Wed Aug 7 17:16:05 2019

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

Defines

#define AST_FORMAT_ADPCM   (1ULL << 5)
#define AST_FORMAT_ALAW   (1ULL << 3)
#define AST_FORMAT_AUDIO_MASK   0xFFFF0000FFFFULL
#define AST_FORMAT_FIRST_VIDEO_BIT   AST_FORMAT_H261
#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, AST_CONTROL_INCOMPLETE = 30, AST_CONTROL_UPDATE_RTP_PEER = 32
}
 

Internal control frame subtype field values.

More...
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 = 1, AST_T38_RATE_4800, AST_T38_RATE_7200, AST_T38_RATE_9600,
  AST_T38_RATE_12000, AST_T38_RATE_14400 = 0
}
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.
struct 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)
struct ast_frameast_frame_enqueue (struct ast_frame *head, struct ast_frame *f, int maxlen, int dupe)
 Appends a frame to the end of a list of frames, truncating the maximum length of the list.
void ast_frame_free (struct ast_frame *fr, int cache)
 Requests a frame to be allocated.
int ast_frame_slinear_sum (struct ast_frame *f1, struct ast_frame *f2)
 Sums two frames of audio samples.
struct ast_frameast_frdup (const struct ast_frame *fr)
 Copies a frame.
struct ast_frameast_frisolate (struct ast_frame *fr)
 Makes a frame independent of any static storage.
struct ast_format_listast_get_format_list (size_t *size)
struct ast_format_listast_get_format_list_index (int index)
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

struct ast_frame ast_null_frame

AST_Smoother



#define ast_smoother_feed(s, f)   __ast_smoother_feed(s, f, 0)
#define ast_smoother_feed_be(s, f)   __ast_smoother_feed(s, f, 0)
#define ast_smoother_feed_le(s, f)   __ast_smoother_feed(s, f, 1)
int __ast_smoother_feed (struct ast_smoother *s, struct ast_frame *f, int swap)
void ast_smoother_free (struct ast_smoother *s)
int ast_smoother_get_flags (struct ast_smoother *smoother)
struct ast_smootherast_smoother_new (int bytes)
struct ast_frameast_smoother_read (struct ast_smoother *s)
void ast_smoother_reconfigure (struct ast_smoother *s, int bytes)
 Reconfigure an existing smoother to output a different number of bytes per frame.
void ast_smoother_reset (struct ast_smoother *s, int bytes)
void ast_smoother_set_flags (struct ast_smoother *smoother, int flags)
int ast_smoother_test_flag (struct ast_smoother *s, int flag)

Detailed Description

Asterisk internal frame definitions.

Definition in file frame.h.


Define Documentation

#define AST_FORMAT_ADPCM   (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(), vox_read(), and vox_write().

#define AST_FORMAT_ALAW   (1ULL << 3)
#define AST_FORMAT_AUDIO_MASK   0xFFFF0000FFFFULL
#define AST_FORMAT_FIRST_VIDEO_BIT   AST_FORMAT_H261

Definition at line 281 of file frame.h.

Referenced by ast_openvstream().

#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)

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)
#define AST_FORMAT_G729A   (1ULL << 8)
#define AST_FORMAT_GSM   (1ULL << 1)
#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 283 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 285 of file frame.h.

Referenced by h263p_encap().

#define AST_FORMAT_H264   (1ULL << 21)

H.264 Video

Definition at line 287 of file frame.h.

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

#define AST_FORMAT_ILBC   (1ULL << 10)
#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 296 of file frame.h.

#define AST_FORMAT_MP4_VIDEO   (1ULL << 22)

MPEG4 Video

Definition at line 289 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 305 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(), audio_audiohook_write_list(), audiohook_read_frame_both(), audiohook_read_frame_single(), background_detect_exec(), bridge_request(), build_conf(), chanspy_exec(), conf_run(), 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(), get_rate_change_result(), handle_jack_audio(), handle_recordfile(), handle_speechcreate(), handle_speechrecognize(), iax_frame_wrap(), ices_exec(), is_encoder(), isAnsweringMachine(), jack_exec(), jack_hook_callback(), linear_alloc(), linear_generator(), load_module(), load_moh_classes(), local_ast_moh_start(), measurenoise(), meetme_menu_admin_extended(), mixmonitor_thread(), mp3_exec(), nbs_request(), nbs_xwrite(), NBScat_exec(), new_outgoing(), 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(), playtones_generator(), record_exec(), 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(), tonepair_generator(), transmit_audio(), wav_read(), and wav_write().

