Wed Aug 18 22:34:23 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"

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

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)
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)

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) }
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, AST_CONTROL_SRCCHANGE = 25
}
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.
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)
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.
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

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 244 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 240 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(), oh323_rtp_read(), pcm_seek(), pcm_write(), and start_rtp().

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

Maximum audio mask

Definition at line 264 of file frame.h.

Referenced by add_sdp(), ast_best_codec(), ast_channel_make_compatible_helper(), ast_closestream(), ast_codec_choose(), ast_filehelper(), ast_openstream_full(), ast_openvstream(), ast_parse_allow_disallow(), ast_playstream(), ast_request(), ast_rtp_read(), ast_translate_available_formats(), ast_translator_best_choice(), ast_writestream(), begin_dial_channel(), filestream_destructor(), func_channel_read(), generator_force(), gtalk_rtp_read(), jingle_rtp_read(), oh323_request(), phone_read(), process_sdp(), set_format(), sip_call(), sip_request_call(), sip_rtp_read(), sip_write(), skinny_request(), transmit_connect_with_sdp(), and transmit_modify_with_sdp().

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

Unsupported audio bits

Definition at line 260 of file frame.h.

#define AST_FORMAT_G722   (1 << 12)

G.722

Definition at line 258 of file frame.h.

Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_raw_write(), ast_slinfactory_feed(), au_seek(), convertcap(), g722tolin16_sample(), g722tolin_sample(), pcm_read(), and rtp_get_rate().

#define AST_FORMAT_G723_1   (1 << 0)

G.723.1 compression

Definition at line 234 of file frame.h.

Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), 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(), register_translator(), and start_rtp().

#define AST_FORMAT_G726   (1 << 11)

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

Definition at line 256 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 242 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(), codec_skinny2ast(), and setup_rtp_connection().

#define AST_FORMAT_G729A   (1 << 8)

G.729A audio

Definition at line 250 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(), g729_write(), load_module(), phone_request(), phone_setup(), phone_write(), and start_rtp().

#define AST_FORMAT_GSM   (1 << 1)

GSM compression

Definition at line 236 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 270 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 272 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 274 of file frame.h.

Referenced by h263p_encap().

#define AST_FORMAT_H264   (1 << 21)

H.264 Video

Definition at line 276 of file frame.h.

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

#define AST_FORMAT_ILBC   (1 << 10)

iLBC Free Compression

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(), ast_codec_interp_len(), convertcap(), ilbc_read(), ilbc_write(), and ilbctolin_sample().

#define AST_FORMAT_JPEG   (1 << 16)

JPEG Images

Definition at line 266 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 248 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 285 of file frame.h.

#define AST_FORMAT_MP4_VIDEO   (1 << 22)

MPEG4 Video

Definition at line 278 of file frame.h.

Referenced by mpeg4_encap().

#define AST_FORMAT_PNG   (1 << 17)

PNG Images

Definition at line 268 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 246 of file frame.h.

Referenced by __ast_play_and_record(), __ast_register_translator(), _moh_class_malloc(), 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_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(), mixmonitor_thread(), 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(), 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(), transmit_audio(), usbradio_new(), usbradio_read(), usbradio_request(), wav_read(), and wav_write().

#define AST_FORMAT_SLINEAR16   (1 << 15)

Raw 16-bit Signed Linear (16000 Hz) PCM

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_slinfactory_feed(), console_new(), lin16tog722_sample(), slin16_to_slin8_sample(), slinear_read(), slinear_write(), and stream_monitor().

#define AST_FORMAT_SPEEX   (1 << 9)

SpeeX Free Compression

Definition at line 252 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 283 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 281 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 286 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)

Raw mu-law data (G.711)

Definition at line 238 of file frame.h.

Referenced by __adsi_transmit_messages(), _ast_adsi_transmit_message_full(), adsi_careful_send(), alarmreceiver_exec(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), calc_energy(), codec_ast2skinny(), codec_skinny2ast(), conf_run(), convertcap(), dahdi_new(), dahdi_read(), dahdi_translate(), dahdi_write(), 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(), start_rtp(), ulawtoalaw_sample(), and ulawtolin_sample().

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

Definition at line 279 of file frame.h.

Referenced by add_sdp(), ast_filehelper(), ast_openvstream(), ast_request(), ast_rtp_read(), ast_translate_available_formats(), check_peer_ok(), create_addr_from_peer(), func_channel_read(), gtalk_new(), gtalk_rtp_read(), jingle_new(), jingle_rtp_read(), sip_new(), and sip_rtp_read().

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

Definition at line 487 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 486 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 __adsi_transmit_messages(), __ast_play_and_record(), action_atxfer(), action_dahdidialoffhook(), agent_ack_sleep(), ast_audiohook_write_list(), ast_bridge_call(), ast_feature_request_and_dial(), ast_jb_put(), background_detect_exec(), cb_events(), channel_spy(), cli_console_dial(), conf_exec(), conf_run(), console_dial(), dahdi_bridge(), dahdi_read(), dictate_exec(), disa_exec(), do_immediate_setup(), echo_exec(), eivr_comm(), gtalk_handle_dtmf(), handle_recordfile(), handle_request(), handle_request_info(), handle_speechrecognize(), jingle_handle_dtmf(), keypad_digit(), 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(), unistim_do_senddigit(), unistim_senddigit_end(), volume_callback(), and wait_for_winner().

