Fri Jul 24 00:41:44 2009

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_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), AST_FRFLAG_FROM_TRANSLATOR = (1 << 1), AST_FRFLAG_FROM_DSP = (1 << 2), AST_FRFLAG_FROM_FILESTREAM = (1 << 3) }
enum  ast_control_frame_type {
  AST_CONTROL_HANGUP = 1, AST_CONTROL_RING = 2, AST_CONTROL_RINGING = 3, AST_CONTROL_ANSWER = 4,
  AST_CONTROL_BUSY = 5, AST_CONTROL_TAKEOFFHOOK = 6, AST_CONTROL_OFFHOOK = 7, AST_CONTROL_CONGESTION = 8,
  AST_CONTROL_FLASH = 9, AST_CONTROL_WINK = 10, AST_CONTROL_OPTION = 11, AST_CONTROL_RADIO_KEY = 12,
  AST_CONTROL_RADIO_UNKEY = 13, AST_CONTROL_PROGRESS = 14, AST_CONTROL_PROCEEDING = 15, AST_CONTROL_HOLD = 16,
  AST_CONTROL_UNHOLD = 17, AST_CONTROL_VIDUPDATE = 18, AST_CONTROL_T38 = 19, AST_CONTROL_SRCUPDATE = 20
}
enum  ast_control_t38 {
  AST_T38_REQUEST_NEGOTIATE = 1, AST_T38_REQUEST_TERMINATE, AST_T38_NEGOTIATED, AST_T38_TERMINATED,
  AST_T38_REFUSED
}
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.
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 255 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 251 of file frame.h.

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

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

Maximum audio mask

Definition at line 275 of file frame.h.

Referenced by add_sdp(), ast_best_codec(), ast_channel_make_compatible_helper(), 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 271 of file frame.h.

#define AST_FORMAT_G722   (1 << 12)

G.722

Definition at line 269 of file frame.h.

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

#define AST_FORMAT_G723_1   (1 << 0)

G.723.1 compression

Definition at line 245 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 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)

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

Definition at line 253 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 261 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 247 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 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)

iLBC Free Compression

Definition at line 265 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 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_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(), read_config(), 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 273 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 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)

Raw mu-law data (G.711)

Definition at line 249 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 290 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 466 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 465 of file frame.h.

Referenced by phone_read().

#define AST_FRAME_DTMF   AST_FRAME_DTMF_END

Definition at line 124 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_dsp_process(), 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(), transmit_audio(), 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 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 438 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_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_invite_replaces(), 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(), 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 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(), 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 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 ast_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 ast_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 ast_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(), 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 352 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 374 of file frame.h.

Referenced by dahdi_setoption().

#define AST_OPTION_FLAG_ACCEPT   1

Definition at line 335 of file frame.h.

#define AST_OPTION_FLAG_ANSWER   5

Definition at line 338 of file frame.h.

#define AST_OPTION_FLAG_QUERY   4

Definition at line 337 of file frame.h.

#define AST_OPTION_FLAG_REJECT   2

Definition at line 336 of file frame.h.

#define AST_OPTION_FLAG_REQUEST   0

Definition at line 334 of file frame.h.

Referenced by ast_bridge_call(), and iax2_setoption().

#define AST_OPTION_FLAG_WTF   6

Definition at line 339 of file frame.h.

#define AST_OPTION_OPRMODE   7

Definition at line 371 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 349 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 368 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 380 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 346 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 343 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 360 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 536 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 541 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 542 of file frame.h.

#define AST_SMOOTHER_FLAG_BE   (1 << 1)

Definition at line 331 of file frame.h.

Referenced by ast_rtp_write().

#define AST_SMOOTHER_FLAG_G729   (1 << 0)

Definition at line 330 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 };

enum ast_control_frame_type

Enumerator:
AST_CONTROL_HANGUP  Other end has hungup
AST_CONTROL_RING  Local ring
AST_CONTROL_RINGING  Remote end is ringing
AST_CONTROL_ANSWER  Remote end has answered
AST_CONTROL_BUSY  Remote end is busy
AST_CONTROL_TAKEOFFHOOK  Make it go off hook
AST_CONTROL_OFFHOOK  Line is off hook
AST_CONTROL_CONGESTION  Congestion (circuits busy)
AST_CONTROL_FLASH  Flash hook
AST_CONTROL_WINK  Wink
AST_CONTROL_OPTION  Set a low-level option
AST_CONTROL_RADIO_KEY  Key Radio
AST_CONTROL_RADIO_UNKEY  Un-Key Radio
AST_CONTROL_PROGRESS  Indicate PROGRESS
AST_CONTROL_PROCEEDING  Indicate CALL PROCEEDING
AST_CONTROL_HOLD  Indicate call is placed on hold
AST_CONTROL_UNHOLD  Indicate call is left from hold
AST_CONTROL_VIDUPDATE  Indicate video frame update
AST_CONTROL_T38  T38 state change request/notification
AST_CONTROL_SRCUPDATE  Indicate source of media has changed

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    AST_CONTROL_T38 = 19,      /*!< T38 state change request/notification */
00319    AST_CONTROL_SRCUPDATE = 20,     /*!< Indicate source of media has changed */
00320 };

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 322 of file frame.h.

