Thu Jul 9 13:41:19 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_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 << 25)
#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, __f->data, __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 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 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_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_digitdetect(), 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(), dahdi_read(), dahdi_translate(), dahdi_write(), dictate_exec(), do_waiting(), eagi_exec(), extenspy_exec(), fax_generator_generate(), find_transcoders(), function_ilink(), 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_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 << 25)

T.140 Text format - ITU T.140, RFC 4351

Definition at line 292 of file frame.h.

Referenced by ast_write().

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

Definition at line 293 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_openvstream(), ast_request(), ast_rtp_read(), ast_translate_available_formats(), check_peer_ok(), check_user_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 467 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, __f->data, __f->samples); } while(0)

Definition at line 466 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_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 = (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 434 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_parking_thread(), do_waiting(), echo_exec(), eivr_comm(), find_cache(), gen_generate(), handle_cli_file_convert(), 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(), 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 __frame_free(), ast_frisolate(), and create_video_frame().

#define AST_MALLOCD_HDR   (1 << 0)

Need the header be free'd?

Definition at line 211 of file frame.h.

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

#define AST_MALLOCD_SRC   (1 << 2)

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

Definition at line 215 of file frame.h.

Referenced by __frame_free(), and ast_frisolate().

#define AST_MIN_OFFSET   32

Definition at line 208 of file frame.h.

Referenced by __ast_smoother_feed().

#define AST_MODEM_T38   1

T.38 Fax-over-IP

Definition at line 219 of file frame.h.

Referenced by ast_frame_dump(), 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 348 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 370 of file frame.h.

Referenced by dahdi_setoption().

#define AST_OPTION_FLAG_ACCEPT   1

Definition at line 331 of file frame.h.

#define AST_OPTION_FLAG_ANSWER   5

Definition at line 334 of file frame.h.

#define AST_OPTION_FLAG_QUERY   4

Definition at line 333 of file frame.h.

#define AST_OPTION_FLAG_REJECT   2

Definition at line 332 of file frame.h.

#define AST_OPTION_FLAG_REQUEST   0

Definition at line 330 of file frame.h.

Referenced by ast_bridge_call(), and iax2_setoption().

#define AST_OPTION_FLAG_WTF   6

Definition at line 335 of file frame.h.

#define AST_OPTION_OPRMODE   7

Definition at line 367 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 345 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 364 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 376 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 342 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 339 of file frame.h.

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

#define AST_OPTION_TXGAIN   5

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

Definition at line 356 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 537 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 542 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 543 of file frame.h.

#define AST_SMOOTHER_FLAG_BE   (1 << 1)

Definition at line 327 of file frame.h.

Referenced by ast_rtp_write().

#define AST_SMOOTHER_FLAG_G729   (1 << 0)

Definition at line 326 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 295 of file frame.h.

00295                             {
00296    AST_CONTROL_HANGUP = 1,    /*!< Other end has hungup */
00297    AST_CONTROL_RING = 2,      /*!< Local ring */
00298    AST_CONTROL_RINGING = 3,   /*!< Remote end is ringing */
00299    AST_CONTROL_ANSWER = 4,    /*!< Remote end has answered */
00300    AST_CONTROL_BUSY = 5,      /*!< Remote end is busy */
00301    AST_CONTROL_TAKEOFFHOOK = 6,  /*!< Make it go off hook */
00302    AST_CONTROL_OFFHOOK = 7,   /*!< Line is off hook */
00303    AST_CONTROL_CONGESTION = 8,   /*!< Congestion (circuits busy) */
00304    AST_CONTROL_FLASH = 9,     /*!< Flash hook */
00305    AST_CONTROL_WINK = 10,     /*!< Wink */
00306    AST_CONTROL_OPTION = 11,   /*!< Set a low-level option */
00307    AST_CONTROL_RADIO_KEY = 12,   /*!< Key Radio */
00308    AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */
00309    AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */
00310    AST_CONTROL_PROCEEDING = 15,  /*!< Indicate CALL PROCEEDING */
00311    AST_CONTROL_HOLD = 16,     /*!< Indicate call is placed on hold */
00312    AST_CONTROL_UNHOLD = 17,   /*!< Indicate call is left from hold */
00313    AST_CONTROL_VIDUPDATE = 18,   /*!< Indicate video frame update */
00314    AST_CONTROL_T38 = 19,      /*!< T38 state change request/notification */
00315    AST_CONTROL_SRCUPDATE = 20,     /*!< Indicate source of media has changed */
00316 };

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 318 of file frame.h.

