Mon Jun 27 16:51:14 2011

Asterisk developer's documentation


frame.h File Reference

Asterisk internal frame definitions. More...

#include <sys/time.h>
#include "asterisk/frame_defs.h"
#include "asterisk/endian.h"
#include "asterisk/linkedlists.h"

Go to the source code of this file.

Data Structures

struct  ast_codec_pref
struct  ast_control_read_action_payload
struct  ast_control_t38_parameters
struct  ast_format_list
 Definition of supported media formats (codecs). More...
struct  ast_frame
 Data structure associated with a single frame of data. More...
union  ast_frame_subclass
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   (1ULL << 5)
#define AST_FORMAT_ALAW   (1ULL << 3)
#define AST_FORMAT_AUDIO_MASK   0xFFFF0000FFFFULL
#define AST_FORMAT_G719   (1ULL << 32)
#define AST_FORMAT_G722   (1ULL << 12)
#define AST_FORMAT_G723_1   (1ULL << 0)
#define AST_FORMAT_G726   (1ULL << 11)
#define AST_FORMAT_G726_AAL2   (1ULL << 4)
#define AST_FORMAT_G729A   (1ULL << 8)
#define AST_FORMAT_GSM   (1ULL << 1)
#define AST_FORMAT_H261   (1ULL << 18)
#define AST_FORMAT_H263   (1ULL << 19)
#define AST_FORMAT_H263_PLUS   (1ULL << 20)
#define AST_FORMAT_H264   (1ULL << 21)
#define AST_FORMAT_ILBC   (1ULL << 10)
#define AST_FORMAT_JPEG   (1ULL << 16)
#define AST_FORMAT_LPC10   (1ULL << 7)
#define AST_FORMAT_MAX_TEXT   (1ULL << 28)
#define AST_FORMAT_MP4_VIDEO   (1ULL << 22)
#define AST_FORMAT_PNG   (1ULL << 17)
#define AST_FORMAT_RESERVED   (1ULL << 63)
#define AST_FORMAT_SIREN14   (1ULL << 14)
#define AST_FORMAT_SIREN7   (1ULL << 13)
#define AST_FORMAT_SLINEAR   (1ULL << 6)
#define AST_FORMAT_SLINEAR16   (1ULL << 15)
#define AST_FORMAT_SPEEX   (1ULL << 9)
#define AST_FORMAT_SPEEX16   (1ULL << 33)
#define AST_FORMAT_T140   (1ULL << 27)
#define AST_FORMAT_T140RED   (1ULL << 26)
#define AST_FORMAT_TESTLAW   (1ULL << 47)
#define AST_FORMAT_TEXT_MASK   (((1ULL << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK))
#define AST_FORMAT_ULAW   (1ULL << 2)
#define AST_FORMAT_VIDEO_MASK   ((((1ULL << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) | 0x7FFF000000000000ULL)
#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_CC_AGENT_TYPE   17
#define AST_OPTION_CHANNEL_WRITE   9
 Handle channel write data If a channel needs to process the data from a func_channel write operation after func_channel_write executes, it can define the setoption callback and process this option. A pointer to an ast_chan_write_info_t will be passed.
#define AST_OPTION_DEVICE_NAME   16
#define AST_OPTION_DIGIT_DETECT   14
#define AST_OPTION_ECHOCAN   8
#define AST_OPTION_FAX_DETECT   15
#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_FORMAT_READ   11
#define AST_OPTION_FORMAT_WRITE   12
#define AST_OPTION_MAKE_COMPATIBLE   13
#define AST_OPTION_OPRMODE   7
#define AST_OPTION_RELAXDTMF   3
#define AST_OPTION_RXGAIN   6
#define AST_OPTION_SECURE_MEDIA   19
#define AST_OPTION_SECURE_SIGNALING   18
#define AST_OPTION_T38_STATE   10
#define AST_OPTION_TDD   2
#define AST_OPTION_TONE_VERIFY   1
#define AST_OPTION_TXGAIN   5
#define AST_SMOOTHER_FLAG_BE   (1 << 1)
#define AST_SMOOTHER_FLAG_G729   (1 << 0)

Enumerations

enum  { AST_FRFLAG_HAS_TIMING_INFO = (1 << 0) }
enum  ast_control_frame_type {
  AST_CONTROL_HANGUP = 1, AST_CONTROL_RING = 2, AST_CONTROL_RINGING = 3, AST_CONTROL_ANSWER = 4,
  AST_CONTROL_BUSY = 5, AST_CONTROL_TAKEOFFHOOK = 6, AST_CONTROL_OFFHOOK = 7, AST_CONTROL_CONGESTION = 8,
  AST_CONTROL_FLASH = 9, AST_CONTROL_WINK = 10, AST_CONTROL_OPTION = 11, AST_CONTROL_RADIO_KEY = 12,
  AST_CONTROL_RADIO_UNKEY = 13, AST_CONTROL_PROGRESS = 14, AST_CONTROL_PROCEEDING = 15, AST_CONTROL_HOLD = 16,
  AST_CONTROL_UNHOLD = 17, AST_CONTROL_VIDUPDATE = 18, _XXX_AST_CONTROL_T38 = 19, AST_CONTROL_SRCUPDATE = 20,
  AST_CONTROL_TRANSFER = 21, AST_CONTROL_CONNECTED_LINE = 22, AST_CONTROL_REDIRECTING = 23, AST_CONTROL_T38_PARAMETERS = 24,
  AST_CONTROL_CC = 25, AST_CONTROL_SRCCHANGE = 26, AST_CONTROL_READ_ACTION = 27, AST_CONTROL_AOC = 28,
  AST_CONTROL_END_OF_Q = 29
}
enum  ast_control_t38 {
  AST_T38_REQUEST_NEGOTIATE = 1, AST_T38_REQUEST_TERMINATE, AST_T38_NEGOTIATED, AST_T38_TERMINATED,
  AST_T38_REFUSED, AST_T38_REQUEST_PARMS
}
enum  ast_control_t38_rate {
  AST_T38_RATE_2400 = 0, AST_T38_RATE_4800, AST_T38_RATE_7200, AST_T38_RATE_9600,
  AST_T38_RATE_12000, AST_T38_RATE_14400
}
enum  ast_control_t38_rate_management { AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF = 0, AST_T38_RATE_MANAGEMENT_LOCAL_TCF }
enum  ast_control_transfer { AST_TRANSFER_SUCCESS = 0, AST_TRANSFER_FAILED }
enum  ast_frame_read_action { AST_FRAME_READ_ACTION_CONNECTED_LINE_MACRO }
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 (format_t codec)
 Get a name from a format Gets a name from a format.
format_t ast_codec_choose (struct ast_codec_pref *pref, format_t 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 (format_t 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 (format_t format)
 Gets duration in ms of interpolation frame for a format.
int ast_codec_pref_append (struct ast_codec_pref *pref, format_t 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, format_t format)
 Get packet size for codec.
format_t 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, format_t 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, format_t format)
 Remove audio a codec from a preference list.
int ast_codec_pref_setsize (struct ast_codec_pref *pref, format_t 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 (format_t 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.
int ast_frame_clear (struct ast_frame *frame)
 Clear all audio samples from an ast_frame. The frame must be AST_FRAME_VOICE and AST_FORMAT_SLINEAR.
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)
format_t ast_getformatbyname (const char *name)
 Gets a format from a name.
char * ast_getformatname (format_t format)
 Get the name of a format.
char * ast_getformatname_multiple (char *buf, size_t size, format_t format)
 Get the names of a set of formats.
int ast_parse_allow_disallow (struct ast_codec_pref *pref, format_t *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   (1ULL << 5)

ADPCM (IMA)

Definition at line 252 of file frame.h.

Referenced by adpcm_sample(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), convertcap(), vox_read(), and vox_write().

#define AST_FORMAT_ALAW   (1ULL << 3)

Raw A-law data (G.711)

Definition at line 248 of file frame.h.

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

#define AST_FORMAT_AUDIO_MASK   0xFFFF0000FFFFULL

Maximum audio mask

Definition at line 274 of file frame.h.

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

#define AST_FORMAT_G719   (1ULL << 32)

G.719 (64 kbps assumed)

Definition at line 298 of file frame.h.

Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_write(), g719read(), g719write(), and process_sdp_a_audio().

#define AST_FORMAT_G722   (1ULL << 12)

G.722

Definition at line 266 of file frame.h.

Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_raw_write(), au_seek(), convertcap(), g722_sample(), pcm_read(), and rtp_get_rate().

#define AST_FORMAT_G723_1   (1ULL << 0)

G.723.1 compression

Definition at line 242 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(), and start_rtp().

#define AST_FORMAT_G726   (1ULL << 11)

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

Definition at line 264 of file frame.h.

Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_codecs_payloads_set_rtpmap_type_rate(), g726_read(), g726_sample(), and g726_write().

#define AST_FORMAT_G726_AAL2   (1ULL << 4)

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

Definition at line 250 of file frame.h.

Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_codecs_payloads_set_rtpmap_type_rate(), ast_rtp_lookup_mime_subtype2(), codec_ast2skinny(), codec_skinny2ast(), and setup_rtp_connection().

#define AST_FORMAT_G729A   (1ULL << 8)

G.729A audio

Definition at line 258 of file frame.h.

Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), 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   (1ULL << 1)

GSM compression

Definition at line 244 of file frame.h.

Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), convertcap(), gsm_read(), gsm_sample(), gsm_write(), wav_read(), and wav_write().

#define AST_FORMAT_H261   (1ULL << 18)

H.261 Video

Definition at line 280 of file frame.h.

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

#define AST_FORMAT_H263   (1ULL << 19)

H.263 Video

Definition at line 282 of file frame.h.

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

#define AST_FORMAT_H263_PLUS   (1ULL << 20)

H.263+ Video

Definition at line 284 of file frame.h.

Referenced by h263p_encap().

#define AST_FORMAT_H264   (1ULL << 21)

H.264 Video

Definition at line 286 of file frame.h.

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

#define AST_FORMAT_ILBC   (1ULL << 10)

iLBC Free Compression

Definition at line 262 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_sample(), and ilbc_write().