00322                      {
00323    AST_T38_REQUEST_NEGOTIATE = 1,   /*!< Request T38 on a channel (voice to fax) */
00324    AST_T38_REQUEST_TERMINATE, /*!< Terminate T38 on a channel (fax to voice) */
00325    AST_T38_NEGOTIATED,     /*!< T38 negotiated (fax mode) */
00326    AST_T38_TERMINATED,     /*!< T38 terminated (back to voice) */
00327    AST_T38_REFUSED         /*!< T38 refused for some reason (usually rejected by remote end) */
00328 };

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 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 204 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.

00205 {
00206    if (f->frametype != AST_FRAME_VOICE) {
00207       ast_log(LOG_WARNING, "Huh?  Can't smooth a non-voice frame!\n");
00208       return -1;
00209    }
00210    if (!s->format) {
00211       s->format = f->subclass;
00212       s->samplesperbyte = (float)f->samples / (float)f->datalen;
00213    } else if (s->format != f->subclass) {
00214       ast_log(LOG_WARNING, "Smoother was working on %d format frames, now trying to feed %d?\n", s->format, f->subclass);
00215       return -1;
00216    }
00217    if (s->len + f->datalen > SMOOTHER_SIZE) {
00218       ast_log(LOG_WARNING, "Out of smoother space\n");
00219       return -1;
00220    }
00221    if (((f->datalen == s->size) ||
00222         ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) &&
00223        !s->opt &&
00224        !s->len &&
00225        (f->offset >= AST_MIN_OFFSET)) {
00226       /* Optimize by sending the frame we just got
00227          on the next read, thus eliminating the douple
00228          copy */
00229       if (swap)
00230          ast_swapcopy_samples(f->data.ptr, f->data.ptr, f->samples);
00231       s->opt = f;
00232       s->opt_needs_swap = swap ? 1 : 0;
00233       return 0;
00234    }
00235 
00236    return smoother_frame_feed(s, f, swap);
00237 }

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 631 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().

00632 {
00633    int x;
00634    char *ret = "unknown";
00635    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00636       if (AST_FORMAT_LIST[x].bits == codec) {
00637          ret = AST_FORMAT_LIST[x].desc;
00638          break;
00639       }
00640    }
00641    return ret;
00642 }

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 1221 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().

01222 {
01223    int x, ret = 0, slot;
01224 
01225    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01226       slot = pref->order[x];
01227 
01228       if (!slot)
01229          break;
01230       if (formats & AST_FORMAT_LIST[slot-1].bits) {
01231          ret = AST_FORMAT_LIST[slot-1].bits;
01232          break;
01233       }
01234    }
01235    if (ret & AST_FORMAT_AUDIO_MASK)
01236       return ret;
01237 
01238    ast_debug(4, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec");
01239 
01240       return find_best ? ast_best_codec(formats) : 0;
01241 }

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 1485 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().

01486 {
01487    int len = 0;
01488 
01489    /* XXX Still need speex, g723, and lpc10 XXX */ 
01490    switch(format) {
01491    case AST_FORMAT_G723_1:
01492       len = (samples / 240) * 20;
01493       break;
01494    case AST_FORMAT_ILBC:
01495       len = (samples / 240) * 50;
01496       break;
01497    case AST_FORMAT_GSM:
01498       len = (samples / 160) * 33;
01499       break;
01500    case AST_FORMAT_G729A:
01501       len = samples / 8;
01502       break;
01503    case AST_FORMAT_SLINEAR:
01504    case AST_FORMAT_SLINEAR16:
01505       len = samples * 2;
01506       break;
01507    case AST_FORMAT_ULAW:
01508    case AST_FORMAT_ALAW:
01509       len = samples;
01510       break;
01511    case AST_FORMAT_G722:
01512    case AST_FORMAT_ADPCM:
01513    case AST_FORMAT_G726:
01514    case AST_FORMAT_G726_AAL2:
01515       len = samples / 2;
01516       break;
01517    default:
01518       ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format));
01519    }
01520 
01521    return len;
01522 }

int ast_codec_get_samples ( struct ast_frame f  ) 

Returns the number of samples contained in the frame.

Definition at line 1441 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().

01442 {
01443    int samples=0;
01444    switch(f->subclass) {
01445    case AST_FORMAT_SPEEX:
01446       samples = speex_samples(f->data.ptr, f->datalen);
01447       break;
01448    case AST_FORMAT_G723_1:
01449       samples = g723_samples(f->data.ptr, f->datalen);
01450       break;
01451    case AST_FORMAT_ILBC:
01452       samples = 240 * (f->datalen / 50);
01453       break;
01454    case AST_FORMAT_GSM:
01455       samples = 160 * (f->datalen / 33);
01456       break;
01457    case AST_FORMAT_G729A:
01458       samples = f->datalen * 8;
01459       break;
01460    case AST_FORMAT_SLINEAR:
01461    case AST_FORMAT_SLINEAR16:
01462       samples = f->datalen / 2;
01463       break;
01464    case AST_FORMAT_LPC10:
01465       /* assumes that the RTP packet contains one LPC10 frame */
01466       samples = 22 * 8;
01467       samples += (((char *)(f->data.ptr))[7] & 0x1) * 8;
01468       break;
01469    case AST_FORMAT_ULAW:
01470    case AST_FORMAT_ALAW:
01471       samples = f->datalen;
01472       break;
01473    case AST_FORMAT_G722:
01474    case AST_FORMAT_ADPCM:
01475    case AST_FORMAT_G726:
01476    case AST_FORMAT_G726_AAL2:
01477       samples = f->datalen * 2;
01478       break;
01479    default:
01480       ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass));
01481    }
01482    return samples;
01483 }

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

Gets duration in ms of interpolation frame for a format.