00318                      {
00319    AST_T38_REQUEST_NEGOTIATE = 1,   /*!< Request T38 on a channel (voice to fax) */
00320    AST_T38_REQUEST_TERMINATE, /*!< Terminate T38 on a channel (fax to voice) */
00321    AST_T38_NEGOTIATED,     /*!< T38 negotiated (fax mode) */
00322    AST_T38_TERMINATED,     /*!< T38 terminated (back to voice) */
00323    AST_T38_REFUSED         /*!< T38 refused for some reason (usually rejected by remote end) */
00324 };

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 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, f->data, 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 646 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().

00647 {
00648    int x;
00649    char *ret = "unknown";
00650    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00651       if (AST_FORMAT_LIST[x].bits == codec) {
00652          ret = AST_FORMAT_LIST[x].desc;
00653          break;
00654       }
00655    }
00656    return ret;
00657 }

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 1204 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().

01205 {
01206    int x, ret = 0, slot;
01207 
01208    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01209       slot = pref->order[x];
01210 
01211       if (!slot)
01212          break;
01213       if (formats & AST_FORMAT_LIST[slot-1].bits) {
01214          ret = AST_FORMAT_LIST[slot-1].bits;
01215          break;
01216       }
01217    }
01218    if (ret & AST_FORMAT_AUDIO_MASK)
01219       return ret;
01220 
01221    ast_debug(4, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec");
01222 
01223       return find_best ? ast_best_codec(formats) : 0;
01224 }

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 1468 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().

01469 {
01470    int len = 0;
01471 
01472    /* XXX Still need speex, g723, and lpc10 XXX */ 
01473    switch(format) {
01474    case AST_FORMAT_G723_1:
01475       len = (samples / 240) * 20;
01476       break;
01477    case AST_FORMAT_ILBC:
01478       len = (samples / 240) * 50;
01479       break;
01480    case AST_FORMAT_GSM:
01481       len = (samples / 160) * 33;
01482       break;
01483    case AST_FORMAT_G729A:
01484       len = samples / 8;
01485       break;
01486    case AST_FORMAT_SLINEAR:
01487    case AST_FORMAT_SLINEAR16:
01488       len = samples * 2;
01489       break;
01490    case AST_FORMAT_ULAW:
01491    case AST_FORMAT_ALAW:
01492       len = samples;
01493       break;
01494    case AST_FORMAT_G722:
01495    case AST_FORMAT_ADPCM:
01496    case AST_FORMAT_G726:
01497    case AST_FORMAT_G726_AAL2:
01498       len = samples / 2;
01499       break;
01500    default:
01501       ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format));
01502    }
01503 
01504    return len;
01505 }

int ast_codec_get_samples ( struct ast_frame f  ) 

Returns the number of samples contained in the frame.

Definition at line 1424 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().

01425 {
01426    int samples=0;
01427    switch(f->subclass) {
01428    case AST_FORMAT_SPEEX:
01429       samples = speex_samples(f->data, f->datalen);
01430       break;
01431    case AST_FORMAT_G723_1:
01432       samples = g723_samples(f->data, f->datalen);
01433       break;
01434    case AST_FORMAT_ILBC:
01435       samples = 240 * (f->datalen / 50);
01436       break;
01437    case AST_FORMAT_GSM:
01438       samples = 160 * (f->datalen / 33);
01439       break;
01440    case AST_FORMAT_G729A:
01441       samples = f->datalen * 8;
01442       break;
01443    case AST_FORMAT_SLINEAR:
01444    case AST_FORMAT_SLINEAR16:
01445       samples = f->datalen / 2;
01446       break;
01447    case AST_FORMAT_LPC10:
01448       /* assumes that the RTP packet contains one LPC10 frame */
01449       samples = 22 * 8;
01450       samples += (((char *)(f->data))[7] & 0x1) * 8;
01451       break;
01452    case AST_FORMAT_ULAW:
01453    case AST_FORMAT_ALAW:
01454       samples = f->datalen;
01455       break;
01456    case AST_FORMAT_G722:
01457    case AST_FORMAT_ADPCM:
01458    case AST_FORMAT_G726:
01459    case AST_FORMAT_G726_AAL2:
01460       samples = f->datalen * 2;
01461       break;
01462    default:
01463       ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass));
01464    }
01465    return samples;
01466 }

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

Gets duration in ms of interpolation frame for a format.