#define AST_FORMAT_JPEG   (1ULL << 16)

JPEG Images

Definition at line 276 of file frame.h.

Referenced by jpeg_read_image(), and jpeg_write_image().

#define AST_FORMAT_LPC10   (1ULL << 7)

LPC10, 180 samples/frame

Definition at line 256 of file frame.h.

Referenced by ast_best_codec(), ast_codec_get_samples(), and lpc10_sample().

#define AST_FORMAT_MAX_TEXT   (1ULL << 28)

Maximum text mask

Definition at line 295 of file frame.h.

#define AST_FORMAT_MP4_VIDEO   (1ULL << 22)

MPEG4 Video

Definition at line 288 of file frame.h.

Referenced by mpeg4_encap().

#define AST_FORMAT_PNG   (1ULL << 17)

PNG Images

Definition at line 278 of file frame.h.

Referenced by phone_read().

#define AST_FORMAT_RESERVED   (1ULL << 63)

Reserved bit - do not use

Definition at line 304 of file frame.h.

#define AST_FORMAT_SIREN14   (1ULL << 14)

G.722.1 Annex C (also known as Siren14, 48kbps assumed)

Definition at line 270 of file frame.h.

Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_write(), process_sdp_a_audio(), siren14read(), and siren14write().

#define AST_FORMAT_SIREN7   (1ULL << 13)

G.722.1 (also known as Siren7, 32kbps assumed)

Definition at line 268 of file frame.h.

Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_write(), process_sdp_a_audio(), siren7read(), and siren7write().

#define AST_FORMAT_SLINEAR   (1ULL << 6)

Raw 16-bit Signed Linear (8000 Hz) PCM

Definition at line 254 of file frame.h.

Referenced by __ast_play_and_record(), _moh_class_malloc(), action_originate(), agent_new(), alsa_new(), alsa_read(), alsa_request(), ast_audiohook_read_frame(), ast_best_codec(), ast_channel_make_compatible_helper(), ast_channel_start_silence_generator(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_call_progress(), ast_dsp_noise(), ast_dsp_process(), ast_dsp_silence(), ast_frame_adjust_volume(), ast_frame_slinear_sum(), ast_rtp_read(), ast_slinfactory_init(), ast_slinfactory_init_rate(), ast_speech_new(), ast_write(), attempt_reconnect(), audio_audiohook_write_list(), audiohook_read_frame_both(), audiohook_read_frame_single(), background_detect_exec(), bridge_request(), build_conf(), chanspy_exec(), conf_run(), connect_link(), dahdi_read(), dahdi_translate(), dahdi_write(), dahdiscan_exec(), dictate_exec(), do_notify(), do_waiting(), eagi_exec(), extenspy_exec(), fax_generator_generate(), find_transcoders(), generic_fax_exec(), generic_recall(), handle_jack_audio(), handle_recordfile(), handle_speechcreate(), handle_speechrecognize(), iax_frame_wrap(), ices_exec(), init_outgoing(), is_encoder(), isAnsweringMachine(), jack_exec(), jack_hook_callback(), linear_alloc(), linear_generator(), load_module(), load_moh_classes(), local_ast_moh_start(), measurenoise(), mixmonitor_thread(), mp3_exec(), nbs_request(), nbs_xwrite(), NBScat_exec(), ogg_vorbis_read(), ogg_vorbis_write(), oh323_rtp_read(), orig_app(), orig_exten(), originate_exec(), oss_new(), oss_read(), oss_request(), parkandannounce_exec(), phone_new(), phone_read(), phone_request(), phone_setup(), phone_write(), pitchshift_cb(), play_sound_file(), playtones_alloc(), rpt(), rpt_call(), rpt_exec(), rpt_tele_thread(), send_waveform_to_channel(), silence_generator_generate(), slin8_sample(), slinear_read(), slinear_write(), socket_process(), softmix_bridge_join(), softmix_bridge_write(), spandsp_fax_read(), speech_background(), spy_generate(), tonepair_alloc(), transmit_audio(), usbradio_new(), usbradio_read(), usbradio_request(), wav_read(), and wav_write().

#define AST_FORMAT_SLINEAR16   (1ULL << 15)

Raw 16-bit Signed Linear (16000 Hz) PCM

Definition at line 272 of file frame.h.

Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_read(), ast_slinfactory_init_rate(), console_new(), pitchshift_cb(), slin16_sample(), slinear_read(), slinear_write(), softmix_bridge_join(), softmix_bridge_write(), stream_monitor(), wav_open(), wav_read(), wav_rewrite(), and wav_write().

#define AST_FORMAT_SPEEX   (1ULL << 9)

SpeeX Free Compression

Definition at line 260 of file frame.h.

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

#define AST_FORMAT_SPEEX16   (1ULL << 33)

SpeeX Wideband (16kHz) Free Compression

Definition at line 300 of file frame.h.

Referenced by ast_best_codec(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_write(), and speex16_sample().

#define AST_FORMAT_T140   (1ULL << 27)

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

Definition at line 293 of file frame.h.

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

#define AST_FORMAT_T140RED   (1ULL << 26)

T.140 RED Text format RFC 4103

Definition at line 291 of file frame.h.

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

#define AST_FORMAT_TESTLAW   (1ULL << 47)

Raw mu-law data (G.711)

Definition at line 302 of file frame.h.

Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), and ast_dsp_process().

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

Definition at line 296 of file frame.h.

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

#define AST_FORMAT_ULAW   (1ULL << 2)

Raw mu-law data (G.711)

Definition at line 246 of file frame.h.

Referenced by __adsi_transmit_messages(), adsi_careful_send(), alarmreceiver_exec(), ast_adsi_transmit_message_full(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), 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(), and ulaw_sample().

#define AST_FORMAT_VIDEO_MASK   ((((1ULL << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) | 0x7FFF000000000000ULL)

Definition at line 289 of file frame.h.

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

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

Definition at line 583 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 582 of file frame.h.

Referenced by phone_read().

#define AST_FRAME_DTMF   AST_FRAME_DTMF_END

Definition at line 128 of file frame.h.

Referenced by __adsi_transmit_messages(), __analog_ss_thread(), __ast_play_and_record(), action_atxfer(), action_dahdidialoffhook(), agent_ack_sleep(), analog_ss_thread(), ast_audiohook_write_list(), ast_bridge_call(), ast_jb_put(), background_detect_exec(), cb_events(), channel_spy(), cli_console_dial(), conf_run(), console_dial(), dahdi_bridge(), dictate_exec(), disa_exec(), do_immediate_setup(), echo_exec(), eivr_comm(), feature_request_and_dial(), 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(), 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 183 of file frame.h.

Referenced by fax_generator_generate(), g719read(), g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), ilbc_read(), ogg_vorbis_read(), pcm_read(), siren14read(), siren7read(), slinear_read(), spandsp_fax_read(), t38_tx_packet_handler(), vox_read(), and wav_read().

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

Definition at line 550 of file frame.h.

Referenced by __adsi_transmit_messages(), __analog_ss_thread(), __ast_answer(), __ast_play_and_record(), __ast_queue_frame(), __ast_read(), __ast_request_and_dial(), adsi_careful_send(), agent_ack_sleep(), agent_read(), analog_ss_thread(), ast_audiohook_read_frame(), ast_autoservice_stop(), ast_bridge_call(), ast_bridge_handle_trip(), ast_channel_clear_softhangup(), ast_channel_destructor(), ast_framehook_attach(), ast_indicate_data(), ast_jb_destroy(), ast_jb_put(), ast_queue_cc_frame(), 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_transfer(), ast_translate(), ast_udptl_bridge(), ast_waitfordigit_full(), ast_write(), ast_writestream(), async_wait(), audio_audiohook_write_list(), autoservice_run(), background_detect_exec(), bridge_handle_dtmf(), calc_cost(), channel_spy(), check_goto_on_transfer(), conf_flush(), conf_free(), conf_run(), create_jb(), dahdi_bridge(), dial_exec_full(), dictate_exec(), disa_exec(), disable_t38(), do_idle_thread(), do_waiting(), echo_exec(), eivr_comm(), feature_request_and_dial(), find_cache(), framehook_detach_and_destroy(), gen_generate(), generic_fax_exec(), handle_cli_file_convert(), handle_recordfile(), handle_speechrecognize(), iax_park_thread(), ices_exec(), isAnsweringMachine(), jack_exec(), jb_empty_and_reset_adaptive(), jb_empty_and_reset_fixed(), jb_get_and_deliver(), launch_asyncagi(), local_bridge_loop(), manage_parkinglot(), masq_park_call(), measurenoise(), moh_files_generator(), monitor_dial(), mp3_exec(), multicast_rtp_write(), NBScat_exec(), read_frame(), receive_dtmf_digits(), receivefax_t38_init(), recordthread(), remote_bridge_loop(), rpt(), run_agi(), send_tone_burst(), send_waveform_to_channel(), sendfax_t38_init(), sendurl_exec(), session_destroy(), speech_background(), spy_generate(), transmit_audio(), transmit_t38(), 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 204 of file frame.h.

Referenced by __get_from_jb(), adjust_frame_for_plc(), 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(), g719read(), 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_cn_rfc3389(), send_tone_burst(), send_waveform_to_channel(), siren14read(), siren7read(), slinear_read(), sms_generate(), spandsp_fax_read(), usbradio_read(), vox_read(), and wav_read().

#define AST_HTML_BEGIN   4

Beginning frame

Definition at line 226 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_DATA   2

Data frame

Definition at line 224 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_END   8

End frame

Definition at line 228 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_LDCOMPLETE   16

Load is complete

Definition at line 230 of file frame.h.

Referenced by ast_frame_dump(), and sendurl_exec().

#define AST_HTML_LINKREJECT   20

Reject link request

Definition at line 238 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_LINKURL   18

Send URL, and track

Definition at line 234 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_NOSUPPORT   17

Peer is unable to support HTML

Definition at line 232 of file frame.h.

Referenced by ast_frame_dump(), and sendurl_exec().