Definition at line 624 of file frame.h.

References AST_FORMAT_ILBC.

Referenced by __get_from_jb(), and jb_get_and_deliver().

00625 { 
00626    return (format == AST_FORMAT_ILBC) ? 30 : 20;
00627 }

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 1081 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().

01082 {
01083    int x, newindex = 0;
01084 
01085    ast_codec_pref_remove(pref, format);
01086 
01087    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01088       if (AST_FORMAT_LIST[x].bits == format) {
01089          newindex = x + 1;
01090          break;
01091       }
01092    }
01093 
01094    if (newindex) {
01095       for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01096          if (!pref->order[x]) {
01097             pref->order[x] = newindex;
01098             break;
01099          }
01100       }
01101    }
01102 
01103    return x;
01104 }

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 984 of file frame.c.

References ast_codec_pref::order.

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

00985 {
00986    int x, differential = (int) 'A', mem;
00987    char *from, *to;
00988 
00989    if (right) {
00990       from = pref->order;
00991       to = buf;
00992       mem = size;
00993    } else {
00994       to = pref->order;
00995       from = buf;
00996       mem = 32;
00997    }
00998 
00999    memset(to, 0, mem);
01000    for (x = 0; x < 32 ; x++) {
01001       if (!from[x])
01002          break;
01003       to[x] = right ? (from[x] + differential) : (from[x] - differential);
01004    }
01005 }

struct ast_format_list ast_codec_pref_getsize ( struct ast_codec_pref pref,
int  format 
)

Get packet size for codec.

Definition at line 1182 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().

01183 {
01184    int x, idx = -1, framems = 0;
01185    struct ast_format_list fmt = { 0, };
01186 
01187    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01188       if (AST_FORMAT_LIST[x].bits == format) {
01189          fmt = AST_FORMAT_LIST[x];
01190          idx = x;
01191          break;
01192       }
01193    }
01194 
01195    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01196       if (pref->order[x] == (idx + 1)) {
01197          framems = pref->framing[x];
01198          break;
01199       }
01200    }
01201 
01202    /* size validation */
01203    if (!framems)
01204       framems = AST_FORMAT_LIST[idx].def_ms;
01205 
01206    if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */
01207       framems -= framems % AST_FORMAT_LIST[idx].inc_ms;
01208 
01209    if (framems < AST_FORMAT_LIST[idx].min_ms)
01210       framems = AST_FORMAT_LIST[idx].min_ms;
01211 
01212    if (framems > AST_FORMAT_LIST[idx].max_ms)
01213       framems = AST_FORMAT_LIST[idx].max_ms;
01214 
01215    fmt.cur_ms = framems;
01216 
01217    return fmt;
01218 }

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 1042 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().

01043 {
01044    int slot = 0;
01045 
01046    if ((idx >= 0) && (idx < sizeof(pref->order))) {
01047       slot = pref->order[idx];
01048    }
01049 
01050    return slot ? AST_FORMAT_LIST[slot - 1].bits : 0;
01051 }

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 1107 of file frame.c.

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

Referenced by create_addr().

01108 {
01109    int x, newindex = 0;
01110 
01111    /* First step is to get the codecs "index number" */
01112    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01113       if (AST_FORMAT_LIST[x].bits == format) {
01114          newindex = x + 1;
01115          break;
01116       }
01117    }
01118    /* Done if its unknown */
01119    if (!newindex)
01120       return;
01121 
01122    /* Now find any existing occurrence, or the end */
01123    for (x = 0; x < 32; x++) {
01124       if (!pref->order[x] || pref->order[x] == newindex)
01125          break;
01126    }
01127 
01128    if (only_if_existing && !pref->order[x])
01129       return;
01130 
01131    /* Move down to make space to insert - either all the way to the end,
01132       or as far as the existing location (which will be overwritten) */
01133    for (; x > 0; x--) {
01134       pref->order[x] = pref->order[x - 1];
01135       pref->framing[x] = pref->framing[x - 1];
01136    }
01137 
01138    /* And insert the new entry */
01139    pref->order[0] = newindex;
01140    pref->framing[0] = 0; /* ? */
01141 }

void ast_codec_pref_remove ( struct ast_codec_pref pref,
int  format 
)

Remove audio a codec from a preference list.

Definition at line 1054 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().

01055 {
01056    struct ast_codec_pref oldorder;
01057    int x, y = 0;
01058    int slot;
01059    int size;
01060 
01061    if (!pref->order[0])
01062       return;
01063 
01064    memcpy(&oldorder, pref, sizeof(oldorder));
01065    memset(pref, 0, sizeof(*pref));
01066 
01067    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01068       slot = oldorder.order[x];
01069       size = oldorder.framing[x];
01070       if (! slot)
01071          break;
01072       if (AST_FORMAT_LIST[slot-1].bits != format) {
01073          pref->order[y] = slot;
01074          pref->framing[y++] = size;
01075       }
01076    }
01077    
01078 }

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

Set packet size for codec.

Definition at line 1144 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().