Definition at line 625 of file frame.h.

References AST_FORMAT_ILBC.

Referenced by __get_from_jb(), and jb_get_and_deliver().

00626 { 
00627    return (format == AST_FORMAT_ILBC) ? 30 : 20;
00628 }

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 1063 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().

01064 {
01065    int x, newindex = 0;
01066 
01067    ast_codec_pref_remove(pref, format);
01068 
01069    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01070       if (AST_FORMAT_LIST[x].bits == format) {
01071          newindex = x + 1;
01072          break;
01073       }
01074    }
01075 
01076    if (newindex) {
01077       for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01078          if (!pref->order[x]) {
01079             pref->order[x] = newindex;
01080             break;
01081          }
01082       }
01083    }
01084 
01085    return x;
01086 }

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 965 of file frame.c.

References ast_codec_pref::order.

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

00966 {
00967    int x, differential = (int) 'A', mem;
00968    char *from, *to;
00969 
00970    if (right) {
00971       from = pref->order;
00972       to = buf;
00973       mem = size;
00974    } else {
00975       to = pref->order;
00976       from = buf;
00977       mem = 32;
00978    }
00979 
00980    memset(to, 0, mem);
00981    for (x = 0; x < 32 ; x++) {
00982       if (!from[x])
00983          break;
00984       to[x] = right ? (from[x] + differential) : (from[x] - differential);
00985    }
00986 }

struct ast_format_list ast_codec_pref_getsize ( struct ast_codec_pref pref,
int  format 
)

Get packet size for codec.

Definition at line 1165 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().

01166 {
01167    int x, index = -1, framems = 0;
01168    struct ast_format_list fmt = { 0, };
01169 
01170    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01171       if (AST_FORMAT_LIST[x].bits == format) {
01172          fmt = AST_FORMAT_LIST[x];
01173          index = x;
01174          break;
01175       }
01176    }
01177 
01178    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01179       if (pref->order[x] == (index + 1)) {
01180          framems = pref->framing[x];
01181          break;
01182       }
01183    }
01184 
01185    /* size validation */
01186    if (!framems)
01187       framems = AST_FORMAT_LIST[index].def_ms;
01188 
01189    if (AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */
01190       framems -= framems % AST_FORMAT_LIST[index].inc_ms;
01191 
01192    if (framems < AST_FORMAT_LIST[index].min_ms)
01193       framems = AST_FORMAT_LIST[index].min_ms;
01194 
01195    if (framems > AST_FORMAT_LIST[index].max_ms)
01196       framems = AST_FORMAT_LIST[index].max_ms;
01197 
01198    fmt.cur_ms = framems;
01199 
01200    return fmt;
01201 }

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 1023 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().

01024 {
01025    int slot = 0;
01026 
01027    
01028    if ((index >= 0) && (index < sizeof(pref->order))) {
01029       slot = pref->order[index];
01030    }
01031 
01032    return slot ? AST_FORMAT_LIST[slot-1].bits : 0;
01033 }

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 1089 of file frame.c.

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

Referenced by create_addr().