#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 175 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 454 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_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(), builtin_atxfer(), calc_cost(), channel_spy(), check_goto_on_transfer(), conf_exec(), conf_flush(), conf_free(), conf_run(), create_jb(), dahdi_bridge(), dial_exec_full(), dictate_exec(), disa_exec(), do_idle_thread(), do_waiting(), echo_exec(), eivr_comm(), find_cache(), gen_generate(), handle_cli_file_convert(), handle_recordfile(), handle_speechrecognize(), iax_park_thread(), ices_exec(), isAnsweringMachine(), 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(), read_frame(), receive_dtmf_digits(), recordthread(), rpt(), 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 196 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(), process_rfc3389(), send_tone_burst(), send_waveform_to_channel(), slinear_read(), sms_generate(), usbradio_read(), vox_read(), and wav_read().

#define AST_HTML_BEGIN   4

Beginning frame

Definition at line 218 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_DATA   2

Data frame

Definition at line 216 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_END   8

End frame

Definition at line 220 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_LDCOMPLETE   16

Load is complete

Definition at line 222 of file frame.h.

Referenced by ast_frame_dump(), and sendurl_exec().

#define AST_HTML_LINKREJECT   20

Reject link request

Definition at line 230 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_LINKURL   18

Send URL, and track

Definition at line 226 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_NOSUPPORT   17

Peer is unable to support HTML

Definition at line 224 of file frame.h.

Referenced by ast_frame_dump(), and sendurl_exec().

#define AST_HTML_UNLINK   19

No more HTML linkage

Definition at line 228 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_URL   1

Sending a URL

Definition at line 214 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 202 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 200 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 204 of file frame.h.

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

#define AST_MIN_OFFSET   32

Definition at line 197 of file frame.h.

Referenced by __ast_smoother_feed().

#define AST_MODEM_T38   1

T.38 Fax-over-IP

Definition at line 208 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 210 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 368 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 390 of file frame.h.

Referenced by dahdi_setoption().

#define AST_OPTION_FLAG_ACCEPT   1

Definition at line 351 of file frame.h.

#define AST_OPTION_FLAG_ANSWER   5

Definition at line 354 of file frame.h.

#define AST_OPTION_FLAG_QUERY   4

Definition at line 353 of file frame.h.

#define AST_OPTION_FLAG_REJECT   2

Definition at line 352 of file frame.h.

#define AST_OPTION_FLAG_REQUEST   0

Definition at line 350 of file frame.h.

Referenced by ast_bridge_call(), and iax2_setoption().

#define AST_OPTION_FLAG_WTF   6

Definition at line 355 of file frame.h.

#define AST_OPTION_OPRMODE   7

Definition at line 387 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 365 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 384 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 396 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 362 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 359 of file frame.h.

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

#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 376 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 557 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 562 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 563 of file frame.h.

#define AST_SMOOTHER_FLAG_BE   (1 << 1)

Definition at line 347 of file frame.h.

Referenced by ast_rtp_write().

#define AST_SMOOTHER_FLAG_G729   (1 << 0)

Definition at line 346 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

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 };

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
_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
AST_CONTROL_SRCCHANGE  Media source has changed and requires a new RTP SSRC

Definition at line 288 of file frame.h.

00288                             {
00289    AST_CONTROL_HANGUP = 1,    /*!< Other end has hungup */
00290    AST_CONTROL_RING = 2,      /*!< Local ring */
00291    AST_CONTROL_RINGING = 3,   /*!< Remote end is ringing */
00292    AST_CONTROL_ANSWER = 4,    /*!< Remote end has answered */
00293    AST_CONTROL_BUSY = 5,      /*!< Remote end is busy */
00294    AST_CONTROL_TAKEOFFHOOK = 6,  /*!< Make it go off hook */
00295    AST_CONTROL_OFFHOOK = 7,   /*!< Line is off hook */
00296    AST_CONTROL_CONGESTION = 8,   /*!< Congestion (circuits busy) */
00297    AST_CONTROL_FLASH = 9,     /*!< Flash hook */
00298    AST_CONTROL_WINK = 10,     /*!< Wink */
00299    AST_CONTROL_OPTION = 11,   /*!< Set a low-level option */
00300    AST_CONTROL_RADIO_KEY = 12,   /*!< Key Radio */
00301    AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */
00302    AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */
00303    AST_CONTROL_PROCEEDING = 15,  /*!< Indicate CALL PROCEEDING */
00304    AST_CONTROL_HOLD = 16,     /*!< Indicate call is placed on hold */
00305    AST_CONTROL_UNHOLD = 17,   /*!< Indicate call is left from hold */
00306    AST_CONTROL_VIDUPDATE = 18,   /*!< Indicate video frame update */
00307    _XXX_AST_CONTROL_T38 = 19, /*!< T38 state change request/notification \deprecated This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead. */
00308    AST_CONTROL_SRCUPDATE = 20,     /*!< Indicate source of media has changed */
00309    AST_CONTROL_T38_PARAMETERS = 24, /*!< T38 state change request/notification with parameters */
00310    AST_CONTROL_SRCCHANGE = 25,  /*!< Media source has changed and requires a new RTP SSRC */
00311 };

enum ast_control_t38

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 313 of file frame.h.