01145 {
01146    int x, idx = -1;
01147 
01148    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01149       if (AST_FORMAT_LIST[x].bits == format) {
01150          idx = x;
01151          break;
01152       }
01153    }
01154 
01155    if (idx < 0)
01156       return -1;
01157 
01158    /* size validation */
01159    if (!framems)
01160       framems = AST_FORMAT_LIST[idx].def_ms;
01161 
01162    if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */
01163       framems -= framems % AST_FORMAT_LIST[idx].inc_ms;
01164 
01165    if (framems < AST_FORMAT_LIST[idx].min_ms)
01166       framems = AST_FORMAT_LIST[idx].min_ms;
01167 
01168    if (framems > AST_FORMAT_LIST[idx].max_ms)
01169       framems = AST_FORMAT_LIST[idx].max_ms;
01170 
01171    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01172       if (pref->order[x] == (idx + 1)) {
01173          pref->framing[x] = framems;
01174          break;
01175       }
01176    }
01177 
01178    return x;
01179 }

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 1007 of file frame.c.

References ast_codec_pref_index(), and ast_getformatname().

Referenced by dump_prefs(), and socket_process().

01008 {
01009    int x, codec; 
01010    size_t total_len, slen;
01011    char *formatname;
01012    
01013    memset(buf,0,size);
01014    total_len = size;
01015    buf[0] = '(';
01016    total_len--;
01017    for(x = 0; x < 32 ; x++) {
01018       if (total_len <= 0)
01019          break;
01020       if (!(codec = ast_codec_pref_index(pref,x)))
01021          break;
01022       if ((formatname = ast_getformatname(codec))) {
01023          slen = strlen(formatname);
01024          if (slen > total_len)
01025             break;
01026          strncat(buf, formatname, total_len - 1); /* safe */
01027          total_len -= slen;
01028       }
01029       if (total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) {
01030          strncat(buf, "|", total_len - 1); /* safe */
01031          total_len--;
01032       }
01033    }
01034    if (total_len) {
01035       strncat(buf, ")", total_len - 1); /* safe */
01036       total_len--;
01037    }
01038 
01039    return size - total_len;
01040 }

static force_inline int ast_format_rate ( int  format  )  [static]

Get the sample rate for a given format.

Definition at line 651 of file frame.h.

References AST_FORMAT_G722, and AST_FORMAT_SLINEAR16.

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

00652 {
00653    if (format == AST_FORMAT_G722 || format == AST_FORMAT_SLINEAR16)
00654       return 16000;
00655 
00656    return 8000;
00657 }

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 1524 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().

01525 {
01526    int count;
01527    short *fdata = f->data.ptr;
01528    short adjust_value = abs(adjustment);
01529 
01530    if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR))
01531       return -1;
01532 
01533    if (!adjustment)
01534       return 0;
01535 
01536    for (count = 0; count < f->samples; count++) {
01537       if (adjustment > 0) {
01538          ast_slinear_saturated_multiply(&fdata[count], &adjust_value);
01539       } else if (adjustment < 0) {
01540          ast_slinear_saturated_divide(&fdata[count], &adjust_value);
01541       }
01542    }
01543 
01544    return 0;
01545 }

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

Dump a frame for debugging purposes

Definition at line 733 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, 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, and term_color().

Referenced by __ast_read(), and ast_write().