#define AST_HTML_UNLINK   19

No more HTML linkage

Definition at line 236 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_URL   1

Sending a URL

Definition at line 222 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 210 of file frame.h.

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

#define AST_MALLOCD_HDR   (1 << 0)

Need the header be free'd?

Definition at line 208 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 212 of file frame.h.

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

#define AST_MIN_OFFSET   32

Definition at line 205 of file frame.h.

Referenced by __ast_smoother_feed().

#define AST_MODEM_T38   1

T.38 Fax-over-IP

Definition at line 216 of file frame.h.

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

#define AST_MODEM_V150   2

V.150 Modem-over-IP

Definition at line 218 of file frame.h.

Referenced by ast_frame_dump().

#define AST_OPTION_AUDIO_MODE   4

Set (or clear) Audio (Not-Clear) Mode Option data is a single signed char value 0 or 1

Definition at line 420 of file frame.h.

Referenced by ast_bridge_call(), dahdi_hangup(), dahdi_setoption(), and iax2_setoption().

#define AST_OPTION_CC_AGENT_TYPE   17

Get the CC agent type from the channel (Read only) Option data is a character buffer of suitable length

Definition at line 487 of file frame.h.

Referenced by ast_channel_get_cc_agent_type(), and dahdi_queryoption().

#define AST_OPTION_CHANNEL_WRITE   9

Handle channel write data If a channel needs to process the data from a func_channel write operation after func_channel_write executes, it can define the setoption callback and process this option. A pointer to an ast_chan_write_info_t will be passed.

Note:
This option should never be passed over the network.

Definition at line 451 of file frame.h.

Referenced by func_channel_write(), and local_setoption().

#define AST_OPTION_DEVICE_NAME   16

Get the device name from the channel (Read only) Option data is a character buffer of suitable length

Definition at line 483 of file frame.h.

Referenced by ast_channel_get_device_name(), and sip_queryoption().

#define AST_OPTION_DIGIT_DETECT   14

Get or set the digit detection state of the channel Option data is a single signed char value 0 or 1

Definition at line 475 of file frame.h.

Referenced by ast_bridge_call(), dahdi_queryoption(), dahdi_setoption(), iax2_setoption(), rcvfax_exec(), sip_queryoption(), sip_setoption(), and sndfax_exec().

#define AST_OPTION_ECHOCAN   8

Explicitly enable or disable echo cancelation for the given channel Option data is a single signed char value 0 or 1

Note:
This option appears to be unused in the code. It is handled, but never set or queried.

Definition at line 443 of file frame.h.

Referenced by dahdi_setoption().

#define AST_OPTION_FAX_DETECT   15

Get or set the fax tone detection state of the channel Option data is a single signed char value 0 or 1

Definition at line 479 of file frame.h.

Referenced by ast_bridge_call(), dahdi_queryoption(), dahdi_setoption(), iax2_setoption(), rcvfax_exec(), and sndfax_exec().

#define AST_OPTION_FLAG_ACCEPT   1

Definition at line 399 of file frame.h.

#define AST_OPTION_FLAG_ANSWER   5

Definition at line 402 of file frame.h.

#define AST_OPTION_FLAG_QUERY   4

Definition at line 401 of file frame.h.

#define AST_OPTION_FLAG_REJECT   2

Definition at line 400 of file frame.h.

#define AST_OPTION_FLAG_REQUEST   0

Definition at line 398 of file frame.h.

Referenced by ast_bridge_call(), and iax2_setoption().

#define AST_OPTION_FLAG_WTF   6

Definition at line 403 of file frame.h.

#define AST_OPTION_FORMAT_READ   11

Request that the channel driver deliver frames in a specific format Option data is a format_t

Definition at line 461 of file frame.h.

Referenced by set_format(), and sip_setoption().

#define AST_OPTION_FORMAT_WRITE   12

Request that the channel driver be prepared to accept frames in a specific format Option data is a format_t

Definition at line 465 of file frame.h.

Referenced by set_format(), and sip_setoption().

#define AST_OPTION_MAKE_COMPATIBLE   13

Request that the channel driver make two channels of the same tech type compatible if possible Option data is an ast_channel

Note:
This option should never be passed over the network

Definition at line 471 of file frame.h.

Referenced by ast_channel_make_compatible_helper(), and sip_setoption().

#define AST_OPTION_OPRMODE   7

Definition at line 436 of file frame.h.

Referenced by dahdi_setoption(), dial_exec_full(), and iax2_setoption().

#define AST_OPTION_RELAXDTMF   3

Relax the parameters for DTMF reception (mainly for radio use) Option data is a single signed char value 0 or 1

Definition at line 416 of file frame.h.

Referenced by ast_bridge_call(), dahdi_setoption(), iax2_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 430 of file frame.h.

Referenced by dahdi_setoption(), func_channel_write_real(), iax2_setoption(), play_record_review(), reset_volumes(), set_talk_volume(), and vm_forwardoptions().

#define AST_OPTION_SECURE_MEDIA   19

Definition at line 492 of file frame.h.

Referenced by iax2_queryoption(), iax2_setoption(), set_security_requirements(), sip_queryoption(), and sip_setoption().

#define AST_OPTION_SECURE_SIGNALING   18

Get or set the security options on a channel Option data is an integer value of 0 or 1

Definition at line 491 of file frame.h.

Referenced by iax2_queryoption(), iax2_setoption(), set_security_requirements(), sip_queryoption(), and sip_setoption().

#define AST_OPTION_T38_STATE   10

Definition at line 457 of file frame.h.

Referenced by ast_channel_get_t38_state(), local_queryoption(), and sip_queryoption().

#define AST_OPTION_TDD   2

Put a compatible channel into TDD (TTY for the hearing-impared) mode Option data is a single signed char value 0 or 1

Definition at line 412 of file frame.h.

Referenced by analog_hangup(), ast_bridge_call(), dahdi_hangup(), dahdi_setoption(), handle_tddmode(), and iax2_setoption().

#define AST_OPTION_TONE_VERIFY   1

Verify touchtones by muting audio transmission (and reception) and verify the tone is still present Option data is a single signed char value 0 or 1

Definition at line 408 of file frame.h.

Referenced by analog_hangup(), ast_bridge_call(), conf_run(), dahdi_hangup(), dahdi_setoption(), iax2_setoption(), rpt(), and rpt_exec().

#define AST_OPTION_TXGAIN   5

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

Definition at line 425 of file frame.h.

Referenced by common_exec(), dahdi_setoption(), func_channel_write_real(), iax2_setoption(), reset_volumes(), and set_listen_volume().

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

Definition at line 653 of file frame.h.

Referenced by ast_rtp_write(), and generic_fax_exec().

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

Definition at line 658 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 659 of file frame.h.

#define AST_SMOOTHER_FLAG_BE   (1 << 1)

Definition at line 395 of file frame.h.

Referenced by ast_rtp_write().

#define AST_SMOOTHER_FLAG_G729   (1 << 0)

Definition at line 394 of file frame.h.

Referenced by __ast_smoother_feed(), ast_smoother_read(), and smoother_frame_feed().


Enumeration Type Documentation

anonymous enum

Enumerator:
AST_FRFLAG_HAS_TIMING_INFO  This frame contains valid timing information

Definition at line 130 of file frame.h.

00130      {
00131    /*! This frame contains valid timing information */
00132    AST_FRFLAG_HAS_TIMING_INFO = (1 << 0),
00133 };

enum ast_control_frame_type

Enumerator:
AST_CONTROL_HANGUP  Other end has hungup
AST_CONTROL_RING  Local ring
AST_CONTROL_RINGING  Remote end is ringing
AST_CONTROL_ANSWER  Remote end has answered
AST_CONTROL_BUSY  Remote end is busy
AST_CONTROL_TAKEOFFHOOK  Make it go off hook
AST_CONTROL_OFFHOOK  Line is off hook
AST_CONTROL_CONGESTION  Congestion (circuits busy)
AST_CONTROL_FLASH  Flash hook
AST_CONTROL_WINK  Wink
AST_CONTROL_OPTION  Set a low-level option
AST_CONTROL_RADIO_KEY  Key Radio
AST_CONTROL_RADIO_UNKEY  Un-Key Radio
AST_CONTROL_PROGRESS  Indicate PROGRESS
AST_CONTROL_PROCEEDING  Indicate CALL PROCEEDING
AST_CONTROL_HOLD  Indicate call is placed on hold
AST_CONTROL_UNHOLD  Indicate call is left from hold
AST_CONTROL_VIDUPDATE  Indicate video frame update
_XXX_AST_CONTROL_T38  T38 state change request/notification
Deprecated:
This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead.
AST_CONTROL_SRCUPDATE  Indicate source of media has changed
AST_CONTROL_TRANSFER  Indicate status of a transfer request
AST_CONTROL_CONNECTED_LINE  Indicate connected line has changed
AST_CONTROL_REDIRECTING  Indicate redirecting id has changed
AST_CONTROL_T38_PARAMETERS 
AST_CONTROL_CC  T38 state change request/notification with parameters Indication that Call completion service is possible
AST_CONTROL_SRCCHANGE  Media source has changed and requires a new RTP SSRC
AST_CONTROL_READ_ACTION  Tell ast_read to take a specific action
AST_CONTROL_AOC  Advice of Charge with encoded generic AOC payload
AST_CONTROL_END_OF_Q  Indicate that this position was the end of the channel queue for a softhangup.

Definition at line 306 of file frame.h.