00313                      {
00314    AST_T38_REQUEST_NEGOTIATE = 1,   /*!< Request T38 on a channel (voice to fax) */
00315    AST_T38_REQUEST_TERMINATE, /*!< Terminate T38 on a channel (fax to voice) */
00316    AST_T38_NEGOTIATED,     /*!< T38 negotiated (fax mode) */
00317    AST_T38_TERMINATED,     /*!< T38 terminated (back to voice) */
00318    AST_T38_REFUSED         /*!< T38 refused for some reason (usually rejected by remote end) */
00319 };

enum ast_control_t38_rate

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 321 of file frame.h.

enum ast_control_t38_rate_management

Enumerator:
AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF 
AST_T38_RATE_MANAGEMENT_LOCAL_TCF 

Definition at line 330 of file frame.h.

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. 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 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 199 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.

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 642 of file frame.c.

References ARRAY_LEN, AST_FORMAT_LIST, and ast_format_list::desc.

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

00643 {
00644    int x;
00645    char *ret = "unknown";
00646    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00647       if (AST_FORMAT_LIST[x].bits == codec) {
00648          ret = AST_FORMAT_LIST[x].desc;
00649          break;
00650       }
00651    }
00652    return ret;
00653 }

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 1199 of file frame.c.

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

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

01200 {
01201    int x, ret = 0, slot;
01202 
01203    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01204       slot = pref->order[x];
01205 
01206       if (!slot)
01207          break;
01208       if (formats & AST_FORMAT_LIST[slot-1].bits) {
01209          ret = AST_FORMAT_LIST[slot-1].bits;
01210          break;
01211       }
01212    }
01213    if (ret & AST_FORMAT_AUDIO_MASK)
01214       return ret;
01215 
01216    ast_debug(4, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec");
01217 
01218       return find_best ? ast_best_codec(formats) : 0;
01219 }

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 1463 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().

01464 {
01465    int len = 0;
01466 
01467    /* XXX Still need speex, g723, and lpc10 XXX */ 
01468    switch(format) {
01469    case AST_FORMAT_G723_1:
01470       len = (samples / 240) * 20;
01471       break;
01472    case AST_FORMAT_ILBC:
01473       len = (samples / 240) * 50;
01474       break;
01475    case AST_FORMAT_GSM:
01476       len = (samples / 160) * 33;
01477       break;
01478    case AST_FORMAT_G729A:
01479       len = samples / 8;
01480       break;
01481    case AST_FORMAT_SLINEAR:
01482    case AST_FORMAT_SLINEAR16:
01483       len = samples * 2;
01484       break;
01485    case AST_FORMAT_ULAW:
01486    case AST_FORMAT_ALAW:
01487       len = samples;
01488       break;
01489    case AST_FORMAT_G722:
01490    case AST_FORMAT_ADPCM:
01491    case AST_FORMAT_G726:
01492    case AST_FORMAT_G726_AAL2:
01493       len = samples / 2;
01494       break;
01495    default:
01496       ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format));
01497    }
01498 
01499    return len;
01500 }

int ast_codec_get_samples ( struct ast_frame f  ) 

Returns the number of samples contained in the frame.

Definition at line 1419 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(), f, g723_samples(), LOG_WARNING, and speex_samples().

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

01420 {
01421    int samples=0;
01422    switch(f->subclass) {
01423    case AST_FORMAT_SPEEX:
01424       samples = speex_samples(f->data.ptr, f->datalen);
01425       break;
01426    case AST_FORMAT_G723_1:
01427       samples = g723_samples(f->data.ptr, f->datalen);
01428       break;
01429    case AST_FORMAT_ILBC:
01430       samples = 240 * (f->datalen / 50);
01431       break;
01432    case AST_FORMAT_GSM:
01433       samples = 160 * (f->datalen / 33);
01434       break;
01435    case AST_FORMAT_G729A:
01436       samples = f->datalen * 8;
01437       break;
01438    case AST_FORMAT_SLINEAR:
01439    case AST_FORMAT_SLINEAR16:
01440       samples = f->datalen / 2;
01441       break;
01442    case AST_FORMAT_LPC10:
01443       /* assumes that the RTP packet contains one LPC10 frame */
01444       samples = 22 * 8;
01445       samples += (((char *)(f->data.ptr))[7] & 0x1) * 8;
01446       break;
01447    case AST_FORMAT_ULAW:
01448    case AST_FORMAT_ALAW:
01449       samples = f->datalen;
01450       break;
01451    case AST_FORMAT_G722:
01452    case AST_FORMAT_ADPCM:
01453    case AST_FORMAT_G726:
01454    case AST_FORMAT_G726_AAL2:
01455       samples = f->datalen * 2;
01456       break;
01457    default:
01458       ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass));
01459    }
01460    return samples;
01461 }

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

Gets duration in ms of interpolation frame for a format.

Definition at line 645 of file frame.h.

References AST_FORMAT_ILBC.

Referenced by __get_from_jb(), and jb_get_and_deliver().

00646 { 
00647    return (format == AST_FORMAT_ILBC) ? 30 : 20;
00648 }

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 1059 of file frame.c.

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

Referenced by ast_parse_allow_disallow().

01060 {
01061    int x, newindex = 0;
01062 
01063    ast_codec_pref_remove(pref, format);
01064 
01065    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01066       if (AST_FORMAT_LIST[x].bits == format) {
01067          newindex = x + 1;
01068          break;
01069       }
01070    }
01071 
01072    if (newindex) {
01073       for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01074          if (!pref->order[x]) {
01075             pref->order[x] = newindex;
01076             break;
01077          }
01078       }
01079    }
01080 
01081    return x;
01082 }

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 962 of file frame.c.

References ast_codec_pref::order.