00734 {
00735    const char noname[] = "unknown";
00736    char ftype[40] = "Unknown Frametype";
00737    char cft[80];
00738    char subclass[40] = "Unknown Subclass";
00739    char csub[80];
00740    char moreinfo[40] = "";
00741    char cn[60];
00742    char cp[40];
00743    char cmn[40];
00744    const char *message = "Unknown";
00745 
00746    if (!name)
00747       name = noname;
00748 
00749 
00750    if (!f) {
00751       ast_verbose("%s [ %s (NULL) ] [%s]\n", 
00752          term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00753          term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 
00754          term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00755       return;
00756    }
00757    /* XXX We should probably print one each of voice and video when the format changes XXX */
00758    if (f->frametype == AST_FRAME_VOICE)
00759       return;
00760    if (f->frametype == AST_FRAME_VIDEO)
00761       return;
00762    switch(f->frametype) {
00763    case AST_FRAME_DTMF_BEGIN:
00764       strcpy(ftype, "DTMF Begin");
00765       subclass[0] = f->subclass;
00766       subclass[1] = '\0';
00767       break;
00768    case AST_FRAME_DTMF_END:
00769       strcpy(ftype, "DTMF End");
00770       subclass[0] = f->subclass;
00771       subclass[1] = '\0';
00772       break;
00773    case AST_FRAME_CONTROL:
00774       strcpy(ftype, "Control");
00775       switch(f->subclass) {
00776       case AST_CONTROL_HANGUP:
00777          strcpy(subclass, "Hangup");
00778          break;
00779       case AST_CONTROL_RING:
00780          strcpy(subclass, "Ring");
00781          break;
00782       case AST_CONTROL_RINGING:
00783          strcpy(subclass, "Ringing");
00784          break;
00785       case AST_CONTROL_ANSWER:
00786          strcpy(subclass, "Answer");
00787          break;
00788       case AST_CONTROL_BUSY:
00789          strcpy(subclass, "Busy");
00790          break;
00791       case AST_CONTROL_TAKEOFFHOOK:
00792          strcpy(subclass, "Take Off Hook");
00793          break;
00794       case AST_CONTROL_OFFHOOK:
00795          strcpy(subclass, "Line Off Hook");
00796          break;
00797       case AST_CONTROL_CONGESTION:
00798          strcpy(subclass, "Congestion");
00799          break;
00800       case AST_CONTROL_FLASH:
00801          strcpy(subclass, "Flash");
00802          break;
00803       case AST_CONTROL_WINK:
00804          strcpy(subclass, "Wink");
00805          break;
00806       case AST_CONTROL_OPTION:
00807          strcpy(subclass, "Option");
00808          break;
00809       case AST_CONTROL_RADIO_KEY:
00810          strcpy(subclass, "Key Radio");
00811          break;
00812       case AST_CONTROL_RADIO_UNKEY:
00813          strcpy(subclass, "Unkey Radio");
00814          break;
00815       case AST_CONTROL_HOLD:
00816          strcpy(subclass, "Hold");
00817          break;
00818       case AST_CONTROL_UNHOLD:
00819          strcpy(subclass, "Unhold");
00820          break;
00821       case AST_CONTROL_T38:
00822          if (f->datalen != sizeof(enum ast_control_t38)) {
00823             message = "Invalid";
00824          } else {
00825             enum ast_control_t38 state = *((enum ast_control_t38 *) f->data.ptr);
00826             if (state == AST_T38_REQUEST_NEGOTIATE)
00827                message = "Negotiation Requested";
00828             else if (state == AST_T38_REQUEST_TERMINATE)
00829                message = "Negotiation Request Terminated";
00830             else if (state == AST_T38_NEGOTIATED)
00831                message = "Negotiated";
00832             else if (state == AST_T38_TERMINATED)
00833                message = "Terminated";
00834             else if (state == AST_T38_REFUSED)
00835                message = "Refused";
00836          }
00837          snprintf(subclass, sizeof(subclass), "T38/%s", message);
00838          break;
00839       case -1:
00840          strcpy(subclass, "Stop generators");
00841          break;
00842       default:
00843          snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass);
00844       }
00845       break;
00846    case AST_FRAME_NULL:
00847       strcpy(ftype, "Null Frame");
00848       strcpy(subclass, "N/A");
00849       break;
00850    case AST_FRAME_IAX:
00851       /* Should never happen */
00852       strcpy(ftype, "IAX Specific");
00853       snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass);
00854       break;
00855    case AST_FRAME_TEXT:
00856       strcpy(ftype, "Text");
00857       strcpy(subclass, "N/A");
00858       ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo));
00859       break;
00860    case AST_FRAME_IMAGE:
00861       strcpy(ftype, "Image");
00862       snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass));
00863       break;
00864    case AST_FRAME_HTML:
00865       strcpy(ftype, "HTML");
00866       switch(f->subclass) {
00867       case AST_HTML_URL:
00868          strcpy(subclass, "URL");
00869          ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo));
00870          break;
00871       case AST_HTML_DATA:
00872          strcpy(subclass, "Data");
00873          break;
00874       case AST_HTML_BEGIN:
00875          strcpy(subclass, "Begin");
00876          break;
00877       case AST_HTML_END:
00878          strcpy(subclass, "End");
00879          break;
00880       case AST_HTML_LDCOMPLETE:
00881          strcpy(subclass, "Load Complete");
00882          break;
00883       case AST_HTML_NOSUPPORT:
00884          strcpy(subclass, "No Support");
00885          break;
00886       case AST_HTML_LINKURL:
00887          strcpy(subclass, "Link URL");
00888          ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo));
00889          break;
00890       case AST_HTML_UNLINK:
00891          strcpy(subclass, "Unlink");
00892          break;
00893       case AST_HTML_LINKREJECT:
00894          strcpy(subclass, "Link Reject");
00895          break;
00896       default:
00897          snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass);
00898          break;
00899       }
00900       break;
00901    case AST_FRAME_MODEM:
00902       strcpy(ftype, "Modem");
00903       switch (f->subclass) {
00904       case AST_MODEM_T38:
00905          strcpy(subclass, "T.38");
00906          break;
00907       case AST_MODEM_V150:
00908          strcpy(subclass, "V.150");
00909          break;
00910       default:
00911          snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass);
00912          break;
00913       }
00914       break;
00915    default:
00916       snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype);
00917    }
00918    if (!ast_strlen_zero(moreinfo))
00919       ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n",  
00920              term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00921              term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)),
00922              f->frametype, 
00923              term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)),
00924              f->subclass, 
00925              term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)),
00926              term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00927    else
00928       ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n",  
00929              term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00930              term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)),
00931              f->frametype, 
00932              term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)),
00933              f->subclass, 
00934              term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00935 }

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

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

void ast_frame_free ( struct ast_frame fr,
int  cache 
)

Requests a frame to be allocated Frees a frame.

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

Definition at line 342 of file frame.c.

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

Referenced by mixmonitor_thread().