01090 {
01091    int x, newindex = 0;
01092 
01093    /* First step is to get the codecs "index number" */
01094    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01095       if (AST_FORMAT_LIST[x].bits == format) {
01096          newindex = x + 1;
01097          break;
01098       }
01099    }
01100    /* Done if its unknown */
01101    if (!newindex)
01102       return;
01103 
01104    /* Now find any existing occurrence, or the end */
01105    for (x = 0; x < 32; x++) {
01106       if (!pref->order[x] || pref->order[x] == newindex)
01107          break;
01108    }
01109 
01110    if (only_if_existing && !pref->order[x])
01111       return;
01112 
01113    /* Move down to make space to insert - either all the way to the end,
01114       or as far as the existing location (which will be overwritten) */
01115    for (; x > 0; x--) {
01116       pref->order[x] = pref->order[x - 1];
01117       pref->framing[x] = pref->framing[x - 1];
01118    }
01119 
01120    /* And insert the new entry */
01121    pref->order[0] = newindex;
01122    pref->framing[0] = 0; /* ? */
01123 }

void ast_codec_pref_remove ( struct ast_codec_pref pref,
int  format 
)

Remove audio a codec from a preference list.

Definition at line 1036 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().

01037 {
01038    struct ast_codec_pref oldorder;
01039    int x, y = 0;
01040    int slot;
01041    int size;
01042 
01043    if (!pref->order[0])
01044       return;
01045 
01046    memcpy(&oldorder, pref, sizeof(oldorder));
01047    memset(pref, 0, sizeof(*pref));
01048 
01049    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01050       slot = oldorder.order[x];
01051       size = oldorder.framing[x];
01052       if (! slot)
01053          break;
01054       if (AST_FORMAT_LIST[slot-1].bits != format) {
01055          pref->order[y] = slot;
01056          pref->framing[y++] = size;
01057       }
01058    }
01059    
01060 }

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

Set packet size for codec.

Definition at line 1126 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().

01127 {
01128    int x, index = -1;
01129 
01130    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01131       if (AST_FORMAT_LIST[x].bits == format) {
01132          index = x;
01133          break;
01134       }
01135    }
01136 
01137    if (index < 0)
01138       return -1;
01139 
01140    /* size validation */
01141    if (!framems)
01142       framems = AST_FORMAT_LIST[index].def_ms;
01143 
01144    if (AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */
01145       framems -= framems % AST_FORMAT_LIST[index].inc_ms;
01146 
01147    if (framems < AST_FORMAT_LIST[index].min_ms)
01148       framems = AST_FORMAT_LIST[index].min_ms;
01149 
01150    if (framems > AST_FORMAT_LIST[index].max_ms)
01151       framems = AST_FORMAT_LIST[index].max_ms;
01152 
01153 
01154    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01155       if (pref->order[x] == (index + 1)) {
01156          pref->framing[x] = framems;
01157          break;
01158       }
01159    }
01160 
01161    return x;
01162 }

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 988 of file frame.c.

References ast_codec_pref_index(), and ast_getformatname().

Referenced by dump_prefs(), and socket_process().

00989 {
00990    int x, codec; 
00991    size_t total_len, slen;
00992    char *formatname;
00993    
00994    memset(buf,0,size);
00995    total_len = size;
00996    buf[0] = '(';
00997    total_len--;
00998    for(x = 0; x < 32 ; x++) {
00999       if (total_len <= 0)
01000          break;
01001       if (!(codec = ast_codec_pref_index(pref,x)))
01002          break;
01003       if ((formatname = ast_getformatname(codec))) {
01004          slen = strlen(formatname);
01005          if (slen > total_len)
01006             break;
01007          strncat(buf, formatname, total_len - 1); /* safe */
01008          total_len -= slen;
01009       }
01010       if (total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) {
01011          strncat(buf, "|", total_len - 1); /* safe */
01012          total_len--;
01013       }
01014    }
01015    if (total_len) {
01016       strncat(buf, ")", total_len - 1); /* safe */
01017       total_len--;
01018    }
01019 
01020    return size - total_len;
01021 }

static force_inline int ast_format_rate ( int  format  )  [static]

Get the sample rate for a given format.

Definition at line 652 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().

00653 {
00654    if (format == AST_FORMAT_G722 || format == AST_FORMAT_SLINEAR16)
00655       return 16000;
00656 
00657    return 8000;
00658 }

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 1507 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(), conf_run(), and volume_callback().