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

00963 {
00964    int x, differential = (int) 'A', mem;
00965    char *from, *to;
00966 
00967    if (right) {
00968       from = pref->order;
00969       to = buf;
00970       mem = size;
00971    } else {
00972       to = pref->order;
00973       from = buf;
00974       mem = 32;
00975    }
00976 
00977    memset(to, 0, mem);
00978    for (x = 0; x < 32 ; x++) {
00979       if (!from[x])
00980          break;
00981       to[x] = right ? (from[x] + differential) : (from[x] - differential);
00982    }
00983 }

struct ast_format_list ast_codec_pref_getsize ( struct ast_codec_pref pref,
int  format 
)

Get packet size for codec.

Definition at line 1160 of file frame.c.

References ARRAY_LEN, AST_FORMAT_LIST, 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().

01161 {
01162    int x, idx = -1, framems = 0;
01163    struct ast_format_list fmt = { 0, };
01164 
01165    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01166       if (AST_FORMAT_LIST[x].bits == format) {
01167          fmt = AST_FORMAT_LIST[x];
01168          idx = x;
01169          break;
01170       }
01171    }
01172 
01173    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01174       if (pref->order[x] == (idx + 1)) {
01175          framems = pref->framing[x];
01176          break;
01177       }
01178    }
01179 
01180    /* size validation */
01181    if (!framems)
01182       framems = AST_FORMAT_LIST[idx].def_ms;
01183 
01184    if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */
01185       framems -= framems % AST_FORMAT_LIST[idx].inc_ms;
01186 
01187    if (framems < AST_FORMAT_LIST[idx].min_ms)
01188       framems = AST_FORMAT_LIST[idx].min_ms;
01189 
01190    if (framems > AST_FORMAT_LIST[idx].max_ms)
01191       framems = AST_FORMAT_LIST[idx].max_ms;
01192 
01193    fmt.cur_ms = framems;
01194 
01195    return fmt;
01196 }

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 1020 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(), handle_cli_iax2_show_peer(), jingle_accept_call(), print_codec_to_cli(), and socket_process().

01021 {
01022    int slot = 0;
01023 
01024    if ((idx >= 0) && (idx < sizeof(pref->order))) {
01025       slot = pref->order[idx];
01026    }
01027 
01028    return slot ? AST_FORMAT_LIST[slot - 1].bits : 0;
01029 }

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 1085 of file frame.c.

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

Referenced by create_addr().

01086 {
01087    int x, newindex = 0;
01088 
01089    /* First step is to get the codecs "index number" */
01090    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01091       if (AST_FORMAT_LIST[x].bits == format) {
01092          newindex = x + 1;
01093          break;
01094       }
01095    }
01096    /* Done if its unknown */
01097    if (!newindex)
01098       return;
01099 
01100    /* Now find any existing occurrence, or the end */
01101    for (x = 0; x < 32; x++) {
01102       if (!pref->order[x] || pref->order[x] == newindex)
01103          break;
01104    }
01105 
01106    if (only_if_existing && !pref->order[x])
01107       return;
01108 
01109    /* Move down to make space to insert - either all the way to the end,
01110       or as far as the existing location (which will be overwritten) */
01111    for (; x > 0; x--) {
01112       pref->order[x] = pref->order[x - 1];
01113       pref->framing[x] = pref->framing[x - 1];
01114    }
01115 
01116    /* And insert the new entry */
01117    pref->order[0] = newindex;
01118    pref->framing[0] = 0; /* ? */
01119 }

void ast_codec_pref_remove ( struct ast_codec_pref pref,
int  format 
)

Remove audio a codec from a preference list.

Definition at line 1032 of file frame.c.

References ARRAY_LEN, AST_FORMAT_LIST, and ast_codec_pref::order.

Referenced by ast_codec_pref_append(), and ast_parse_allow_disallow().

01033 {
01034    struct ast_codec_pref oldorder;
01035    int x, y = 0;
01036    int slot;
01037    int size;
01038 
01039    if (!pref->order[0])
01040       return;
01041 
01042    memcpy(&oldorder, pref, sizeof(oldorder));
01043    memset(pref, 0, sizeof(*pref));
01044 
01045    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01046       slot = oldorder.order[x];
01047       size = oldorder.framing[x];
01048       if (! slot)
01049          break;
01050       if (AST_FORMAT_LIST[slot-1].bits != format) {
01051          pref->order[y] = slot;
01052          pref->framing[y++] = size;
01053       }
01054    }
01055    
01056 }

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

Set packet size for codec.

Definition at line 1122 of file frame.c.

References ARRAY_LEN, AST_FORMAT_LIST, 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_a_audio().

01123 {
01124    int x, idx = -1;
01125 
01126    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01127       if (AST_FORMAT_LIST[x].bits == format) {
01128          idx = x;
01129          break;
01130       }
01131    }
01132 
01133    if (idx < 0)
01134       return -1;
01135 
01136    /* size validation */
01137    if (!framems)
01138       framems = AST_FORMAT_LIST[idx].def_ms;
01139 
01140    if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */
01141       framems -= framems % AST_FORMAT_LIST[idx].inc_ms;
01142 
01143    if (framems < AST_FORMAT_LIST[idx].min_ms)
01144       framems = AST_FORMAT_LIST[idx].min_ms;
01145 
01146    if (framems > AST_FORMAT_LIST[idx].max_ms)
01147       framems = AST_FORMAT_LIST[idx].max_ms;
01148 
01149    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01150       if (pref->order[x] == (idx + 1)) {
01151          pref->framing[x] = framems;
01152          break;
01153       }
01154    }
01155 
01156    return x;
01157 }

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 985 of file frame.c.