00343 {
00344    if (ast_test_flag(fr, AST_FRFLAG_FROM_TRANSLATOR)) {
00345       ast_translate_frame_freed(fr);
00346    } else if (ast_test_flag(fr, AST_FRFLAG_FROM_DSP)) {
00347       ast_dsp_frame_freed(fr);
00348    } else if (ast_test_flag(fr, AST_FRFLAG_FROM_FILESTREAM)) {
00349       ast_filestream_frame_freed(fr);
00350    }
00351 
00352    if (!fr->mallocd)
00353       return;
00354 
00355 #if !defined(LOW_MEMORY)
00356    if (cache && fr->mallocd == AST_MALLOCD_HDR) {
00357       /* Cool, only the header is malloc'd, let's just cache those for now 
00358        * to keep things simple... */
00359       struct ast_frame_cache *frames;
00360 
00361       if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames))) 
00362           && frames->size < FRAME_CACHE_MAX_SIZE) {
00363          AST_LIST_INSERT_HEAD(&frames->list, fr, frame_list);
00364          frames->size++;
00365          return;
00366       }
00367    }
00368 #endif
00369    
00370    if (fr->mallocd & AST_MALLOCD_DATA) {
00371       if (fr->data.ptr) 
00372          ast_free(fr->data.ptr - fr->offset);
00373    }
00374    if (fr->mallocd & AST_MALLOCD_SRC) {
00375       if (fr->src)
00376          ast_free((char *)fr->src);
00377    }
00378    if (fr->mallocd & AST_MALLOCD_HDR) {
00379 #ifdef TRACE_FRAMES
00380       AST_LIST_LOCK(&headerlist);
00381       headers--;
00382       AST_LIST_REMOVE(&headerlist, fr, frame_list);
00383       AST_LIST_UNLOCK(&headerlist);
00384 #endif         
00385       ast_free(fr);
00386    }
00387 }

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 1547 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.

01548 {
01549    int count;
01550    short *data1, *data2;
01551 
01552    if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR))
01553       return -1;
01554 
01555    if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR))
01556       return -1;
01557 
01558    if (f1->samples != f2->samples)
01559       return -1;
01560 
01561    for (count = 0, data1 = f1->data.ptr, data2 = f2->data.ptr;
01562         count < f1->samples;
01563         count++, data1++, data2++)
01564       ast_slinear_saturated_add(data1, data2);
01565 
01566    return 0;
01567 }

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 453 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_jb_put(), ast_rtp_write(), ast_slinfactory_feed(), audiohook_read_frame_single(), autoservice_run(), recordthread(), rpt(), and transmit_audio().

00454 {
00455    struct ast_frame *out = NULL;
00456    int len, srclen = 0;
00457    void *buf = NULL;
00458 
00459 #if !defined(LOW_MEMORY)
00460    struct ast_frame_cache *frames;
00461 #endif
00462 
00463    /* Start with standard stuff */
00464    len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen;
00465    /* If we have a source, add space for it */
00466    /*
00467     * XXX Watch out here - if we receive a src which is not terminated
00468     * properly, we can be easily attacked. Should limit the size we deal with.
00469     */
00470    if (f->src)
00471       srclen = strlen(f->src);
00472    if (srclen > 0)
00473       len += srclen + 1;
00474    
00475 #if !defined(LOW_MEMORY)
00476    if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) {
00477       AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) {
00478          if (out->mallocd_hdr_len >= len) {
00479             size_t mallocd_len = out->mallocd_hdr_len;
00480 
00481             AST_LIST_REMOVE_CURRENT(frame_list);
00482             memset(out, 0, sizeof(*out));
00483             out->mallocd_hdr_len = mallocd_len;
00484             buf = out;
00485             frames->size--;
00486             break;
00487          }
00488       }
00489       AST_LIST_TRAVERSE_SAFE_END;
00490    }
00491 #endif
00492 
00493    if (!buf) {
00494       if (!(buf = ast_calloc_cache(1, len)))
00495          return NULL;
00496       out = buf;
00497       out->mallocd_hdr_len = len;
00498    }
00499 
00500    out->frametype = f->frametype;
00501    out->subclass = f->subclass;
00502    out->datalen = f->datalen;
00503    out->samples = f->samples;
00504    out->delivery = f->delivery;
00505    /* Set us as having malloc'd header only, so it will eventually
00506       get freed. */
00507    out->mallocd = AST_MALLOCD_HDR;
00508    out->offset = AST_FRIENDLY_OFFSET;
00509    if (out->datalen) {
00510       out->data.ptr = buf + sizeof(*out) + AST_FRIENDLY_OFFSET;
00511       memcpy(out->data.ptr, f->data.ptr, out->datalen);  
00512    } else {
00513       out->data.uint32 = f->data.uint32;
00514    }
00515    if (srclen > 0) {
00516       /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */
00517       char *src;
00518       out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen;
00519       src = (char *) out->src;
00520       /* Must have space since we allocated for it */
00521       strcpy(src, f->src);
00522    }
00523    ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO);
00524    out->ts = f->ts;
00525    out->len = f->len;
00526    out->seqno = f->seqno;
00527    return out;
00528 }

struct ast_frame* ast_frisolate ( struct ast_frame fr  ) 

Makes a frame independent of any static storage.

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

Definition at line 394 of file frame.c.

References ast_clear_flag, ast_copy_flags, ast_frame_header_new(), ast_free, AST_FRFLAG_FROM_DSP, AST_FRFLAG_FROM_TRANSLATOR, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, ast_malloc, AST_MALLOCD_DATA, AST_MALLOCD_HDR, AST_MALLOCD_SRC, ast_strdup, ast_test_flag, ast_frame::data, ast_frame::datalen, ast_frame::frametype, 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(), and jpeg_read_image().