00306                             {
00307    AST_CONTROL_HANGUP = 1,    /*!< Other end has hungup */
00308    AST_CONTROL_RING = 2,      /*!< Local ring */
00309    AST_CONTROL_RINGING = 3,   /*!< Remote end is ringing */
00310    AST_CONTROL_ANSWER = 4,    /*!< Remote end has answered */
00311    AST_CONTROL_BUSY = 5,      /*!< Remote end is busy */
00312    AST_CONTROL_TAKEOFFHOOK = 6,  /*!< Make it go off hook */
00313    AST_CONTROL_OFFHOOK = 7,   /*!< Line is off hook */
00314    AST_CONTROL_CONGESTION = 8,   /*!< Congestion (circuits busy) */
00315    AST_CONTROL_FLASH = 9,     /*!< Flash hook */
00316    AST_CONTROL_WINK = 10,     /*!< Wink */
00317    AST_CONTROL_OPTION = 11,   /*!< Set a low-level option */
00318    AST_CONTROL_RADIO_KEY = 12,   /*!< Key Radio */
00319    AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */
00320    AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */
00321    AST_CONTROL_PROCEEDING = 15,  /*!< Indicate CALL PROCEEDING */
00322    AST_CONTROL_HOLD = 16,     /*!< Indicate call is placed on hold */
00323    AST_CONTROL_UNHOLD = 17,   /*!< Indicate call is left from hold */
00324    AST_CONTROL_VIDUPDATE = 18,   /*!< Indicate video frame update */
00325    _XXX_AST_CONTROL_T38 = 19, /*!< T38 state change request/notification \deprecated This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead. */
00326    AST_CONTROL_SRCUPDATE = 20,     /*!< Indicate source of media has changed */
00327    AST_CONTROL_TRANSFER = 21,      /*!< Indicate status of a transfer request */
00328    AST_CONTROL_CONNECTED_LINE = 22,/*!< Indicate connected line has changed */
00329    AST_CONTROL_REDIRECTING = 23,    /*!< Indicate redirecting id has changed */
00330    AST_CONTROL_T38_PARAMETERS = 24, /*! T38 state change request/notification with parameters */
00331    AST_CONTROL_CC = 25, /*!< Indication that Call completion service is possible */
00332    AST_CONTROL_SRCCHANGE = 26,  /*!< Media source has changed and requires a new RTP SSRC */
00333    AST_CONTROL_READ_ACTION = 27, /*!< Tell ast_read to take a specific action */
00334    AST_CONTROL_AOC = 28,           /*!< Advice of Charge with encoded generic AOC payload */
00335    AST_CONTROL_END_OF_Q = 29,    /*!< Indicate that this position was the end of the channel queue for a softhangup. */
00336 };

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)
AST_T38_REQUEST_PARMS  request far end T.38 parameters for a channel in 'negotiating' state

Definition at line 355 of file frame.h.

00355                      {
00356    AST_T38_REQUEST_NEGOTIATE = 1,   /*!< Request T38 on a channel (voice to fax) */
00357    AST_T38_REQUEST_TERMINATE, /*!< Terminate T38 on a channel (fax to voice) */
00358    AST_T38_NEGOTIATED,     /*!< T38 negotiated (fax mode) */
00359    AST_T38_TERMINATED,     /*!< T38 terminated (back to voice) */
00360    AST_T38_REFUSED,     /*!< T38 refused for some reason (usually rejected by remote end) */
00361    AST_T38_REQUEST_PARMS,     /*!< request far end T.38 parameters for a channel in 'negotiating' state */
00362 };

enum ast_control_t38_rate

Enumerator:
AST_T38_RATE_2400 
AST_T38_RATE_4800 
AST_T38_RATE_7200 
AST_T38_RATE_9600 
AST_T38_RATE_12000 
AST_T38_RATE_14400 

Definition at line 364 of file frame.h.

enum ast_control_t38_rate_management

Enumerator:
AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF 
AST_T38_RATE_MANAGEMENT_LOCAL_TCF 

Definition at line 373 of file frame.h.

enum ast_control_transfer

Enumerator:
AST_TRANSFER_SUCCESS  Transfer request on the channel worked
AST_TRANSFER_FAILED  Transfer request on the channel failed

Definition at line 389 of file frame.h.

00389                           {
00390    AST_TRANSFER_SUCCESS = 0, /*!< Transfer request on the channel worked */
00391    AST_TRANSFER_FAILED,      /*!< Transfer request on the channel failed */
00392 };

enum ast_frame_read_action

Enumerator:
AST_FRAME_READ_ACTION_CONNECTED_LINE_MACRO 

Definition at line 338 of file frame.h.

enum ast_frame_type

Frame types.

Note:
It is important that the values of each frame type are never changed, because it will break backwards compatability with older versions. This is because these constants are transmitted directly over IAX2.
Enumerator:
AST_FRAME_DTMF_END  DTMF end event, subclass is the digit
AST_FRAME_VOICE  Voice data, subclass is AST_FORMAT_*
AST_FRAME_VIDEO  Video frame, maybe?? :)
AST_FRAME_CONTROL  A control frame, subclass is AST_CONTROL_*
AST_FRAME_NULL  An empty, useless frame
AST_FRAME_IAX  Inter Asterisk Exchange private frame type
AST_FRAME_TEXT  Text messages
AST_FRAME_IMAGE  Image Frames
AST_FRAME_HTML  HTML Frame
AST_FRAME_CNG  Comfort Noise frame (subclass is level of CNG in -dBov), body may include zero or more 8-bit quantization coefficients
AST_FRAME_MODEM  Modem-over-IP data streams
AST_FRAME_DTMF_BEGIN  DTMF begin event, subclass is the digit

Definition at line 101 of file frame.h.

00101                     {
00102    /*! DTMF end event, subclass is the digit */
00103    AST_FRAME_DTMF_END = 1,
00104    /*! Voice data, subclass is AST_FORMAT_* */
00105    AST_FRAME_VOICE,
00106    /*! Video frame, maybe?? :) */
00107    AST_FRAME_VIDEO,
00108    /*! A control frame, subclass is AST_CONTROL_* */
00109    AST_FRAME_CONTROL,
00110    /*! An empty, useless frame */
00111    AST_FRAME_NULL,
00112    /*! Inter Asterisk Exchange private frame type */
00113    AST_FRAME_IAX,
00114    /*! Text messages */
00115    AST_FRAME_TEXT,
00116    /*! Image Frames */
00117    AST_FRAME_IMAGE,
00118    /*! HTML Frame */
00119    AST_FRAME_HTML,
00120    /*! Comfort Noise frame (subclass is level of CNG in -dBov), 
00121        body may include zero or more 8-bit quantization coefficients */
00122    AST_FRAME_CNG,
00123    /*! Modem-over-IP data streams */
00124    AST_FRAME_MODEM,  
00125    /*! DTMF begin event, subclass is the digit */
00126    AST_FRAME_DTMF_BEGIN,
00127 };


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_getformatname(), ast_log(), AST_MIN_OFFSET, AST_SMOOTHER_FLAG_G729, ast_swapcopy_samples(), f, ast_smoother::flags, ast_smoother::format, ast_smoother::len, LOG_WARNING, ast_smoother::opt, ast_smoother::opt_needs_swap, ast_smoother::samplesperbyte, ast_smoother::size, 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.codec;
00212       s->samplesperbyte = (float)f->samples / (float)f->datalen;
00213    } else if (s->format != f->subclass.codec) {
00214       ast_log(LOG_WARNING, "Smoother was working on %s format frames, now trying to feed %s?\n",
00215          ast_getformatname(s->format), ast_getformatname(f->subclass.codec));
00216       return -1;
00217    }
00218    if (s->len + f->datalen > SMOOTHER_SIZE) {
00219       ast_log(LOG_WARNING, "Out of smoother space\n");
00220       return -1;
00221    }
00222    if (((f->datalen == s->size) ||
00223         ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) &&
00224        !s->opt &&
00225        !s->len &&
00226        (f->offset >= AST_MIN_OFFSET)) {
00227       /* Optimize by sending the frame we just got
00228          on the next read, thus eliminating the douple
00229          copy */
00230       if (swap)
00231          ast_swapcopy_samples(f->data.ptr, f->data.ptr, f->samples);
00232       s->opt = f;
00233       s->opt_needs_swap = swap ? 1 : 0;
00234       return 0;
00235    }
00236 
00237    return smoother_frame_feed(s, f, swap);
00238 }

char* ast_codec2str ( format_t  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 651 of file frame.c.

References ARRAY_LEN, AST_FORMAT_LIST, and ast_format_list::desc.

Referenced by moh_alloc(), and show_codec_n().

00652 {
00653    int x;
00654    char *ret = "unknown";
00655    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00656       if (AST_FORMAT_LIST[x].bits == codec) {
00657          ret = AST_FORMAT_LIST[x].desc;
00658          break;
00659       }
00660    }
00661    return ret;
00662 }

format_t ast_codec_choose ( struct ast_codec_pref pref,
format_t  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 1224 of file frame.c.

References ARRAY_LEN, ast_best_codec(), ast_debug, 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().

01225 {
01226    int x, slot;
01227    format_t ret = 0;
01228 
01229    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01230       slot = pref->order[x];
01231 
01232       if (!slot)
01233          break;
01234       if (formats & AST_FORMAT_LIST[slot-1].bits) {
01235          ret = AST_FORMAT_LIST[slot-1].bits;
01236          break;
01237       }
01238    }
01239    if (ret & AST_FORMAT_AUDIO_MASK)
01240       return ret;
01241 
01242    ast_debug(4, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec");
01243 
01244       return find_best ? ast_best_codec(formats) : 0;
01245 }

int ast_codec_get_len ( format_t  format,
int  samples 
)

Returns the number of bytes for the number of samples of the given format.

Definition at line 1507 of file frame.c.

References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G719, 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_SIREN14, AST_FORMAT_SIREN7, AST_FORMAT_SLINEAR, AST_FORMAT_SLINEAR16, AST_FORMAT_TESTLAW, AST_FORMAT_ULAW, ast_getformatname(), ast_log(), len(), and LOG_WARNING.

Referenced by moh_generate(), and monmp3thread().