References ast_codec_pref_index(), and ast_getformatname().

Referenced by dump_prefs(), and socket_process().

00986 {
00987    int x, codec; 
00988    size_t total_len, slen;
00989    char *formatname;
00990    
00991    memset(buf,0,size);
00992    total_len = size;
00993    buf[0] = '(';
00994    total_len--;
00995    for(x = 0; x < 32 ; x++) {
00996       if (total_len <= 0)
00997          break;
00998       if (!(codec = ast_codec_pref_index(pref,x)))
00999          break;
01000       if ((formatname = ast_getformatname(codec))) {
01001          slen = strlen(formatname);
01002          if (slen > total_len)
01003             break;
01004          strncat(buf, formatname, total_len - 1); /* safe */
01005          total_len -= slen;
01006       }
01007       if (total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) {
01008          strncat(buf, "|", total_len - 1); /* safe */
01009          total_len--;
01010       }
01011    }
01012    if (total_len) {
01013       strncat(buf, ")", total_len - 1); /* safe */
01014       total_len--;
01015    }
01016 
01017    return size - total_len;
01018 }

static force_inline int ast_format_rate ( int  format  )  [static]

Get the sample rate for a given format.

Definition at line 672 of file frame.h.

References AST_FORMAT_G722, and AST_FORMAT_SLINEAR16.

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

00673 {
00674    if (format == AST_FORMAT_G722 || format == AST_FORMAT_SLINEAR16)
00675       return 16000;
00676 
00677    return 8000;
00678 }

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 1502 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(), audiohook_volume_callback(), conf_run(), and volume_callback().

01503 {
01504    int count;
01505    short *fdata = f->data.ptr;
01506    short adjust_value = abs(adjustment);
01507 
01508    if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR))
01509       return -1;
01510 
01511    if (!adjustment)
01512       return 0;
01513 
01514    for (count = 0; count < f->samples; count++) {
01515       if (adjustment > 0) {
01516          ast_slinear_saturated_multiply(&fdata[count], &adjust_value);
01517       } else if (adjustment < 0) {
01518          ast_slinear_saturated_divide(&fdata[count], &adjust_value);
01519       }
01520    }
01521 
01522    return 0;
01523 }

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

Dump a frame for debugging purposes

Definition at line 744 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, f, ast_control_t38_parameters::request_response, and term_color().

Referenced by __ast_read(), and ast_write().