#define AST_FORMAT_SLINEAR16   (1ULL << 15)
#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(), and speex_sample().

#define AST_FORMAT_SPEEX16   (1ULL << 33)

SpeeX Wideband (16kHz) Free Compression

Definition at line 301 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 294 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 292 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 303 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 297 of file frame.h.

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

#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)

Definition at line 616 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 615 of file frame.h.

Referenced by phone_read().

#define AST_FRAME_DTMF   AST_FRAME_DTMF_END
#define AST_FRAME_SET_BUFFER ( fr,
_base,
_ofs,
_datalen   ) 
Value:
{              \
   (fr)->data.ptr = (char *)_base + (_ofs);  \
   (fr)->offset = (_ofs);        \
   (fr)->datalen = (_datalen);      \
   }

Set the various field of a frame to point to a buffer. Typically you set the base address of the buffer, the offset as AST_FRIENDLY_OFFSET, and the datalen as the amount of bytes queued. The remaining things (to be done manually) is set the number of samples, which cannot be derived from the datalen unless you know the number of bits per sample.

Definition at line 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 583 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_dsp_process(), ast_framehook_attach(), ast_generic_bridge(), 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_agi_read_frame(), async_wait(), audio_audiohook_write_list(), autoservice_run(), background_detect_exec(), bridge_handle_dtmf(), calc_cost(), channel_spy(), check_bridge(), conf_flush(), conf_free(), conf_run(), create_jb(), dahdi_bridge(), dahdi_read(), dial_exec_full(), dictate_exec(), disa_exec(), disable_t38(), 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(), iax2_bridge(), ices_exec(), isAnsweringMachine(), jack_exec(), jb_empty_and_reset_adaptive(), jb_empty_and_reset_fixed(), jb_get_and_deliver(), local_bridge_loop(), manage_parked_call(), measurenoise(), moh_files_generator(), monitor_dial(), mp3_exec(), multicast_rtp_write(), NBScat_exec(), read_frame(), receive_dtmf_digits(), receivefax_t38_init(), record_exec(), recordthread(), remote_bridge_loop(), run_agi(), send_tone_burst(), send_waveform_to_channel(), sendfax_t38_init(), sendurl_exec(), session_destroy(), sip_read(), sip_rtp_read(), speech_background(), spy_generate(), transmit_audio(), transmit_t38(), wait_for_answer(), wait_for_hangup(), wait_for_winner(), waitforring_exec(), and waitstream_core().

#define AST_FRIENDLY_OFFSET   64

Offset into a frame's data buffer.

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

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

Definition at line 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(), playtones_generator(), process_cn_rfc3389(), send_tone_burst(), send_waveform_to_channel(), siren14read(), siren7read(), slinear_read(), sms_generate(), spandsp_fax_read(), tonepair_generator(), 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 453 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 520 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 484 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 516 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 508 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 476 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 512 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 432 of file frame.h.

#define AST_OPTION_FLAG_ANSWER   5

Definition at line 435 of file frame.h.

#define AST_OPTION_FLAG_QUERY   4

Definition at line 434 of file frame.h.

#define AST_OPTION_FLAG_REJECT   2

Definition at line 433 of file frame.h.

#define AST_OPTION_FLAG_REQUEST   0

Definition at line 431 of file frame.h.

Referenced by ast_bridge_call(), and iax2_setoption().

#define AST_OPTION_FLAG_WTF   6

Definition at line 436 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 494 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 498 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 504 of file frame.h.

Referenced by ast_channel_make_compatible_helper(), and sip_setoption().

#define AST_OPTION_OPRMODE   7

Definition at line 469 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 449 of file frame.h.

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

#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 463 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
#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 524 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 490 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 445 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 441 of file frame.h.

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

#define AST_OPTION_TXGAIN   5

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

Definition at line 458 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 686 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 691 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 692 of file frame.h.

#define AST_SMOOTHER_FLAG_BE   (1 << 1)

Definition at line 428 of file frame.h.

Referenced by ast_rtp_write().

#define AST_SMOOTHER_FLAG_G729   (1 << 0)

Definition at line 427 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 };

Internal control frame subtype field values.