01508 {
01509    int len = 0;
01510 
01511    /* XXX Still need speex, and lpc10 XXX */ 
01512    switch(format) {
01513    case AST_FORMAT_G723_1:
01514       len = (samples / 240) * 20;
01515       break;
01516    case AST_FORMAT_ILBC:
01517       len = (samples / 240) * 50;
01518       break;
01519    case AST_FORMAT_GSM:
01520       len = (samples / 160) * 33;
01521       break;
01522    case AST_FORMAT_G729A:
01523       len = samples / 8;
01524       break;
01525    case AST_FORMAT_SLINEAR:
01526    case AST_FORMAT_SLINEAR16:
01527       len = samples * 2;
01528       break;
01529    case AST_FORMAT_ULAW:
01530    case AST_FORMAT_ALAW:
01531    case AST_FORMAT_TESTLAW:
01532       len = samples;
01533       break;
01534    case AST_FORMAT_G722:
01535    case AST_FORMAT_ADPCM:
01536    case AST_FORMAT_G726:
01537    case AST_FORMAT_G726_AAL2:
01538       len = samples / 2;
01539       break;
01540    case AST_FORMAT_SIREN7:
01541       /* 16,000 samples per second at 32kbps is 4,000 bytes per second */
01542       len = samples / (16000 / 4000);
01543       break;
01544    case AST_FORMAT_SIREN14:
01545       /* 32,000 samples per second at 48kbps is 6,000 bytes per second */
01546       len = (int) samples / ((float) 32000 / 6000);
01547       break;
01548    case AST_FORMAT_G719:
01549       /* 48,000 samples per second at 64kbps is 8,000 bytes per second */
01550       len = (int) samples / ((float) 48000 / 8000);
01551       break;
01552    default:
01553       ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format));
01554    }
01555 
01556    return len;
01557 }

int ast_codec_get_samples ( struct ast_frame f  ) 

Returns the number of samples contained in the frame.

Definition at line 1445 of file frame.c.

References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G719, 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_SIREN14, AST_FORMAT_SIREN7, AST_FORMAT_SLINEAR, AST_FORMAT_SLINEAR16, AST_FORMAT_SPEEX, AST_FORMAT_SPEEX16, AST_FORMAT_TESTLAW, AST_FORMAT_ULAW, ast_getformatname_multiple(), 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().

01446 {
01447    int samples = 0;
01448    char tmp[64];
01449 
01450    switch (f->subclass.codec) {
01451    case AST_FORMAT_SPEEX:
01452       samples = speex_samples(f->data.ptr, f->datalen);
01453       break;
01454    case AST_FORMAT_SPEEX16:
01455       samples = 2 * speex_samples(f->data.ptr, f->datalen);
01456       break;
01457    case AST_FORMAT_G723_1:
01458       samples = g723_samples(f->data.ptr, f->datalen);
01459       break;
01460    case AST_FORMAT_ILBC:
01461       samples = 240 * (f->datalen / 50);
01462       break;
01463    case AST_FORMAT_GSM:
01464       samples = 160 * (f->datalen / 33);
01465       break;
01466    case AST_FORMAT_G729A:
01467       samples = f->datalen * 8;
01468       break;
01469    case AST_FORMAT_SLINEAR:
01470    case AST_FORMAT_SLINEAR16:
01471       samples = f->datalen / 2;
01472       break;
01473    case AST_FORMAT_LPC10:
01474       /* assumes that the RTP packet contains one LPC10 frame */
01475       samples = 22 * 8;
01476       samples += (((char *)(f->data.ptr))[7] & 0x1) * 8;
01477       break;
01478    case AST_FORMAT_ULAW:
01479    case AST_FORMAT_ALAW:
01480    case AST_FORMAT_TESTLAW:
01481       samples = f->datalen;
01482       break;
01483    case AST_FORMAT_G722:
01484    case AST_FORMAT_ADPCM:
01485    case AST_FORMAT_G726:
01486    case AST_FORMAT_G726_AAL2:
01487       samples = f->datalen * 2;
01488       break;
01489    case AST_FORMAT_SIREN7:
01490       /* 16,000 samples per second at 32kbps is 4,000 bytes per second */
01491       samples = f->datalen * (16000 / 4000);
01492       break;
01493    case AST_FORMAT_SIREN14:
01494       /* 32,000 samples per second at 48kbps is 6,000 bytes per second */
01495       samples = (int) f->datalen * ((float) 32000 / 6000);
01496       break;
01497    case AST_FORMAT_G719:
01498       /* 48,000 samples per second at 64kbps is 8,000 bytes per second */
01499       samples = (int) f->datalen * ((float) 48000 / 8000);
01500       break;
01501    default:
01502       ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), f->subclass.codec));
01503    }
01504    return samples;
01505 }

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

Gets duration in ms of interpolation frame for a format.

Definition at line 749 of file frame.h.

References AST_FORMAT_ILBC.

Referenced by __get_from_jb(), and jb_get_and_deliver().

00750 { 
00751    return (format == AST_FORMAT_ILBC) ? 30 : 20;
00752 }

int ast_codec_pref_append ( struct ast_codec_pref pref,
format_t  format 
)

Append a audio codec to a preference list, removing it first if it was already there.

Definition at line 1084 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().

01085 {
01086    int x, newindex = 0;
01087 
01088    ast_codec_pref_remove(pref, format);
01089 
01090    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01091       if (AST_FORMAT_LIST[x].bits == format) {
01092          newindex = x + 1;
01093          break;
01094       }
01095    }
01096 
01097    if (newindex) {
01098       for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01099          if (!pref->order[x]) {
01100             pref->order[x] = newindex;
01101             break;
01102          }
01103       }
01104    }
01105 
01106    return x;
01107 }

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.

Note:
Due to a misunderstanding in how codec preferences are stored, this list starts at 'B', not 'A'. For backwards compatibility reasons, this cannot change.
Parameters:
pref A codec preference list structure
buf A string denoting codec preference, appropriate for use in line transmission
size Size of buf
right Boolean: if 0, convert from buf to pref; if 1, convert from pref to buf.

Definition at line 987 of file frame.c.

References ast_codec_pref::order.

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

00988 {
00989    int x, differential = (int) 'A', mem;
00990    char *from, *to;
00991 
00992    if (right) {
00993       from = pref->order;
00994       to = buf;
00995       mem = size;
00996    } else {
00997       to = pref->order;
00998       from = buf;
00999       mem = sizeof(format_t) * 8;
01000    }
01001 
01002    memset(to, 0, mem);
01003    for (x = 0; x < sizeof(format_t) * 8; x++) {
01004       if (!from[x])
01005          break;
01006       to[x] = right ? (from[x] + differential) : (from[x] - differential);
01007    }
01008 }

struct ast_format_list ast_codec_pref_getsize ( struct ast_codec_pref pref,
format_t  format 
)

Get packet size for codec.

Definition at line 1185 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_write(), handle_open_receive_channel_ack_message(), skinny_set_rtp_peer(), and transmit_connect().

01186 {
01187    int x, idx = -1, framems = 0;
01188    struct ast_format_list fmt = { 0, };
01189 
01190    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01191       if (AST_FORMAT_LIST[x].bits == format) {
01192          fmt = AST_FORMAT_LIST[x];
01193          idx = x;
01194          break;
01195       }
01196    }
01197 
01198    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01199       if (pref->order[x] == (idx + 1)) {
01200          framems = pref->framing[x];
01201          break;
01202       }
01203    }
01204 
01205    /* size validation */
01206    if (!framems)
01207       framems = AST_FORMAT_LIST[idx].def_ms;
01208 
01209    if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */
01210       framems -= framems % AST_FORMAT_LIST[idx].inc_ms;
01211 
01212    if (framems < AST_FORMAT_LIST[idx].min_ms)
01213       framems = AST_FORMAT_LIST[idx].min_ms;
01214 
01215    if (framems > AST_FORMAT_LIST[idx].max_ms)
01216       framems = AST_FORMAT_LIST[idx].max_ms;
01217 
01218    fmt.cur_ms = framems;
01219 
01220    return fmt;
01221 }

format_t ast_codec_pref_index ( struct ast_codec_pref pref,
int  index 
)

Codec located at a particular place in the preference index.

Definition at line 1046 of file frame.c.

References AST_FORMAT_LIST, ast_format_list::bits, and ast_codec_pref::order.

Referenced by _sip_show_peer(), _skinny_show_line(), 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().

01047 {
01048    int slot = 0;
01049 
01050    if ((idx >= 0) && (idx < sizeof(pref->order))) {
01051       slot = pref->order[idx];
01052    }
01053 
01054    return slot ? AST_FORMAT_LIST[slot - 1].bits : 0;
01055 }

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,
format_t  format,
int  only_if_existing 
)

Prepend an audio codec to a preference list, removing it first if it was already there.

Definition at line 1110 of file frame.c.

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

Referenced by create_addr().

01111 {
01112    int x, newindex = 0;
01113 
01114    /* First step is to get the codecs "index number" */
01115    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01116       if (AST_FORMAT_LIST[x].bits == format) {
01117          newindex = x + 1;
01118          break;
01119       }
01120    }
01121    /* Done if its unknown */
01122    if (!newindex)
01123       return;
01124 
01125    /* Now find any existing occurrence, or the end */
01126    for (x = 0; x < sizeof(format_t) * 8; x++) {
01127       if (!pref->order[x] || pref->order[x] == newindex)
01128          break;
01129    }
01130 
01131    if (only_if_existing && !pref->order[x])
01132       return;
01133 
01134    /* Move down to make space to insert - either all the way to the end,
01135       or as far as the existing location (which will be overwritten) */
01136    for (; x > 0; x--) {
01137       pref->order[x] = pref->order[x - 1];
01138       pref->framing[x] = pref->framing[x - 1];
01139    }
01140 
01141    /* And insert the new entry */
01142    pref->order[0] = newindex;
01143    pref->framing[0] = 0; /* ? */
01144 }

void ast_codec_pref_remove ( struct ast_codec_pref pref,
format_t  format 
)

Remove audio a codec from a preference list.

Definition at line 1058 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().

01059 {
01060    struct ast_codec_pref oldorder;
01061    int x, y = 0;
01062    int slot;
01063    int size;
01064 
01065    if (!pref->order[0])
01066       return;
01067 
01068    memcpy(&oldorder, pref, sizeof(oldorder));
01069    memset(pref, 0, sizeof(*pref));
01070 
01071    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01072       slot = oldorder.order[x];
01073       size = oldorder.framing[x];
01074       if (! slot)
01075          break;
01076       if (AST_FORMAT_LIST[slot-1].bits != format) {
01077          pref->order[y] = slot;
01078          pref->framing[y++] = size;
01079       }
01080    }
01081 }

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

Set packet size for codec.