00745 {
00746    const char noname[] = "unknown";
00747    char ftype[40] = "Unknown Frametype";
00748    char cft[80];
00749    char subclass[40] = "Unknown Subclass";
00750    char csub[80];
00751    char moreinfo[40] = "";
00752    char cn[60];
00753    char cp[40];
00754    char cmn[40];
00755    const char *message = "Unknown";
00756 
00757    if (!name)
00758       name = noname;
00759 
00760 
00761    if (!f) {
00762       ast_verbose("%s [ %s (NULL) ] [%s]\n", 
00763          term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00764          term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 
00765          term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00766       return;
00767    }
00768    /* XXX We should probably print one each of voice and video when the format changes XXX */
00769    if (f->frametype == AST_FRAME_VOICE)
00770       return;
00771    if (f->frametype == AST_FRAME_VIDEO)
00772       return;
00773    switch(f->frametype) {
00774    case AST_FRAME_DTMF_BEGIN:
00775       strcpy(ftype, "DTMF Begin");
00776       subclass[0] = f->subclass;
00777       subclass[1] = '\0';
00778       break;
00779    case AST_FRAME_DTMF_END:
00780       strcpy(ftype, "DTMF End");
00781       subclass[0] = f->subclass;
00782       subclass[1] = '\0';
00783       break;
00784    case AST_FRAME_CONTROL:
00785       strcpy(ftype, "Control");
00786       switch(f->subclass) {
00787       case AST_CONTROL_HANGUP:
00788          strcpy(subclass, "Hangup");
00789          break;
00790       case AST_CONTROL_RING:
00791          strcpy(subclass, "Ring");
00792          break;
00793       case AST_CONTROL_RINGING:
00794          strcpy(subclass, "Ringing");
00795          break;
00796       case AST_CONTROL_ANSWER:
00797          strcpy(subclass, "Answer");
00798          break;
00799       case AST_CONTROL_BUSY:
00800          strcpy(subclass, "Busy");
00801          break;
00802       case AST_CONTROL_TAKEOFFHOOK:
00803          strcpy(subclass, "Take Off Hook");
00804          break;
00805       case AST_CONTROL_OFFHOOK:
00806          strcpy(subclass, "Line Off Hook");
00807          break;
00808       case AST_CONTROL_CONGESTION:
00809          strcpy(subclass, "Congestion");
00810          break;
00811       case AST_CONTROL_FLASH:
00812          strcpy(subclass, "Flash");
00813          break;
00814       case AST_CONTROL_WINK:
00815          strcpy(subclass, "Wink");
00816          break;
00817       case AST_CONTROL_OPTION:
00818          strcpy(subclass, "Option");
00819          break;
00820       case AST_CONTROL_RADIO_KEY:
00821          strcpy(subclass, "Key Radio");
00822          break;
00823       case AST_CONTROL_RADIO_UNKEY:
00824          strcpy(subclass, "Unkey Radio");
00825          break;
00826       case AST_CONTROL_HOLD:
00827          strcpy(subclass, "Hold");
00828          break;
00829       case AST_CONTROL_UNHOLD:
00830          strcpy(subclass, "Unhold");
00831          break;
00832       case AST_CONTROL_T38_PARAMETERS:
00833          if (f->datalen != sizeof(struct ast_control_t38_parameters)) {
00834             message = "Invalid";
00835          } else {
00836             struct ast_control_t38_parameters *parameters = f->data.ptr;
00837             enum ast_control_t38 state = parameters->request_response;
00838             if (state == AST_T38_REQUEST_NEGOTIATE)
00839                message = "Negotiation Requested";
00840             else if (state == AST_T38_REQUEST_TERMINATE)
00841                message = "Negotiation Request Terminated";
00842             else if (state == AST_T38_NEGOTIATED)
00843                message = "Negotiated";
00844             else if (state == AST_T38_TERMINATED)
00845                message = "Terminated";
00846             else if (state == AST_T38_REFUSED)
00847                message = "Refused";
00848          }
00849          snprintf(subclass, sizeof(subclass), "T38_Parameters/%s", message);
00850          break;
00851       case -1:
00852          strcpy(subclass, "Stop generators");
00853          break;
00854       default:
00855          snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass);
00856       }
00857       break;
00858    case AST_FRAME_NULL:
00859       strcpy(ftype, "Null Frame");
00860       strcpy(subclass, "N/A");
00861       break;
00862    case AST_FRAME_IAX:
00863       /* Should never happen */
00864       strcpy(ftype, "IAX Specific");
00865       snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass);
00866       break;
00867    case AST_FRAME_TEXT:
00868       strcpy(ftype, "Text");
00869       strcpy(subclass, "N/A");
00870       ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo));
00871       break;
00872    case AST_FRAME_IMAGE:
00873       strcpy(ftype, "Image");
00874       snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass));
00875       break;
00876    case AST_FRAME_HTML:
00877       strcpy(ftype, "HTML");
00878       switch(f->subclass) {
00879       case AST_HTML_URL:
00880          strcpy(subclass, "URL");
00881          ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo));
00882          break;
00883       case AST_HTML_DATA:
00884          strcpy(subclass, "Data");
00885          break;
00886       case AST_HTML_BEGIN:
00887          strcpy(subclass, "Begin");
00888          break;
00889       case AST_HTML_END:
00890          strcpy(subclass, "End");
00891          break;
00892       case AST_HTML_LDCOMPLETE:
00893          strcpy(subclass, "Load Complete");
00894          break;
00895       case AST_HTML_NOSUPPORT:
00896          strcpy(subclass, "No Support");
00897          break;
00898       case AST_HTML_LINKURL:
00899          strcpy(subclass, "Link URL");
00900          ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo));
00901          break;
00902       case AST_HTML_UNLINK:
00903          strcpy(subclass, "Unlink");
00904          break;
00905       case AST_HTML_LINKREJECT:
00906          strcpy(subclass, "Link Reject");
00907          break;
00908       default:
00909          snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass);
00910          break;
00911       }
00912       break;
00913    case AST_FRAME_MODEM:
00914       strcpy(ftype, "Modem");
00915       switch (f->subclass) {
00916       case AST_MODEM_T38:
00917          strcpy(subclass, "T.38");
00918          break;
00919       case AST_MODEM_V150:
00920          strcpy(subclass, "V.150");
00921          break;
00922       default:
00923          snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass);
00924          break;
00925       }
00926       break;
00927    default:
00928       snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype);
00929    }
00930    if (!ast_strlen_zero(moreinfo))
00931       ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n",  
00932              term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00933              term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)),
00934              f->frametype, 
00935              term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)),
00936              f->subclass, 
00937              term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)),
00938              term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00939    else
00940       ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n",  
00941              term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00942              term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)),
00943              f->frametype, 
00944              term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)),
00945              f->subclass, 
00946              term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00947 }

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 365 of file frame.c.

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

Referenced by mixmonitor_thread().

00366 {
00367    struct ast_frame *next;
00368 
00369    for (next = AST_LIST_NEXT(frame, frame_list);
00370         frame;
00371         frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) {
00372       __frame_free(frame, cache);
00373    }
00374 }

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 1525 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.

01526 {
01527    int count;
01528    short *data1, *data2;
01529 
01530    if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR))
01531       return -1;
01532 
01533    if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR))
01534       return -1;
01535 
01536    if (f1->samples != f2->samples)
01537       return -1;
01538 
01539    for (count = 0, data1 = f1->data.ptr, data2 = f2->data.ptr;
01540         count < f1->samples;
01541         count++, data1++, data2++)
01542       ast_slinear_saturated_add(data1, data2);
01543 
01544    return 0;
01545 }

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 464 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, f, frame_cache, frames, ast_frame::frametype, ast_frame::len, len(), ast_frame::mallocd, ast_frame::mallocd_hdr_len, ast_frame::offset, ast_frame::ptr, ast_frame::samples, ast_frame::seqno, 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_single(), autoservice_run(), process_rfc2833(), recordthread(), and rpt().

