639          format |= AST_FORMAT_LIST[x].bits;
00640          if (!all)
00641             break;
00642       }
00643    }
00644 
00645    return format;
00646 }

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 569 of file frame.c.

References ARRAY_LEN, ast_format_list::bits, and ast_format_list::name.

Referenced by __ast_play_and_record(), __ast_read(), __ast_register_translator(), _sip_show_peer(), add_codec_to_answer(), add_codec_to_sdp(), add_tcodec_to_sdp(), add_vcodec_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(), handle_cli_core_show_file_formats(), handle_cli_core_show_translation(), handle_cli_iax2_show_channels(), handle_cli_iax2_show_peer(), handle_cli_moh_show_classes(), handle_core_show_image_formats(), iax2_request(), iax_show_provisioning(), jingle_show_channels(), login_exec(), 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(), sip_request_call(), sip_rtp_read(), socket_process(), start_rtp(), unistim_request(), unistim_rtp_read(), and unistim_write().

00570 {
00571    int x;
00572    char *ret = "unknown";
00573    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00574       if (AST_FORMAT_LIST[x].bits == format) {
00575          ret = AST_FORMAT_LIST[x].name;
00576          break;
00577       }
00578    }
00579    return ret;
00580 }

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 582 of file frame.c.

References ARRAY_LEN, ast_copy_string(), ast_format_list::bits, len(), and name.

Referenced by __ast_read(), _sip_show_peer(), add_sdp(), ast_streamfile(), function_iaxpeer(), function_sippeer(), gtalk_is_answered(), gtalk_newcall(), handle_cli_iax2_show_peer(), handle_showchan(), handle_skinny_show_line(), iax2_bridge(), process_sdp(), serialize_showchan(), set_format(), show_channels_cb(), sip_new(), sip_request_call(), sip_show_channel(), sip_show_settings(), and sip_write().

00583 {
00584    int x;
00585    unsigned len;
00586    char *start, *end = buf;
00587 
00588    if (!size)
00589       return buf;
00590    snprintf(end, size, "0x%x (", format);
00591    len = strlen(end);
00592    end += len;
00593    size -= len;
00594    start = end;
00595    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00596       if (AST_FORMAT_LIST[x].bits & format) {
00597          snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name);
00598          len = strlen(end);
00599          end += len;
00600          size -= len;
00601       }
00602    }
00603    if (start == end)
00604       ast_copy_string(start, "nothing)", size);
00605    else if (size > 1)
00606       *(end -1) = ')';
00607    return buf;
00608 }

int 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.

Returns:
Returns number of errors encountered during parsing

Definition at line 1227 of file frame.c.

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

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

01228 {
01229    int errors = 0;
01230    char *parse = NULL, *this = NULL, *psize = NULL;
01231    int format = 0, framems = 0;
01232 
01233    parse = ast_strdupa(list);
01234    while ((this = strsep(&parse, ","))) {
01235       framems = 0;
01236       if ((psize = strrchr(this, ':'))) {
01237          *psize++ = '\0';
01238          ast_debug(1, "Packetization for codec: %s is %s\n", this, psize);
01239          framems = atoi(psize);
01240          if (framems < 0) {
01241             framems = 0;
01242             errors++;
01243             ast_log(LOG_WARNING, "Bad packetization value for codec %s\n", this);
01244          }
01245       }
01246       if (!(format = ast_getformatbyname(this))) {
01247          ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this);
01248          errors++;
01249          continue;
01250       }
01251 
01252       if (mask) {
01253          if (allowing)
01254             *mask |= format;
01255          else
01256             *mask &= ~format;
01257       }
01258 
01259       /* Set up a preference list for audio. Do not include video in preferences 
01260          since we can not transcode video and have to use whatever is offered
01261        */
01262       if (pref && (format & AST_FORMAT_AUDIO_MASK)) {
01263          if (strcasecmp(this, "all")) {
01264             if (allowing) {
01265                ast_codec_pref_append(pref, format);
01266                ast_codec_pref_setsize(pref, format, framems);
01267             }
01268             else
01269                ast_codec_pref_remove(pref, format);
01270          } else if (!allowing) {
01271             memset(pref, 0, sizeof(*pref));
01272          }
01273       }
01274    }
01275    return errors;
01276 }

void ast_smoother_free ( struct ast_smoother s  ) 

Definition at line 284 of file frame.c.

References ast_free.

Referenced by ast_rtp_destroy(), and ast_rtp_write().

00285 {
00286    ast_free(s);
00287 }

int ast_smoother_get_flags ( struct ast_smoother smoother  ) 

Definition at line 184 of file frame.c.

References ast_smoother::flags.

00185 {
00186    return s->flags;
00187 }

struct ast_smoother* ast_smoother_new ( int  bytes  )  [read]

Definition at line 174 of file frame.c.

References ast_malloc, ast_smoother_reset(), and s.

Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().

00175 {
00176    struct ast_smoother *s;
00177    if (size < 1)
00178       return NULL;
00179    if ((s = ast_malloc(sizeof(*s))))
00180       ast_smoother_reset(s, size);
00181    return s;
00182 }

struct ast_frame* ast_smoother_read ( struct ast_smoother s  )  [read]

Definition at line 234 of file frame.c.