Warning:
IAX2 sends these values out over the wire. To prevent future incompatibilities, pick the next value in the enum from whatever is on the current trunk. If you lose the merge race you need to fix the previous branches to match what is on trunk. In addition you need to change chan_iax2 to explicitly allow the control frame over the wire if it makes sense for the frame to be passed to another Asterisk instance.
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 

T38 state change request/notification with parameters

AST_CONTROL_CC 

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.

AST_CONTROL_INCOMPLETE 

Indication that the extension dialed is incomplete

AST_CONTROL_UPDATE_RTP_PEER 

Interrupt the bridge and have it update the peer

Definition at line 319 of file frame.h.

00319                             {
00320    AST_CONTROL_HANGUP = 1,       /*!< Other end has hungup */
00321    AST_CONTROL_RING = 2,         /*!< Local ring */
00322    AST_CONTROL_RINGING = 3,      /*!< Remote end is ringing */
00323    AST_CONTROL_ANSWER = 4,       /*!< Remote end has answered */
00324    AST_CONTROL_BUSY = 5,         /*!< Remote end is busy */
00325    AST_CONTROL_TAKEOFFHOOK = 6,  /*!< Make it go off hook */
00326    AST_CONTROL_OFFHOOK = 7,      /*!< Line is off hook */
00327    AST_CONTROL_CONGESTION = 8,      /*!< Congestion (circuits busy) */
00328    AST_CONTROL_FLASH = 9,        /*!< Flash hook */
00329    AST_CONTROL_WINK = 10,        /*!< Wink */
00330    AST_CONTROL_OPTION = 11,      /*!< Set a low-level option */
00331    AST_CONTROL_RADIO_KEY = 12,      /*!< Key Radio */
00332    AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */
00333    AST_CONTROL_PROGRESS = 14,    /*!< Indicate PROGRESS */
00334    AST_CONTROL_PROCEEDING = 15,  /*!< Indicate CALL PROCEEDING */
00335    AST_CONTROL_HOLD = 16,        /*!< Indicate call is placed on hold */
00336    AST_CONTROL_UNHOLD = 17,      /*!< Indicate call is left from hold */
00337    AST_CONTROL_VIDUPDATE = 18,      /*!< Indicate video frame update */
00338    _XXX_AST_CONTROL_T38 = 19,    /*!< T38 state change request/notification \deprecated This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead. */
00339    AST_CONTROL_SRCUPDATE = 20,      /*!< Indicate source of media has changed */
00340    AST_CONTROL_TRANSFER = 21,    /*!< Indicate status of a transfer request */
00341    AST_CONTROL_CONNECTED_LINE = 22,/*!< Indicate connected line has changed */
00342    AST_CONTROL_REDIRECTING = 23, /*!< Indicate redirecting id has changed */
00343    AST_CONTROL_T38_PARAMETERS = 24,/*!< T38 state change request/notification with parameters */
00344    AST_CONTROL_CC = 25,       /*!< Indication that Call completion service is possible */
00345    AST_CONTROL_SRCCHANGE = 26,      /*!< Media source has changed and requires a new RTP SSRC */
00346    AST_CONTROL_READ_ACTION = 27, /*!< Tell ast_read to take a specific action */
00347    AST_CONTROL_AOC = 28,         /*!< Advice of Charge with encoded generic AOC payload */
00348    AST_CONTROL_END_OF_Q = 29,    /*!< Indicate that this position was the end of the channel queue for a softhangup. */
00349    AST_CONTROL_INCOMPLETE = 30,  /*!< Indication that the extension dialed is incomplete */
00350    AST_CONTROL_UPDATE_RTP_PEER = 32, /*!< Interrupt the bridge and have it update the peer */
00351 
00352    /*
00353     * WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
00354     *
00355     * IAX2 sends these values out over the wire.  To prevent future
00356     * incompatibilities, pick the next value in the enum from whatever
00357     * is on the current trunk.  If you lose the merge race you need to
00358     * fix the previous branches to match what is on trunk.  In addition
00359     * you need to change chan_iax2 to explicitly allow the control
00360     * frame over the wire if it makes sense for the frame to be passed
00361     * to another Asterisk instance.
00362     *
00363     * WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
00364     */
00365 };

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 384 of file frame.h.