Definition at line 1147 of file frame.c.

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

Referenced by ast_parse_allow_disallow(), and process_sdp_a_audio().

01148 {
01149    int x, idx = -1;
01150 
01151    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01152       if (AST_FORMAT_LIST[x].bits == format) {
01153          idx = x;
01154          break;
01155       }
01156    }
01157 
01158    if (idx < 0)
01159       return -1;
01160 
01161    /* size validation */
01162    if (!framems)
01163       framems = AST_FORMAT_LIST[idx].def_ms;
01164 
01165    if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */
01166       framems -= framems % AST_FORMAT_LIST[idx].inc_ms;
01167 
01168    if (framems < AST_FORMAT_LIST[idx].min_ms)
01169       framems = AST_FORMAT_LIST[idx].min_ms;
01170 
01171    if (framems > AST_FORMAT_LIST[idx].max_ms)
01172       framems = AST_FORMAT_LIST[idx].max_ms;
01173 
01174    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01175       if (pref->order[x] == (idx + 1)) {
01176          pref->framing[x] = framems;
01177          break;
01178       }
01179    }
01180 
01181    return x;
01182 }

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 1010 of file frame.c.

References ast_codec_pref_index(), and ast_getformatname().

Referenced by dump_prefs(), and socket_process().

01011 {
01012    int x;
01013    format_t codec; 
01014    size_t total_len, slen;
01015    char *formatname;
01016    
01017    memset(buf, 0, size);
01018    total_len = size;
01019    buf[0] = '(';
01020    total_len--;
01021    for (x = 0; x < sizeof(format_t) * 8; x++) {
01022       if (total_len <= 0)
01023          break;
01024       if (!(codec = ast_codec_pref_index(pref,x)))
01025          break;
01026       if ((formatname = ast_getformatname(codec))) {
01027          slen = strlen(formatname);
01028          if (slen > total_len)
01029             break;
01030          strncat(buf, formatname, total_len - 1); /* safe */
01031          total_len -= slen;
01032       }
01033       if (total_len && x < sizeof(format_t) * 8 - 1 && ast_codec_pref_index(pref, x + 1)) {
01034          strncat(buf, "|", total_len - 1); /* safe */
01035          total_len--;
01036       }
01037    }
01038    if (total_len) {
01039       strncat(buf, ")", total_len - 1); /* safe */
01040       total_len--;
01041    }
01042 
01043    return size - total_len;
01044 }

static force_inline int ast_format_rate ( format_t  format  )  [static]

Get the sample rate for a given format.

Definition at line 776 of file frame.h.

References AST_FORMAT_G719, AST_FORMAT_G722, AST_FORMAT_SIREN14, AST_FORMAT_SIREN7, AST_FORMAT_SLINEAR16, and AST_FORMAT_SPEEX16.

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

00777 {
00778    switch (format) {
00779    case AST_FORMAT_G722:
00780    case AST_FORMAT_SLINEAR16:
00781    case AST_FORMAT_SIREN7:
00782    case AST_FORMAT_SPEEX16:
00783       return 16000;
00784    case AST_FORMAT_SIREN14:
00785       return 32000;
00786    case AST_FORMAT_G719:
00787       return 48000;
00788    default:
00789       return 8000;
00790    }
00791 }

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 1559 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().

01560 {
01561    int count;
01562    short *fdata = f->data.ptr;
01563    short adjust_value = abs(adjustment);
01564 
01565    if ((f->frametype != AST_FRAME_VOICE) || (f->subclass.codec != AST_FORMAT_SLINEAR))
01566       return -1;
01567 
01568    if (!adjustment)
01569       return 0;
01570 
01571    for (count = 0; count < f->samples; count++) {
01572       if (adjustment > 0) {
01573          ast_slinear_saturated_multiply(&fdata[count], &adjust_value);
01574       } else if (adjustment < 0) {
01575          ast_slinear_saturated_divide(&fdata[count], &adjust_value);
01576       }
01577    }
01578 
01579    return 0;
01580 }

int ast_frame_clear ( struct ast_frame frame  ) 

Clear all audio samples from an ast_frame. The frame must be AST_FRAME_VOICE and AST_FORMAT_SLINEAR.

Definition at line 1604 of file frame.c.

References AST_LIST_NEXT, ast_frame::data, ast_frame::datalen, ast_frame::next, and ast_frame::ptr.

Referenced by ast_audiohook_write_frame(), and mute_callback().

01605 {
01606    struct ast_frame *next;
01607 
01608    for (next = AST_LIST_NEXT(frame, frame_list);
01609        frame;
01610        frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) {
01611       memset(frame->data.ptr, 0, frame->datalen);
01612    }
01613    return 0;
01614 }

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

Dump a frame for debugging purposes

Definition at line 769 of file frame.c.

References AST_CONTROL_ANSWER, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_FLASH, AST_CONTROL_HANGUP, AST_CONTROL_HOLD, AST_CONTROL_OFFHOOK, AST_CONTROL_OPTION, AST_CONTROL_RADIO_KEY, AST_CONTROL_RADIO_UNKEY, AST_CONTROL_RING, AST_CONTROL_RINGING, AST_CONTROL_T38_PARAMETERS, AST_CONTROL_TAKEOFFHOOK, AST_CONTROL_UNHOLD, AST_CONTROL_WINK, ast_copy_string(), AST_FRAME_CONTROL, AST_FRAME_DTMF_BEGIN, AST_FRAME_DTMF_END, AST_FRAME_HTML, AST_FRAME_IAX, AST_FRAME_IMAGE, AST_FRAME_MODEM, AST_FRAME_NULL, AST_FRAME_TEXT, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_getformatname(), AST_HTML_BEGIN, AST_HTML_DATA, AST_HTML_END, AST_HTML_LDCOMPLETE, AST_HTML_LINKREJECT, AST_HTML_LINKURL, AST_HTML_NOSUPPORT, AST_HTML_UNLINK, AST_HTML_URL, AST_MODEM_T38, AST_MODEM_V150, ast_strlen_zero(), AST_T38_NEGOTIATED, AST_T38_REFUSED, AST_T38_REQUEST_NEGOTIATE, AST_T38_REQUEST_TERMINATE, AST_T38_TERMINATED, ast_verbose, COLOR_BLACK, COLOR_BRCYAN, COLOR_BRGREEN, COLOR_BRMAGENTA, COLOR_BRRED, COLOR_YELLOW, f, ast_control_t38_parameters::request_response, and term_color().

Referenced by __ast_read(), and ast_write().

00770 {
00771    const char noname[] = "unknown";
00772    char ftype[40] = "Unknown Frametype";
00773    char cft[80];
00774    char subclass[40] = "Unknown Subclass";
00775    char csub[80];
00776    char moreinfo[40] = "";
00777    char cn[60];
00778    char cp[40];
00779    char cmn[40];
00780    const char *message = "Unknown";
00781 
00782    if (!name)
00783       name = noname;
00784 
00785 
00786    if (!f) {
00787       ast_verbose("%s [ %s (NULL) ] [%s]\n", 
00788          term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00789          term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 
00790          term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00791       return;
00792    }
00793    /* XXX We should probably print one each of voice and video when the format changes XXX */
00794    if (f->frametype == AST_FRAME_VOICE)
00795       return;
00796    if (f->frametype == AST_FRAME_VIDEO)
00797       return;
00798    switch(f->frametype) {
00799    case AST_FRAME_DTMF_BEGIN:
00800       strcpy(ftype, "DTMF Begin");
00801       subclass[0] = f->subclass.integer;
00802       subclass[1] = '\0';
00803       break;
00804    case AST_FRAME_DTMF_END:
00805       strcpy(ftype, "DTMF End");
00806       subclass[0] = f->subclass.integer;
00807       subclass[1] = '\0';
00808       break;
00809    case AST_FRAME_CONTROL:
00810       strcpy(ftype, "Control");
00811       switch (f->subclass.integer) {
00812       case AST_CONTROL_HANGUP:
00813          strcpy(subclass, "Hangup");
00814          break;
00815       case AST_CONTROL_RING:
00816          strcpy(subclass, "Ring");
00817          break;
00818       case AST_CONTROL_RINGING:
00819          strcpy(subclass, "Ringing");
00820          break;
00821       case AST_CONTROL_ANSWER:
00822          strcpy(subclass, "Answer");
00823          break;
00824       case AST_CONTROL_BUSY:
00825          strcpy(subclass, "Busy");
00826          break;
00827       case AST_CONTROL_TAKEOFFHOOK:
00828          strcpy(subclass, "Take Off Hook");
00829          break;
00830       case AST_CONTROL_OFFHOOK:
00831          strcpy(subclass, "Line Off Hook");
00832          break;
00833       case AST_CONTROL_CONGESTION:
00834          strcpy(subclass, "Congestion");
00835          break;
00836       case AST_CONTROL_FLASH:
00837          strcpy(subclass, "Flash");
00838          break;
00839       case AST_CONTROL_WINK:
00840          strcpy(subclass, "Wink");
00841          break;
00842       case AST_CONTROL_OPTION:
00843          strcpy(subclass, "Option");
00844          break;
00845       case AST_CONTROL_RADIO_KEY:
00846          strcpy(subclass, "Key Radio");
00847          break;
00848       case AST_CONTROL_RADIO_UNKEY:
00849          strcpy(subclass, "Unkey Radio");
00850          break;
00851       case AST_CONTROL_HOLD:
00852          strcpy(subclass, "Hold");
00853          break;
00854       case AST_CONTROL_UNHOLD:
00855          strcpy(subclass, "Unhold");
00856          break;
00857       case AST_CONTROL_T38_PARAMETERS:
00858          if (f->datalen != sizeof(struct ast_control_t38_parameters)) {
00859             message = "Invalid";
00860          } else {
00861             struct ast_control_t38_parameters *parameters = f->data.ptr;
00862             enum ast_control_t38 state = parameters->request_response;
00863             if (state == AST_T38_REQUEST_NEGOTIATE)
00864                message = "Negotiation Requested";
00865             else if (state == AST_T38_REQUEST_TERMINATE)
00866                message = "Negotiation Request Terminated";
00867             else if (state == AST_T38_NEGOTIATED)
00868                message = "Negotiated";
00869             else if (state == AST_T38_TERMINATED)
00870                message = "Terminated";
00871             else if (state == AST_T38_REFUSED)
00872                message = "Refused";
00873          }
00874          snprintf(subclass, sizeof(subclass), "T38_Parameters/%s", message);
00875          break;
00876       case -1:
00877          strcpy(subclass, "Stop generators");
00878          break;
00879       default:
00880          snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass.integer);
00881       }
00882       break;
00883    case AST_FRAME_NULL:
00884       strcpy(ftype, "Null Frame");
00885       strcpy(subclass, "N/A");
00886       break;
00887    case AST_FRAME_IAX:
00888       /* Should never happen */
00889       strcpy(ftype, "IAX Specific");
00890       snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass.integer);
00891       break;
00892    case AST_FRAME_TEXT:
00893       strcpy(ftype, "Text");
00894       strcpy(subclass, "N/A");
00895       ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo));
00896       break;
00897    case AST_FRAME_IMAGE:
00898       strcpy(ftype, "Image");
00899       snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass.codec));
00900       break;
00901    case AST_FRAME_HTML:
00902       strcpy(ftype, "HTML");
00903       switch (f->subclass.integer) {
00904       case AST_HTML_URL:
00905          strcpy(subclass, "URL");
00906          ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo));
00907          break;
00908       case AST_HTML_DATA:
00909          strcpy(subclass, "Data");
00910          break;
00911       case AST_HTML_BEGIN:
00912          strcpy(subclass, "Begin");
00913          break;
00914       case AST_HTML_END:
00915          strcpy(subclass, "End");
00916          break;
00917       case AST_HTML_LDCOMPLETE:
00918          strcpy(subclass, "Load Complete");
00919          break;
00920       case AST_HTML_NOSUPPORT:
00921          strcpy(subclass, "No Support");
00922          break;
00923       case AST_HTML_LINKURL:
00924          strcpy(subclass, "Link URL");
00925          ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo));
00926          break;
00927       case AST_HTML_UNLINK:
00928          strcpy(subclass, "Unlink");
00929          break;
00930       case AST_HTML_LINKREJECT:
00931          strcpy(subclass, "Link Reject");
00932          break;
00933       default:
00934          snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass.integer);
00935          break;
00936       }
00937       break;
00938    case AST_FRAME_MODEM:
00939       strcpy(ftype, "Modem");
00940       switch (f->subclass.integer) {
00941       case AST_MODEM_T38:
00942          strcpy(subclass, "T.38");
00943          break;
00944       case AST_MODEM_V150:
00945          strcpy(subclass, "V.150");
00946          break;
00947       default:
00948          snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass.integer);
00949          break;
00950       }
00951       break;
00952    default:
00953       snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype);
00954    }
00955    if (!ast_strlen_zero(moreinfo))
00956       ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n",  
00957              term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00958              term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)),
00959              f->frametype, 
00960              term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)),
00961              f->subclass.integer, 
00962              term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)),
00963              term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00964    else
00965       ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n",  
00966              term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00967              term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)),
00968              f->frametype, 
00969              term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)),
00970              f->subclass.integer, 
00971              term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00972 }