00395 {
00396    struct ast_frame *out;
00397    void *newdata;
00398 
00399    ast_clear_flag(fr, AST_FRFLAG_FROM_TRANSLATOR);
00400    ast_clear_flag(fr, AST_FRFLAG_FROM_DSP);
00401 
00402    if (!(fr->mallocd & AST_MALLOCD_HDR)) {
00403       /* Allocate a new header if needed */
00404       if (!(out = ast_frame_header_new()))
00405          return NULL;
00406       out->frametype = fr->frametype;
00407       out->subclass = fr->subclass;
00408       out->datalen = fr->datalen;
00409       out->samples = fr->samples;
00410       out->offset = fr->offset;
00411       out->data = fr->data;
00412       /* Copy the timing data */
00413       ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO);
00414       if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) {
00415          out->ts = fr->ts;
00416          out->len = fr->len;
00417          out->seqno = fr->seqno;
00418       }
00419    } else
00420       out = fr;
00421    
00422    if (!(fr->mallocd & AST_MALLOCD_SRC)) {
00423       if (fr->src) {
00424          if (!(out->src = ast_strdup(fr->src))) {
00425             if (out != fr)
00426                ast_free(out);
00427             return NULL;
00428          }
00429       }
00430    } else
00431       out->src = fr->src;
00432    
00433    if (!(fr->mallocd & AST_MALLOCD_DATA))  {
00434       if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) {
00435          if (out->src != fr->src)
00436             ast_free((void *) out->src);
00437          if (out != fr)
00438             ast_free(out);
00439          return NULL;
00440       }
00441       newdata += AST_FRIENDLY_OFFSET;
00442       out->offset = AST_FRIENDLY_OFFSET;
00443       out->datalen = fr->datalen;
00444       memcpy(newdata, fr->data.ptr, fr->datalen);
00445       out->data.ptr = newdata;
00446    }
00447 
00448    out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA;
00449    
00450    return out;
00451 }

struct ast_format_list* ast_get_format_list ( size_t *  size  ) 

Definition at line 546 of file frame.c.

References ARRAY_LEN, and AST_FORMAT_LIST.

00547 {
00548    *size = ARRAY_LEN(AST_FORMAT_LIST);
00549    return AST_FORMAT_LIST;
00550 }

struct ast_format_list* ast_get_format_list_index ( int  index  ) 

Definition at line 541 of file frame.c.

References AST_FORMAT_LIST.

00542 {
00543    return &AST_FORMAT_LIST[idx];
00544 }

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 613 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().

00614 {
00615    int x, all, format = 0;
00616 
00617    all = strcasecmp(name, "all") ? 0 : 1;
00618    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00619       if (all || 
00620            !strcasecmp(AST_FORMAT_LIST[x].name,name) ||
00621            !strcasecmp(AST_FORMAT_LIST[x].name, ast_expand_codec_alias(name))) {
00622          format |= AST_FORMAT_LIST[x].bits;
00623          if (!all)
00624             break;
00625       }
00626    }
00627 
00628    return format;
00629 }

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 552 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_peer_capabilities(), show_codecs(), sip_request_call(), sip_rtp_read(), socket_process(), start_rtp(), unistim_request(), unistim_rtp_read(), and unistim_write().

00553 {
00554    int x;
00555    char *ret = "unknown";
00556    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00557       if (AST_FORMAT_LIST[x].bits == format) {
00558          ret = AST_FORMAT_LIST[x].name;
00559          break;
00560       }
00561    }
00562    return ret;
00563 }

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 565 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().

00566 {
00567    int x;
00568    unsigned len;
00569    char *start, *end = buf;
00570 
00571    if (!size)
00572       return buf;
00573    snprintf(end, size, "0x%x (", format);
00574    len = strlen(end);
00575    end += len;
00576    size -= len;
00577    start = end;
00578    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00579       if (AST_FORMAT_LIST[x].bits & format) {
00580          snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name);
00581          len = strlen(end);
00582          end += len;
00583          size -= len;
00584       }
00585    }
00586    if (start == end)
00587       ast_copy_string(start, "nothing)", size);
00588    else if (size > 1)
00589       *(end -1) = ')';
00590    return buf;
00591 }

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 1243 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().

01244 {
01245    int errors = 0;
01246    char *parse = NULL, *this = NULL, *psize = NULL;
01247    int format = 0, framems = 0;
01248 
01249    parse = ast_strdupa(list);
01250    while ((this = strsep(&parse, ","))) {
01251       framems = 0;
01252       if ((psize = strrchr(this, ':'))) {
01253          *psize++ = '\0';
01254          ast_debug(1, "Packetization for codec: %s is %s\n", this, psize);
01255          framems = atoi(psize);
01256          if (framems < 0) {
01257             framems = 0;
01258             errors++;
01259             ast_log(LOG_WARNING, "Bad packetization value for codec %s\n", this);
01260          }
01261       }
01262       if (!(format = ast_getformatbyname(this))) {
01263          ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this);
01264          errors++;
01265          continue;
01266       }
01267 
01268       if (mask) {
01269          if (allowing)
01270             *mask |= format;
01271          else
01272             *mask &= ~format;
01273       }
01274 
01275       /* Set up a preference list for audio. Do not include video in preferences 
01276          since we can not transcode video and have to use whatever is offered
01277        */
01278       if (pref && (format & AST_FORMAT_AUDIO_MASK)) {
01279          if (strcasecmp(this, "all")) {
01280             if (allowing) {
01281                ast_codec_pref_append(pref, format);
01282                ast_codec_pref_setsize(pref, format, framems);
01283             }
01284             else
01285                ast_codec_pref_remove(pref, format);
01286          } else if (!allowing) {
01287             memset(pref, 0, sizeof(*pref));
01288          }
01289       }
01290    }
01291    return errors;
01292 }

void ast_smoother_free ( struct ast_smoother s  ) 

Definition at line 289 of file frame.c.