00465 {
00466    struct ast_frame *out = NULL;
00467    int len, srclen = 0;
00468    void *buf = NULL;
00469 
00470 #if !defined(LOW_MEMORY)
00471    struct ast_frame_cache *frames;
00472 #endif
00473 
00474    /* Start with standard stuff */
00475    len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen;
00476    /* If we have a source, add space for it */
00477    /*
00478     * XXX Watch out here - if we receive a src which is not terminated
00479     * properly, we can be easily attacked. Should limit the size we deal with.
00480     */
00481    if (f->src)
00482       srclen = strlen(f->src);
00483    if (srclen > 0)
00484       len += srclen + 1;
00485    
00486 #if !defined(LOW_MEMORY)
00487    if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) {
00488       AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) {
00489          if (out->mallocd_hdr_len >= len) {
00490             size_t mallocd_len = out->mallocd_hdr_len;
00491 
00492             AST_LIST_REMOVE_CURRENT(frame_list);
00493             memset(out, 0, sizeof(*out));
00494             out->mallocd_hdr_len = mallocd_len;
00495             buf = out;
00496             frames->size--;
00497             break;
00498          }
00499       }
00500       AST_LIST_TRAVERSE_SAFE_END;
00501    }
00502 #endif
00503 
00504    if (!buf) {
00505       if (!(buf = ast_calloc_cache(1, len)))
00506          return NULL;
00507       out = buf;
00508       out->mallocd_hdr_len = len;
00509    }
00510 
00511    out->frametype = f->frametype;
00512    out->subclass = f->subclass;
00513    out->datalen = f->datalen;
00514    out->samples = f->samples;
00515    out->delivery = f->delivery;
00516    /* Set us as having malloc'd header only, so it will eventually
00517       get freed. */
00518    out->mallocd = AST_MALLOCD_HDR;
00519    out->offset = AST_FRIENDLY_OFFSET;
00520    if (out->datalen) {
00521       out->data.ptr = buf + sizeof(*out) + AST_FRIENDLY_OFFSET;
00522       memcpy(out->data.ptr, f->data.ptr, out->datalen);  
00523    } else {
00524       out->data.uint32 = f->data.uint32;
00525    }
00526    if (srclen > 0) {
00527       /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */
00528       char *src;
00529       out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen;
00530       src = (char *) out->src;
00531       /* Must have space since we allocated for it */
00532       strcpy(src, f->src);
00533    }
00534    ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO);
00535    out->ts = f->ts;
00536    out->len = f->len;
00537    out->seqno = f->seqno;
00538    return out;
00539 }

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 381 of file frame.c.

References ast_copy_flags, ast_frame_header_new(), ast_frdup(), ast_free, 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, ast_frame::ts, and ast_frame::uint32.

Referenced by __ast_answer(), ast_rtp_read(), ast_slinfactory_feed(), ast_trans_frameout(), ast_write(), autoservice_run(), dahdi_decoder_frameout(), dahdi_encoder_frameout(), jpeg_read_image(), and read_frame().

00382 {
00383    struct ast_frame *out;
00384    void *newdata;
00385 
00386    /* if none of the existing frame is malloc'd, let ast_frdup() do it
00387       since it is more efficient
00388    */
00389    if (fr->mallocd == 0) {
00390       return ast_frdup(fr);
00391    }
00392 
00393    /* if everything is already malloc'd, we are done */
00394    if ((fr->mallocd & (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) ==
00395        (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) {
00396       return fr;
00397    }
00398 
00399    if (!(fr->mallocd & AST_MALLOCD_HDR)) {
00400       /* Allocate a new header if needed */
00401       if (!(out = ast_frame_header_new())) {
00402          return NULL;
00403       }
00404       out->frametype = fr->frametype;
00405       out->subclass = fr->subclass;
00406       out->datalen = fr->datalen;
00407       out->samples = fr->samples;
00408       out->offset = fr->offset;
00409       /* Copy the timing data */
00410       ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO);
00411       if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) {
00412          out->ts = fr->ts;
00413          out->len = fr->len;
00414          out->seqno = fr->seqno;
00415       }
00416    } else {
00417       out = fr;
00418    }
00419    
00420    if (!(fr->mallocd & AST_MALLOCD_SRC) && fr->src) {
00421       if (!(out->src = ast_strdup(fr->src))) {
00422          if (out != fr) {
00423             ast_free(out);
00424          }
00425          return NULL;
00426       }
00427    } else {
00428       out->src = fr->src;
00429       fr->src = NULL;
00430       fr->mallocd &= ~AST_MALLOCD_SRC;
00431    }
00432    
00433    if (!(fr->mallocd & AST_MALLOCD_DATA))  {
00434       if (!fr->datalen) {
00435          out->data.uint32 = fr->data.uint32;
00436          out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC;
00437          return out;
00438       }
00439       if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) {
00440          if (out->src != fr->src) {
00441             ast_free((void *) out->src);
00442          }
00443          if (out != fr) {
00444             ast_free(out);
00445          }
00446          return NULL;
00447       }
00448       newdata += AST_FRIENDLY_OFFSET;
00449       out->offset = AST_FRIENDLY_OFFSET;
00450       out->datalen = fr->datalen;
00451       memcpy(newdata, fr->data.ptr, fr->datalen);
00452       out->data.ptr = newdata;
00453    } else {
00454       out->data = fr->data;
00455       memset(&fr->data, 0, sizeof(fr->data));
00456       fr->mallocd &= ~AST_MALLOCD_DATA;
00457    }
00458 
00459    out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA;
00460    
00461    return out;
00462 }

struct ast_format_list* ast_get_format_list ( size_t *  size  ) 

Definition at line 557 of file frame.c.

References ARRAY_LEN, and AST_FORMAT_LIST.