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 371 of file frame.c.

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

Referenced by mixmonitor_thread().

00372 {
00373    struct ast_frame *next;
00374 
00375    for (next = AST_LIST_NEXT(frame, frame_list);
00376         frame;
00377         frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) {
00378       __frame_free(frame, cache);
00379    }
00380 }

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 1582 of file frame.c.

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

01583 {
01584    int count;
01585    short *data1, *data2;
01586 
01587    if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass.codec != AST_FORMAT_SLINEAR))
01588       return -1;
01589 
01590    if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass.codec != AST_FORMAT_SLINEAR))
01591       return -1;
01592 
01593    if (f1->samples != f2->samples)
01594       return -1;
01595 
01596    for (count = 0, data1 = f1->data.ptr, data2 = f2->data.ptr;
01597         count < f1->samples;
01598         count++, data1++, data2++)
01599       ast_slinear_saturated_add(data1, data2);
01600 
01601    return 0;
01602 }

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(), ast_frame_subclass::codec, ast_frame::data, ast_frame::datalen, ast_frame::delivery, f, frame_cache, frames, ast_frame::frametype, len(), ast_frame::len, ast_frame::mallocd, ast_frame::mallocd_hdr_len, ast_frame::offset, ast_frame::ptr, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, ast_frame::ts, and ast_frame::uint32.

Referenced by __ast_queue_frame(), ast_frisolate(), ast_indicate_data(), ast_jb_put(), ast_rtp_write(), ast_slinfactory_feed(), audiohook_read_frame_single(), autoservice_run(), multicast_rtp_write(), process_dtmf_rfc2833(), recordthread(), and rpt().

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.codec = f->subclass.codec;
00519    out->datalen = f->datalen;
00520    out->samples = f->samples;
00521    out->delivery = f->delivery;
00522    /* Set us as having malloc'd header only, so it will eventually
00523       get freed. */
00524    out->mallocd = AST_MALLOCD_HDR;
00525    out->offset = AST_FRIENDLY_OFFSET;
00526    if (out->datalen) {
00527       out->data.ptr = buf + sizeof(*out) + AST_FRIENDLY_OFFSET;
00528       memcpy(out->data.ptr, f->data.ptr, out->datalen);  
00529    } else {
00530       out->data.uint32 = f->data.uint32;
00531    }
00532    if (srclen > 0) {
00533       /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */
00534       char *src;
00535       out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen;
00536       src = (char *) out->src;
00537       /* Must have space since we allocated for it */
00538       strcpy(src, f->src);
00539    }
00540    ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO);
00541    out->ts = f->ts;
00542    out->len = f->len;
00543    out->seqno = f->seqno;
00544    return out;
00545 }

struct ast_frame* ast_frisolate ( struct ast_frame fr  ) 

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 387 of file frame.c.

References ast_copy_flags, ast_frame_header_new(), ast_frdup(), ast_free, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, ast_malloc, AST_MALLOCD_DATA, AST_MALLOCD_HDR, AST_MALLOCD_SRC, ast_strdup, ast_test_flag, ast_frame_subclass::codec, ast_frame::data, ast_frame::datalen, ast_frame::frametype, ast_frame::len, ast_frame::mallocd, ast_frame::offset, ast_frame::ptr, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, ast_frame::ts, and ast_frame::uint32.

Referenced by __ast_answer(), ast_rtp_read(), ast_safe_sleep_conditional(), ast_slinfactory_feed(), ast_trans_frameout(), ast_write(), autoservice_run(), dahdi_decoder_frameout(), dahdi_encoder_frameout(), feature_request_and_dial(), jpeg_read_image(), read_frame(), spandsp_fax_read(), and t38_tx_packet_handler().

00388 {
00389    struct ast_frame *out;
00390    void *newdata;
00391 
00392    /* if none of the existing frame is malloc'd, let ast_frdup() do it
00393       since it is more efficient
00394    */
00395    if (fr->mallocd == 0) {
00396       return ast_frdup(fr);
00397    }
00398 
00399    /* if everything is already malloc'd, we are done */
00400    if ((fr->mallocd & (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) ==
00401        (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) {
00402       return fr;
00403    }
00404 
00405    if (!(fr->mallocd & AST_MALLOCD_HDR)) {
00406       /* Allocate a new header if needed */
00407       if (!(out = ast_frame_header_new())) {
00408          return NULL;
00409       }
00410       out->frametype = fr->frametype;
00411       out->subclass.codec = fr->subclass.codec;
00412       out->datalen = fr->datalen;
00413       out->samples = fr->samples;
00414       out->offset = fr->offset;
00415       /* Copy the timing data */
00416       ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO);
00417       if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) {
00418          out->ts = fr->ts;
00419          out->len = fr->len;
00420          out->seqno = fr->seqno;
00421       }
00422    } else {
00423       out = fr;
00424    }
00425    
00426    if (!(fr->mallocd & AST_MALLOCD_SRC) && fr->src) {
00427       if (!(out->src = ast_strdup(fr->src))) {
00428          if (out != fr) {
00429             ast_free(out);
00430          }
00431          return NULL;
00432       }
00433    } else {
00434       out->src = fr->src;
00435       fr->src = NULL;
00436       fr->mallocd &= ~AST_MALLOCD_SRC;
00437    }
00438    
00439    if (!(fr->mallocd & AST_MALLOCD_DATA))  {
00440       if (!fr->datalen) {
00441          out->data.uint32 = fr->data.uint32;
00442          out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC;
00443          return out;
00444       }
00445       if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) {
00446          if (out->src != fr->src) {
00447             ast_free((void *) out->src);
00448          }
00449          if (out != fr) {
00450             ast_free(out);
00451          }
00452          return NULL;
00453       }
00454       newdata += AST_FRIENDLY_OFFSET;
00455       out->offset = AST_FRIENDLY_OFFSET;
00456       out->datalen = fr->datalen;
00457       memcpy(newdata, fr->data.ptr, fr->datalen);
00458       out->data.ptr = newdata;
00459    } else {
00460       out->data = fr->data;
00461       memset(&fr->data, 0, sizeof(fr->data));
00462       fr->mallocd &= ~AST_MALLOCD_DATA;
00463    }
00464 
00465    out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA;
00466    
00467    return out;
00468 }

struct ast_format_list* ast_get_format_list ( size_t *  size  ) 

Definition at line 563 of file frame.c.

References ARRAY_LEN, and AST_FORMAT_LIST.

Referenced by ast_data_add_codecs(), complete_trans_path_choice(), and handle_cli_core_show_translation().

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

struct ast_format_list* ast_get_format_list_index ( int  index  ) 

Definition at line 558 of file frame.c.

References AST_FORMAT_LIST.

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

format_t 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 632 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(), and try_suggested_sip_codec().