01508 {
01509    int count;
01510    short *fdata = f->data;
01511    short adjust_value = abs(adjustment);
01512 
01513    if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR))
01514       return -1;
01515 
01516    if (!adjustment)
01517       return 0;
01518 
01519    for (count = 0; count < f->samples; count++) {
01520       if (adjustment > 0) {
01521          ast_slinear_saturated_multiply(&fdata[count], &adjust_value);
01522       } else if (adjustment < 0) {
01523          ast_slinear_saturated_divide(&fdata[count], &adjust_value);
01524       }
01525    }
01526 
01527    return 0;
01528 }

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

Dump a frame for debugging purposes

Definition at line 748 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().

00749 {
00750    const char noname[] = "unknown";
00751    char ftype[40] = "Unknown Frametype";
00752    char cft[80];
00753    char subclass[40] = "Unknown Subclass";
00754    char csub[80];
00755    char moreinfo[40] = "";
00756    char cn[60];
00757    char cp[40];
00758    char cmn[40];
00759    const char *message = "Unknown";
00760 
00761    if (!name)
00762       name = noname;
00763 
00764 
00765    if (!f) {
00766       ast_verbose("%s [ %s (NULL) ] [%s]\n", 
00767          term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00768          term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 
00769          term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00770       return;
00771    }
00772    /* XXX We should probably print one each of voice and video when the format changes XXX */
00773    if (f->frametype == AST_FRAME_VOICE)
00774       return;
00775    if (f->frametype == AST_FRAME_VIDEO)
00776       return;
00777    switch(f->frametype) {
00778    case AST_FRAME_DTMF_BEGIN:
00779       strcpy(ftype, "DTMF Begin");
00780       subclass[0] = f->subclass;
00781       subclass[1] = '\0';
00782       break;
00783    case AST_FRAME_DTMF_END:
00784       strcpy(ftype, "DTMF End");
00785       subclass[0] = f->subclass;
00786       subclass[1] = '\0';
00787       break;
00788    case AST_FRAME_CONTROL:
00789       strcpy(ftype, "Control");
00790       switch(f->subclass) {
00791       case AST_CONTROL_HANGUP:
00792          strcpy(subclass, "Hangup");
00793          break;
00794       case AST_CONTROL_RING:
00795          strcpy(subclass, "Ring");
00796          break;
00797       case AST_CONTROL_RINGING:
00798          strcpy(subclass, "Ringing");
00799          break;
00800       case AST_CONTROL_ANSWER:
00801          strcpy(subclass, "Answer");
00802          break;
00803       case AST_CONTROL_BUSY:
00804          strcpy(subclass, "Busy");
00805          break;
00806       case AST_CONTROL_TAKEOFFHOOK:
00807          strcpy(subclass, "Take Off Hook");
00808          break;
00809       case AST_CONTROL_OFFHOOK:
00810          strcpy(subclass, "Line Off Hook");
00811          break;
00812       case AST_CONTROL_CONGESTION:
00813          strcpy(subclass, "Congestion");
00814          break;
00815       case AST_CONTROL_FLASH:
00816          strcpy(subclass, "Flash");
00817          break;
00818       case AST_CONTROL_WINK:
00819          strcpy(subclass, "Wink");
00820          break;
00821       case AST_CONTROL_OPTION:
00822          strcpy(subclass, "Option");
00823          break;
00824       case AST_CONTROL_RADIO_KEY:
00825          strcpy(subclass, "Key Radio");
00826          break;
00827       case AST_CONTROL_RADIO_UNKEY:
00828          strcpy(subclass, "Unkey Radio");
00829          break;
00830       case AST_CONTROL_HOLD:
00831          strcpy(subclass, "Hold");
00832          break;
00833       case AST_CONTROL_UNHOLD:
00834          strcpy(subclass, "Unhold");
00835          break;
00836       case AST_CONTROL_T38:
00837          if (f->datalen != sizeof(enum ast_control_t38)) {
00838             message = "Invalid";
00839          } else {
00840             enum ast_control_t38 state = *((enum ast_control_t38 *) f->data);
00841             if (state == AST_T38_REQUEST_NEGOTIATE)
00842                message = "Negotiation Requested";
00843             else if (state == AST_T38_REQUEST_TERMINATE)
00844                message = "Negotiation Request Terminated";