00384                      {
00385    AST_T38_REQUEST_NEGOTIATE = 1,   /*!< Request T38 on a channel (voice to fax) */
00386    AST_T38_REQUEST_TERMINATE, /*!< Terminate T38 on a channel (fax to voice) */
00387    AST_T38_NEGOTIATED,     /*!< T38 negotiated (fax mode) */
00388    AST_T38_TERMINATED,     /*!< T38 terminated (back to voice) */
00389    AST_T38_REFUSED,     /*!< T38 refused for some reason (usually rejected by remote end) */
00390    AST_T38_REQUEST_PARMS,     /*!< request far end T.38 parameters for a channel in 'negotiating' state */
00391 };

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 393 of file frame.h.

00393                           {
00394    AST_T38_RATE_2400 = 1,
00395    AST_T38_RATE_4800,
00396    AST_T38_RATE_7200,
00397    AST_T38_RATE_9600,
00398    AST_T38_RATE_12000,
00399    /* Set to 0 so it's taken as default when unspecified.
00400     * See ITU-T T.38 Implementors' Guide (11 May 2012),
00401     * Table H.2: if the T38MaxBitRate attribute is omitted
00402     * it should use a default of 14400. */
00403    AST_T38_RATE_14400 = 0,
00404 };

Enumerator:
AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF 
AST_T38_RATE_MANAGEMENT_LOCAL_TCF 

Definition at line 406 of file frame.h.

Enumerator:
AST_TRANSFER_SUCCESS 

Transfer request on the channel worked

AST_TRANSFER_FAILED 

Transfer request on the channel failed

Definition at line 422 of file frame.h.

00422                           {
00423    AST_TRANSFER_SUCCESS = 0, /*!< Transfer request on the channel worked */
00424    AST_TRANSFER_FAILED,      /*!< Transfer request on the channel failed */
00425 };

Enumerator:
AST_FRAME_READ_ACTION_CONNECTED_LINE_MACRO 

Definition at line 367 of file frame.h.

Frame types.

Note:
It is important that the values of each frame type are never changed, because it will break backwards compatability with older versions. This is because these constants are transmitted directly over IAX2.
Enumerator:
AST_FRAME_DTMF_END 

DTMF end event, subclass is the digit

AST_FRAME_VOICE 

Voice data, subclass is AST_FORMAT_*

AST_FRAME_VIDEO 

Video frame, maybe?? :)

AST_FRAME_CONTROL 

A control frame, subclass is AST_CONTROL_*

AST_FRAME_NULL 

An empty, useless frame

AST_FRAME_IAX 

Inter Asterisk Exchange private frame type

AST_FRAME_TEXT 

Text messages

AST_FRAME_IMAGE 

Image Frames

AST_FRAME_HTML 

HTML Frame

AST_FRAME_CNG 

Comfort Noise frame (subclass is level of CNG in -dBov), body may include zero or more 8-bit quantization coefficients

AST_FRAME_MODEM 

Modem-over-IP data streams

AST_FRAME_DTMF_BEGIN 

DTMF begin event, subclass is the digit

Definition at line 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 208 of file frame.c.

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

00209 {
00210    if (f->frametype != AST_FRAME_VOICE) {
00211       ast_log(LOG_WARNING, "Huh?  Can't smooth a non-voice frame!\n");
00212       return -1;
00213    }
00214    if (!s->format) {
00215       s->format = f->subclass.codec;
00216       s->samplesperbyte = (float)f->samples / (float)f->datalen;
00217    } else if (s->format != f->subclass.codec) {
00218       ast_log(LOG_WARNING, "Smoother was working on %s format frames, now trying to feed %s?\n",
00219          ast_getformatname(s->format), ast_getformatname(f->subclass.codec));
00220       return -1;
00221    }
00222    if (s->len + f->datalen > SMOOTHER_SIZE) {
00223       ast_log(LOG_WARNING, "Out of smoother space\n");
00224       return -1;
00225    }
00226    if (((f->datalen == s->size) ||
00227         ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) &&
00228        !s->opt &&
00229        !s->len &&
00230        (f->offset >= AST_MIN_OFFSET)) {
00231       /* Optimize by sending the frame we just got
00232          on the next read, thus eliminating the douple
00233          copy */
00234       if (swap)
00235          ast_swapcopy_samples(f->data.ptr, f->data.ptr, f->samples);
00236       s->opt = f;
00237       s->opt_needs_swap = swap ? 1 : 0;
00238       return 0;
00239    }
00240 
00241    return smoother_frame_feed(s, f, swap);
00242 }

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 660 of file frame.c.