References ast_free, and s.

Referenced by ast_rtp_destroy(), and ast_rtp_write().

00290 {
00291    ast_free(s);
00292 }

int ast_smoother_get_flags ( struct ast_smoother smoother  ) 

Definition at line 189 of file frame.c.

References s.

00190 {
00191    return s->flags;
00192 }

struct ast_smoother* ast_smoother_new ( int  bytes  ) 

Definition at line 179 of file frame.c.

References ast_malloc, ast_smoother_reset(), and s.

Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().

00180 {
00181    struct ast_smoother *s;
00182    if (size < 1)
00183       return NULL;
00184    if ((s = ast_malloc(sizeof(*s))))
00185       ast_smoother_reset(s, size);
00186    return s;
00187 }

struct ast_frame* ast_smoother_read ( struct ast_smoother s  ) 

Definition at line 239 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().

00240 {
00241    struct ast_frame *opt;
00242    int len;
00243 
00244    /* IF we have an optimization frame, send it */
00245    if (s->opt) {
00246       if (s->opt->offset < AST_FRIENDLY_OFFSET)
00247          ast_log(LOG_WARNING, "Returning a frame of inappropriate offset (%d).\n",
00248                      s->opt->offset);
00249       opt = s->opt;
00250       s->opt = NULL;
00251       return opt;
00252    }
00253 
00254    /* Make sure we have enough data */
00255    if (s->len < s->size) {
00256       /* Or, if this is a G.729 frame with VAD on it, send it immediately anyway */
00257       if (!((s->flags & AST_SMOOTHER_FLAG_G729) && (s->size % 10)))
00258          return NULL;
00259    }
00260    len = s->size;
00261    if (len > s->len)
00262       len = s->len;
00263    /* Make frame */
00264    s->f.frametype = AST_FRAME_VOICE;
00265    s->f.subclass = s->format;
00266    s->f.data.ptr = s->framedata + AST_FRIENDLY_OFFSET;
00267    s->f.offset = AST_FRIENDLY_OFFSET;
00268    s->f.datalen = len;
00269    /* Samples will be improper given VAD, but with VAD the concept really doesn't even exist */
00270    s->f.samples = len * s->samplesperbyte;   /* XXX rounding */
00271    s->f.delivery = s->delivery;
00272    /* Fill Data */
00273    memcpy(s->f.data.ptr, s->data, len);
00274    s->len -= len;
00275    /* Move remaining data to the front if applicable */
00276    if (s->len) {
00277       /* In principle this should all be fine because if we are sending
00278          G.729 VAD, the next timestamp will take over anyawy */
00279       memmove(s->data, s->data + len, s->len);
00280       if (!ast_tvzero(s->delivery)) {
00281          /* If we have delivery time, increment it, otherwise, leave it at 0 */
00282          s->delivery = ast_tvadd(s->delivery, ast_samp2tv(s->f.samples, ast_format_rate(s->format)));
00283       }
00284    }
00285    /* Return frame */
00286    return &s->f;
00287 }

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 157 of file frame.c.

References s, and smoother_frame_feed().

Referenced by ast_rtp_codec_setpref().

00158 {
00159    /* if there is no change, then nothing to do */
00160    if (s->size == bytes) {
00161       return;
00162    }
00163    /* set the new desired output size */
00164    s->size = bytes;
00165    /* if there is no 'optimized' frame in the smoother,
00166     *   then there is nothing left to do
00167     */
00168    if (!s->opt) {
00169       return;
00170    }
00171    /* there is an 'optimized' frame here at the old size,
00172     * but it must now be put into the buffer so the data
00173     * can be extracted at the new size
00174     */
00175    smoother_frame_feed(s, s->opt, s->opt_needs_swap);
00176    s->opt = NULL;
00177 }

void ast_smoother_reset ( struct ast_smoother s,
int  bytes 
)

Definition at line 151 of file frame.c.

References s.

Referenced by ast_smoother_new().

00152 {
00153    memset(s, 0, sizeof(*s));
00154    s->size = bytes;
00155 }

void ast_smoother_set_flags ( struct ast_smoother smoother,
int  flags 
)

Definition at line 194 of file frame.c.

References s.

Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().

00195 {
00196    s->flags = flags;
00197 }

int ast_smoother_test_flag ( struct ast_smoother s,
int  flag 
)

Definition at line 199 of file frame.c.

References s.

Referenced by ast_rtp_write().

00200 {
00201    return (s->flags & flag);
00202 }

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

Definition at line 530 of file frame.c.

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

00531 {
00532    int i;
00533    unsigned short *dst_s = dst;
00534    const unsigned short *src_s = src;
00535 
00536    for (i = 0; i < samples; i++)
00537       dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8);
00538 }


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 127 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(), 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_rfc2833(), process_sdp(), send_dtmf(), sip_read(), sip_rtp_read(), skinny_rtp_read(), unistim_rtp_read(), and wakeup_sub().


Generated on Fri Jul 24 00:41:44 2009 for Asterisk - the Open Source PBX by  doxygen 1.4.7