00845             else if (state == AST_T38_NEGOTIATED)
00846                message = "Negotiated";
00847             else if (state == AST_T38_TERMINATED)
00848                message = "Terminated";
00849             else if (state == AST_T38_REFUSED)
00850                message = "Refused";
00851          }
00852          snprintf(subclass, sizeof(subclass), "T38/%s", message);
00853          break;
00854       case -1:
00855          strcpy(subclass, "Stop generators");
00856          break;
00857       default:
00858          snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass);
00859       }
00860       break;
00861    case AST_FRAME_NULL:
00862       strcpy(ftype, "Null Frame");
00863       strcpy(subclass, "N/A");
00864       break;
00865    case AST_FRAME_IAX:
00866       /* Should never happen */
00867       strcpy(ftype, "IAX Specific");
00868       snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass);
00869       break;
00870    case AST_FRAME_TEXT:
00871       strcpy(ftype, "Text");
00872       strcpy(subclass, "N/A");
00873       ast_copy_string(moreinfo, f->data, sizeof(moreinfo));
00874       break;
00875    case AST_FRAME_IMAGE:
00876       strcpy(ftype, "Image");
00877       snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass));
00878       break;
00879    case AST_FRAME_HTML:
00880       strcpy(ftype, "HTML");
00881       switch(f->subclass) {
00882       case AST_HTML_URL:
00883          strcpy(subclass, "URL");
00884          ast_copy_string(moreinfo, f->data, sizeof(moreinfo));
00885          break;
00886       case AST_HTML_DATA:
00887          strcpy(subclass, "Data");
00888          break;
00889       case AST_HTML_BEGIN:
00890          strcpy(subclass, "Begin");
00891          break;
00892       case AST_HTML_END:
00893          strcpy(subclass, "End");
00894          break;
00895       case AST_HTML_LDCOMPLETE:
00896          strcpy(subclass, "Load Complete");
00897          break;
00898       case AST_HTML_NOSUPPORT:
00899          strcpy(subclass, "No Support");
00900          break;
00901       case AST_HTML_LINKURL:
00902          strcpy(subclass, "Link URL");
00903          ast_copy_string(moreinfo, f->data, sizeof(moreinfo));
00904          break;
00905       case AST_HTML_UNLINK:
00906          strcpy(subclass, "Unlink");
00907          break;
00908       case AST_HTML_LINKREJECT:
00909          strcpy(subclass, "Link Reject");
00910          break;
00911       default:
00912          snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass);
00913          break;
00914       }
00915       break;
00916    case AST_FRAME_MODEM:
00917       strcpy(ftype, "Modem");
00918       switch (f->subclass) {
00919       case AST_MODEM_T38:
00920          strcpy(subclass, "T.38");
00921          break;
00922       case AST_MODEM_V150:
00923          strcpy(subclass, "V.150");
00924          break;
00925       default:
00926          snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass);
00927          break;
00928       }
00929       break;
00930    default:
00931       snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype);
00932    }
00933    if (!ast_strlen_zero(moreinfo))
00934       ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n",  
00935              term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00936              term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)),
00937              f->frametype, 
00938              term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)),
00939              f->subclass, 
00940              term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)),
00941              term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00942    else
00943       ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n",  
00944              term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00945              term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)),
00946              f->frametype, 
00947              term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)),
00948              f->subclass, 
00949              term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00950 }