References ast_format_rate(), AST_FRAME_VOICE, AST_FRIENDLY_OFFSET, ast_log(), ast_samp2tv(), AST_SMOOTHER_FLAG_G729, ast_tvadd(), ast_tvzero(), ast_smoother::data, ast_frame::data, ast_frame::datalen, ast_smoother::delivery, ast_frame::delivery, ast_smoother::f, ast_smoother::flags, ast_smoother::format, ast_smoother::framedata, ast_frame::frametype, ast_smoother::len, len(), LOG_WARNING, ast_frame::offset, ast_smoother::opt, ast_frame::ptr, ast_frame::samples, ast_smoother::samplesperbyte, ast_smoother::size, and ast_frame::subclass.

Referenced by ast_rtp_write().

00235 {
00236    struct ast_frame *opt;
00237    int len;
00238 
00239    /* IF we have an optimization frame, send it */
00240    if (s->opt) {
00241       if (s->opt->offset < AST_FRIENDLY_OFFSET)
00242          ast_log(LOG_WARNING, "Returning a frame of inappropriate offset (%d).\n",
00243                      s->opt->offset);
00244       opt = s->opt;
00245       s->opt = NULL;
00246       return opt;
00247    }
00248 
00249    /* Make sure we have enough data */
00250    if (s->len < s->size) {
00251       /* Or, if this is a G.729 frame with VAD on it, send it immediately anyway */
00252       if (!((s->flags & AST_SMOOTHER_FLAG_G729) && (s->len % 10)))
00253          return NULL;
00254    }
00255    len = s->size;
00256    if (len > s->len)
00257       len = s->len;
00258    /* Make frame */
00259    s->f.frametype = AST_FRAME_VOICE;
00260    s->f.subclass = s->format;
00261    s->f.data.ptr = s->framedata + AST_FRIENDLY_OFFSET;
00262    s->f.offset = AST_FRIENDLY_OFFSET;
00263    s->f.datalen = len;
00264    /* Samples will be improper given VAD, but with VAD the concept really doesn't even exist */
00265    s->f.samples = len * s->samplesperbyte;   /* XXX rounding */
00266    s->f.delivery = s->delivery;
00267    /* Fill Data */
00268    memcpy(s->f.data.ptr, s->data, len);
00269    s->len -= len;
00270    /* Move remaining data to the front if applicable */
00271    if (s->len) {
00272       /* In principle this should all be fine because if we are sending
00273          G.729 VAD, the next timestamp will take over anyawy */
00274       memmove(s->data, s->data + len, s->len);
00275       if (!ast_tvzero(s->delivery)) {
00276          /* If we have delivery time, increment it, otherwise, leave it at 0 */
00277          s->delivery = ast_tvadd(s->delivery, ast_samp2tv(s->f.samples, ast_format_rate(s->format)));
00278       }
00279    }
00280    /* Return frame */
00281    return &s->f;
00282 }

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 152 of file frame.c.

References ast_smoother::opt, ast_smoother::opt_needs_swap, ast_smoother::size, and smoother_frame_feed().

Referenced by ast_rtp_codec_setpref().

00153 {
00154    /* if there is no change, then nothing to do */
00155    if (s->size == bytes) {
00156       return;
00157    }
00158    /* set the new desired output size */
00159    s->size = bytes;
00160    /* if there is no 'optimized' frame in the smoother,
00161     *   then there is nothing left to do
00162     */
00163    if (!s->opt) {
00164       return;
00165    }
00166    /* there is an 'optimized' frame here at the old size,
00167     * but it must now be put into the buffer so the data
00168     * can be extracted at the new size
00169     */
00170    smoother_frame_feed(s, s->opt, s->opt_needs_swap);
00171    s->opt = NULL;
00172 }

void ast_smoother_reset ( struct ast_smoother s,
int  bytes 
)

Definition at line 146 of file frame.c.

References ast_smoother::size.

Referenced by ast_smoother_new().

00147 {
00148    memset(s, 0, sizeof(*s));
00149    s->size = bytes;
00150 }

void ast_smoother_set_flags ( struct ast_smoother smoother,
int  flags 
)

Definition at line 189 of file frame.c.

References ast_smoother::flags.

Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().

00190 {
00191    s->flags = flags;
00192 }

int ast_smoother_test_flag ( struct ast_smoother s,
int  flag 
)

Definition at line 194 of file frame.c.

References ast_smoother::flags.

Referenced by ast_rtp_write().

00195 {
00196    return (s->flags & flag);
00197 }

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

Definition at line 547 of file frame.c.

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

00548 {
00549    int i;
00550    unsigned short *dst_s = dst;
00551    const unsigned short *src_s = src;
00552 
00553    for (i = 0; i < samples; i++)
00554       dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8);
00555 }


Variable Documentation


Generated on 3 Mar 2010 for Asterisk - the Open Source PBX by  doxygen 1.6.1