00558 {
00559    *size = ARRAY_LEN(AST_FORMAT_LIST);
00560    return AST_FORMAT_LIST;
00561 }

struct ast_format_list* ast_get_format_list_index ( int  index  ) 

Definition at line 552 of file frame.c.

References AST_FORMAT_LIST.

00553 {
00554    return &AST_FORMAT_LIST[idx];
00555 }

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 624 of file frame.c.

References ARRAY_LEN, ast_expand_codec_alias(), AST_FORMAT_LIST, 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().

00625 {
00626    int x, all, format = 0;
00627 
00628    all = strcasecmp(name, "all") ? 0 : 1;
00629    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00630       if (all || 
00631            !strcasecmp(AST_FORMAT_LIST[x].name,name) ||
00632            !strcasecmp(AST_FORMAT_LIST[x].name, ast_expand_codec_alias(name))) {
00633          format |= AST_FORMAT_LIST[x].bits;
00634          if (!all)
00635             break;
00636       }
00637    }
00638 
00639    return format;
00640 }

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 563 of file frame.c.

References ARRAY_LEN, AST_FORMAT_LIST, 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().

00564 {
00565    int x;
00566    char *ret = "unknown";
00567    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00568       if (AST_FORMAT_LIST[x].bits == format) {
00569          ret = AST_FORMAT_LIST[x].name;
00570          break;
00571       }
00572    }
00573    return ret;
00574 }

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 576 of file frame.c.

References ARRAY_LEN, ast_copy_string(), AST_FORMAT_LIST, 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(), process_sdp(), serialize_showchan(), set_format(), show_channels_cb(), sip_new(), sip_request_call(), sip_show_channel(), sip_show_settings(), and sip_write().

00577 {
00578    int x;
00579    unsigned len;
00580    char *start, *end = buf;
00581 
00582    if (!size)
00583       return buf;
00584    snprintf(end, size, "0x%x (", format);
00585    len = strlen(end);
00586    end += len;
00587    size -= len;
00588    start = end;
00589    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00590       if (AST_FORMAT_LIST[x].bits & format) {
00591          snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name);
00592          len = strlen(end);
00593          end += len;
00594          size -= len;
00595       }
00596    }
00597    if (start == end)
00598       ast_copy_string(start, "nothing)", size);
00599    else if (size > 1)
00600       *(end -1) = ')';
00601    return buf;
00602 }

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 1221 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().

01222 {
01223    int errors = 0;
01224    char *parse = NULL, *this = NULL, *psize = NULL;
01225    int format = 0, framems = 0;
01226 
01227    parse = ast_strdupa(list);
01228    while ((this = strsep(&parse, ","))) {
01229       framems = 0;
01230       if ((psize = strrchr(this, ':'))) {
01231          *psize++ = '\0';
01232          ast_debug(1, "Packetization for codec: %s is %s\n", this, psize);
01233          framems = atoi(psize);
01234          if (framems < 0) {
01235             framems = 0;
01236             errors++;
01237             ast_log(LOG_WARNING, "Bad packetization value for codec %s\n", this);
01238          }
01239       }
01240       if (!(format = ast_getformatbyname(this))) {
01241          ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this);
01242          errors++;
01243          continue;
01244       }
01245 
01246       if (mask) {
01247          if (allowing)
01248             *mask |= format;
01249          else
01250             *mask &= ~format;
01251       }
01252 
01253       /* Set up a preference list for audio. Do not include video in preferences 
01254          since we can not transcode video and have to use whatever is offered
01255        */
01256       if (pref && (format & AST_FORMAT_AUDIO_MASK)) {
01257          if (strcasecmp(this, "all")) {
01258             if (allowing) {
01259                ast_codec_pref_append(pref, format);
01260                ast_codec_pref_setsize(pref, format, framems);
01261             }
01262             else
01263                ast_codec_pref_remove(pref, format);
01264          } else if (!allowing) {
01265             memset(pref, 0, sizeof(*pref));
01266          }
01267       }
01268    }
01269    return errors;
01270 }

void ast_smoother_free ( struct ast_smoother s  ) 

Definition at line 284 of file frame.c.

References ast_free, and s.

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 s.

00185 {
00186    return s->flags;
00187 }

struct ast_smoother* ast_smoother_new ( int  bytes  ) 

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  ) 

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(), len(), LOG_WARNING, and s.

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 s, 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 s.

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 s.

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 s.

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 541 of file frame.c.

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

00542 {
00543    int i;
00544    unsigned short *dst_s = dst;
00545    const unsigned short *src_s = src;
00546 
00547    for (i = 0; i < samples; i++)
00548       dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8);
00549 }


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 122 of file frame.c.

Referenced by __ast_read(), __oh323_rtp_create(), __oh323_update_info(), agent_new(), agent_read(), ast_channel_masquerade(), ast_channel_setwhentohangup_tv(), ast_do_masquerade(), ast_rtcp_read(), ast_rtp_read(), ast_softhangup_nolock(), ast_udptl_read(), conf_run(), console_read(), create_dtmf_frame(), gtalk_rtp_read(), handle_request_invite(), handle_response_invite(), iax2_read(), jingle_rtp_read(), local_read(), mgcp_rtp_read(), oh323_read(), oh323_rtp_read(), process_sdp(), sip_read(), sip_rtp_read(), skinny_rtp_read(), unistim_rtp_read(), and wakeup_sub().


Generated on Wed Aug 18 22:34:23 2010 for Asterisk - the Open Source PBX by  doxygen 1.4.7