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 373 of file frame.c.

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

Referenced by mixmonitor_thread().

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

int ast_frame_slinear_sum ( struct ast_frame f1,
struct ast_frame f2 
)

Sums two frames of audio samples.

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

Definition at line 1530 of file frame.c.

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

01531 {
01532    int count;
01533    short *data1, *data2;
01534 
01535    if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR))
01536       return -1;
01537 
01538    if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR))
01539       return -1;
01540 
01541    if (f1->samples != f2->samples)
01542       return -1;
01543 
01544    for (count = 0, data1 = f1->data, data2 = f2->data;
01545         count < f1->samples;
01546         count++, data1++, data2++)
01547       ast_slinear_saturated_add(data1, data2);
01548 
01549    return 0;
01550 }

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 470 of file frame.c.

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

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

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

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 389 of file frame.c.

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

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

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

struct ast_format_list* ast_get_format_list ( size_t *  size  ) 

Definition at line 561 of file frame.c.

References ARRAY_LEN, and AST_FORMAT_LIST.

00562 {
00563    *size = ARRAY_LEN(AST_FORMAT_LIST);
00564    return AST_FORMAT_LIST;
00565 }

struct ast_format_list* ast_get_format_list_index ( int  index  ) 

Definition at line 556 of file frame.c.

References AST_FORMAT_LIST.

00557 {
00558    return &AST_FORMAT_LIST[index];
00559 }

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 628 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().

00629 {
00630    int x, all, format = 0;
00631 
00632    all = strcasecmp(name, "all") ? 0 : 1;
00633    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00634       if (all || 
00635            !strcasecmp(AST_FORMAT_LIST[x].name,name) ||
00636            !strcasecmp(AST_FORMAT_LIST[x].name,ast_expand_codec_alias(name))) {
00637          format |= AST_FORMAT_LIST[x].bits;
00638          if (!all)
00639             break;
00640       }
00641    }
00642 
00643    return format;
00644 }

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 567 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().

00568 {
00569    int x;
00570    char *ret = "unknown";
00571    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00572       if (AST_FORMAT_LIST[x].bits == format) {
00573          ret = AST_FORMAT_LIST[x].name;
00574          break;
00575       }
00576    }
00577    return ret;
00578 }

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 580 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().

00581 {
00582    int x;
00583    unsigned len;
00584    char *start, *end = buf;
00585 
00586    if (!size)
00587       return buf;
00588    snprintf(end, size, "0x%x (", format);
00589    len = strlen(end);
00590    end += len;
00591    size -= len;
00592    start = end;
00593    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00594       if (AST_FORMAT_LIST[x].bits & format) {
00595          snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name);
00596          len = strlen(end);
00597          end += len;
00598          size -= len;
00599       }
00600    }
00601    if (start == end)
00602       ast_copy_string(start, "nothing)", size);
00603    else if (size > 1)
00604       *(end -1) = ')';
00605    return buf;
00606 }

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 1226 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().

01227 {
01228    int errors = 0;
01229    char *parse = NULL, *this = NULL, *psize = NULL;
01230    int format = 0, framems = 0;
01231 
01232    parse = ast_strdupa(list);
01233    while ((this = strsep(&parse, ","))) {
01234       framems = 0;
01235       if ((psize = strrchr(this, ':'))) {
01236          *psize++ = '\0';
01237          ast_debug(1, "Packetization for codec: %s is %s\n", this, psize);
01238          framems = atoi(psize);
01239          if (framems < 0) {
01240             framems = 0;
01241             errors++;
01242             ast_log(LOG_WARNING, "Bad packetization value for codec %s\n", this);
01243          }
01244       }
01245       if (!(format = ast_getformatbyname(this))) {
01246          ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this);
01247          errors++;
01248          continue;
01249       }
01250 
01251       if (mask) {
01252          if (allowing)
01253             *mask |= format;
01254          else
01255             *mask &= ~format;
01256       }
01257 
01258       /* Set up a preference list for audio. Do not include video in preferences 
01259          since we can not transcode video and have to use whatever is offered
01260        */
01261       if (pref && (format & AST_FORMAT_AUDIO_MASK)) {
01262          if (strcasecmp(this, "all")) {
01263             if (allowing) {
01264                ast_codec_pref_append(pref, format);
01265                ast_codec_pref_setsize(pref, format, framems);
01266             }
01267             else
01268                ast_codec_pref_remove(pref, format);
01269          } else if (!allowing) {
01270             memset(pref, 0, sizeof(*pref));
01271          }
01272       }
01273    }
01274    return errors;
01275 }

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 = 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, 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 545 of file frame.c.

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

00546 {
00547    int i;
00548    unsigned short *dst_s = dst;
00549    const unsigned short *src_s = src;
00550 
00551    for (i = 0; i < samples; i++)
00552       dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8);
00553 }


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(), ast_do_masquerade(), ast_rtcp_read(), ast_rtp_read(), ast_softhangup_nolock(), ast_udptl_read(), conf_run(), console_read(), features_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 Thu Jul 9 13:41:19 2009 for Asterisk - the Open Source PBX by  doxygen 1.4.7