00633 {
00634    int x, all;
00635    format_t format = 0;
00636 
00637    all = strcasecmp(name, "all") ? 0 : 1;
00638    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00639       if (all || 
00640            !strcasecmp(AST_FORMAT_LIST[x].name,name) ||
00641            !strcasecmp(AST_FORMAT_LIST[x].name, ast_expand_codec_alias(name))) {
00642          format |= AST_FORMAT_LIST[x].bits;
00643          if (!all)
00644             break;
00645       }
00646    }
00647 
00648    return format;
00649 }

char* ast_getformatname ( format_t  format  ) 

Get the name of a format.

Parameters:
format id of format
Returns:
A static string containing the name of the format or "unknown" if unknown.

Definition at line 569 of file frame.c.

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

Referenced by __ast_read(), __ast_register_translator(), __ast_smoother_feed(), _sip_show_peer(), _skinny_show_line(), add_codec_to_answer(), add_codec_to_sdp(), add_sdp(), add_tcodec_to_sdp(), add_vcodec_to_sdp(), agent_call(), ast_channel_make_compatible_helper(), ast_codec_get_len(), ast_codec_pref_string(), ast_do_masquerade(), ast_dsp_process(), ast_frame_dump(), ast_openvstream(), ast_rtp_instance_bridge(), ast_rtp_write(), ast_slinfactory_feed(), ast_stopstream(), ast_streamfile(), ast_translate_path_to_str(), ast_translator_build_path(), ast_unregister_translator(), ast_writestream(), background_detect_exec(), bridge_channel_join(), bridge_make_compatible(), conf_run(), dahdi_read(), dahdi_write(), do_waiting(), dump_versioned_codec(), eagi_exec(), func_channel_read(), function_iaxpeer(), function_sippeer(), g719write(), g726_write(), g729_write(), gsm_write(), gtalk_rtp_read(), gtalk_show_channels(), gtalk_write(), h263_write(), h264_write(), 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(), handle_open_receive_channel_ack_message(), iax2_request(), iax_show_provisioning(), ilbc_write(), isAnsweringMachine(), jack_hook_callback(), jingle_rtp_read(), jingle_show_channels(), jingle_write(), login_exec(), mgcp_rtp_read(), mgcp_write(), misdn_write(), moh_files_release(), moh_release(), nbs_request(), nbs_xwrite(), ogg_vorbis_write(), oh323_rtp_read(), oh323_write(), pcm_write(), phone_setup(), phone_write(), print_codec_to_cli(), print_frame(), process_sdp_a_audio(), rebuild_matrix(), register_translator(), remote_bridge_loop(), set_format(), set_local_capabilities(), set_peer_capabilities(), setup_rtp_connection(), sip_request_call(), sip_rtp_read(), sip_write(), siren14write(), siren7write(), skinny_new(), skinny_rtp_read(), skinny_set_rtp_peer(), skinny_write(), slinear_write(), socket_process(), start_rtp(), unistim_new(), unistim_request(), unistim_rtp_read(), unistim_write(), vox_write(), and wav_write().

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

char* ast_getformatname_multiple ( char *  buf,
size_t  size,
format_t  format 
)

Get the names of a set of formats.

Parameters:
buf a buffer for the output string
size size of buf (bytes)
format the format (combined IDs of codecs) Prints a list of readable codec names corresponding to "format". ex: for format=AST_FORMAT_GSM|AST_FORMAT_SPEEX|AST_FORMAT_ILBC it will return "0x602 (GSM|SPEEX|ILBC)"
Returns:
The return value is buf.

Definition at line 582 of file frame.c.

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

Referenced by __ast_read(), _sip_show_peer(), _skinny_show_device(), _skinny_show_line(), add_sdp(), alsa_request(), ast_best_codec(), ast_bridge_new(), ast_codec_get_samples(), ast_request(), ast_streamfile(), bridge_make_compatible(), console_request(), find_best_technology(), function_iaxpeer(), function_sippeer(), gtalk_is_answered(), gtalk_newcall(), gtalk_write(), handle_capabilities_res_message(), handle_cli_core_show_channeltype(), handle_cli_iax2_show_peer(), handle_showchan(), jingle_write(), mgcp_request(), mgcp_write(), oh323_request(), oh323_write(), oss_request(), phone_request(), process_sdp(), serialize_showchan(), set_format(), setup_rtp_connection(), show_channels_cb(), sip_new(), sip_request_call(), sip_show_channel(), sip_show_settings(), sip_write(), skinny_new(), skinny_request(), skinny_write(), smart_bridge_operation(), socket_process(), start_rtp(), unistim_new(), unistim_request(), and unistim_write().

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

int ast_parse_allow_disallow ( struct ast_codec_pref pref,
format_t 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 1247 of file frame.c.

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

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

01248 {
01249    int errors = 0, framems = 0;
01250    char *parse = NULL, *this = NULL, *psize = NULL;
01251    format_t format = 0;
01252 
01253    parse = ast_strdupa(list);
01254    while ((this = strsep(&parse, ","))) {
01255       framems = 0;
01256       if ((psize = strrchr(this, ':'))) {
01257          *psize++ = '\0';
01258          ast_debug(1, "Packetization for codec: %s is %s\n", this, psize);
01259          framems = atoi(psize);
01260          if (framems < 0) {
01261             framems = 0;
01262             errors++;
01263             ast_log(LOG_WARNING, "Bad packetization value for codec %s\n", this);
01264          }
01265       }
01266       if (!(format = ast_getformatbyname(this))) {
01267          ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this);
01268          errors++;
01269          continue;
01270       }
01271 
01272       if (mask) {
01273          if (allowing)
01274             *mask |= format;
01275          else
01276             *mask &= ~format;
01277       }
01278 
01279       /* Set up a preference list for audio. Do not include video in preferences 
01280          since we can not transcode video and have to use whatever is offered
01281        */
01282       if (pref && (format & AST_FORMAT_AUDIO_MASK)) {
01283          if (strcasecmp(this, "all")) {
01284             if (allowing) {
01285                ast_codec_pref_append(pref, format);
01286                ast_codec_pref_setsize(pref, format, framems);
01287             }
01288             else
01289                ast_codec_pref_remove(pref, format);
01290          } else if (!allowing) {
01291             memset(pref, 0, sizeof(*pref));
01292          }
01293       }
01294    }
01295    return errors;
01296 }

void ast_smoother_free ( struct ast_smoother s  ) 

Definition at line 290 of file frame.c.

References ast_free.

Referenced by ast_rtp_destroy(), ast_rtp_write(), destroy_session(), and generic_fax_exec().

00291 {
00292    ast_free(s);
00293 }

int ast_smoother_get_flags ( struct ast_smoother smoother  ) 

Definition at line 189 of file frame.c.

References ast_smoother::flags.

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, and ast_smoother_reset().

Referenced by ast_rtp_write(), and generic_fax_exec().

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 240 of file frame.c.

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

Referenced by ast_rtp_write(), and generic_fax_exec().

00241 {
00242    struct ast_frame *opt;
00243    int len;
00244 
00245    /* IF we have an optimization frame, send it */
00246    if (s->opt) {
00247       if (s->opt->offset < AST_FRIENDLY_OFFSET)
00248          ast_log(LOG_WARNING, "Returning a frame of inappropriate offset (%d).\n",
00249                      s->opt->offset);
00250       opt = s->opt;
00251       s->opt = NULL;
00252       return opt;
00253    }
00254 
00255    /* Make sure we have enough data */
00256    if (s->len < s->size) {
00257       /* Or, if this is a G.729 frame with VAD on it, send it immediately anyway */
00258       if (!((s->flags & AST_SMOOTHER_FLAG_G729) && (s->len % 10)))
00259          return NULL;
00260    }
00261    len = s->size;
00262    if (len > s->len)
00263       len = s->len;
00264    /* Make frame */
00265    s->f.frametype = AST_FRAME_VOICE;
00266    s->f.subclass.codec = s->format;
00267    s->f.data.ptr = s->framedata + AST_FRIENDLY_OFFSET;
00268    s->f.offset = AST_FRIENDLY_OFFSET;
00269    s->f.datalen = len;
00270    /* Samples will be improper given VAD, but with VAD the concept really doesn't even exist */
00271    s->f.samples = len * s->samplesperbyte;   /* XXX rounding */
00272    s->f.delivery = s->delivery;
00273    /* Fill Data */
00274    memcpy(s->f.data.ptr, s->data, len);
00275    s->len -= len;
00276    /* Move remaining data to the front if applicable */
00277    if (s->len) {
00278       /* In principle this should all be fine because if we are sending
00279          G.729 VAD, the next timestamp will take over anyawy */
00280       memmove(s->data, s->data + len, s->len);
00281       if (!ast_tvzero(s->delivery)) {
00282          /* If we have delivery time, increment it, otherwise, leave it at 0 */
00283          s->delivery = ast_tvadd(s->delivery, ast_samp2tv(s->f.samples, ast_format_rate(s->format)));
00284       }
00285    }
00286    /* Return frame */
00287    return &s->f;
00288 }

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

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.

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 ast_smoother::flags.

Referenced by 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 ast_smoother::flags.

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 547 of file frame.c.

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

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


Variable Documentation

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 __analog_handle_event(), __ast_channel_masquerade(), __ast_read(), __oh323_rtp_create(), __oh323_update_info(), agent_new(), agent_read(), ast_channel_setwhentohangup_tv(), ast_do_masquerade(), ast_rtcp_read(), ast_rtp_read(), ast_softhangup_nolock(), ast_udptl_read(), bridge_read(), conf_run(), console_read(), create_dtmf_frame(), dahdi_read(), gtalk_rtp_read(), handle_request_invite(), handle_response_invite(), iax2_read(), jingle_rtp_read(), local_read(), mgcp_rtp_read(), multicast_rtp_read(), oh323_read(), oh323_rtp_read(), process_sdp(), sip_read(), sip_rtp_read(), skinny_rtp_read(), spandsp_fax_read(), unistim_rtp_read(), and wakeup_sub().


Generated on Mon Jun 27 16:51:14 2011 for Asterisk - The Open Source Telephony Project by  doxygen 1.4.7