References ARRAY_LEN, and ast_format_list::desc.

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

00661 {
00662    int x;
00663    char *ret = "unknown";
00664    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00665       if (AST_FORMAT_LIST[x].bits == codec) {
00666          ret = AST_FORMAT_LIST[x].desc;
00667          break;
00668       }
00669    }
00670    return ret;
00671 }

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 1249 of file frame.c.

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

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

01250 {
01251    int x, slot;
01252    format_t ret = 0;
01253 
01254    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01255       slot = pref->order[x];
01256 
01257       if (!slot)
01258          break;
01259       if (formats & AST_FORMAT_LIST[slot-1].bits) {
01260          ret = AST_FORMAT_LIST[slot-1].bits;
01261          break;
01262       }
01263    }
01264    if (ret & AST_FORMAT_AUDIO_MASK)
01265       return ret;
01266 
01267    ast_debug(4, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec");
01268 
01269       return find_best ? ast_best_codec(formats) : 0;
01270 }

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 1532 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().

01533 {
01534    int len = 0;
01535 
01536    /* XXX Still need speex, and lpc10 XXX */ 
01537    switch(format) {
01538    case AST_FORMAT_G723_1:
01539       len = (samples / 240) * 20;
01540       break;
01541    case AST_FORMAT_ILBC:
01542       len = (samples / 240) * 50;
01543       break;
01544    case AST_FORMAT_GSM:
01545       len = (samples / 160) * 33;
01546       break;
01547    case AST_FORMAT_G729A:
01548       len = samples / 8;
01549       break;
01550    case AST_FORMAT_SLINEAR:
01551    case AST_FORMAT_SLINEAR16:
01552       len = samples * 2;
01553       break;
01554    case AST_FORMAT_ULAW:
01555    case AST_FORMAT_ALAW:
01556    case AST_FORMAT_TESTLAW:
01557       len = samples;
01558       break;
01559    case AST_FORMAT_G722:
01560    case AST_FORMAT_ADPCM:
01561    case AST_FORMAT_G726:
01562    case AST_FORMAT_G726_AAL2:
01563       len = samples / 2;
01564       break;
01565    case AST_FORMAT_SIREN7:
01566       /* 16,000 samples per second at 32kbps is 4,000 bytes per second */
01567       len = samples / (16000 / 4000);
01568       break;
01569    case AST_FORMAT_SIREN14:
01570       /* 32,000 samples per second at 48kbps is 6,000 bytes per second */
01571       len = (int) samples / ((float) 32000 / 6000);
01572       break;
01573    case AST_FORMAT_G719:
01574       /* 48,000 samples per second at 64kbps is 8,000 bytes per second */
01575       len = (int) samples / ((float) 48000 / 8000);
01576       break;
01577    default:
01578       ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format));
01579    }
01580 
01581    return len;
01582 }

int ast_codec_get_samples ( struct ast_frame f  ) 

Returns the number of samples contained in the frame.

Definition at line 1470 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(), ast_frame_subclass::codec, ast_frame::data, ast_frame::datalen, g723_samples(), LOG_WARNING, ast_frame::ptr, speex_samples(), and ast_frame::subclass.

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

01471 {
01472    int samples = 0;
01473    char tmp[64];
01474 
01475    switch (f->subclass.codec) {
01476    case AST_FORMAT_SPEEX:
01477       samples = speex_samples(f->data.ptr, f->datalen);
01478       break;
01479    case AST_FORMAT_SPEEX16:
01480       samples = 2 * speex_samples(f->data.ptr, f->datalen);
01481       break;
01482    case AST_FORMAT_G723_1:
01483       samples = g723_samples(f->data.ptr, f->datalen);
01484       break;
01485    case AST_FORMAT_ILBC:
01486       samples = 240 * (f->datalen / 50);
01487       break;
01488    case AST_FORMAT_GSM:
01489       samples = 160 * (f->datalen / 33);
01490       break;
01491    case AST_FORMAT_G729A:
01492       samples = f->datalen * 8;
01493       break;
01494    case AST_FORMAT_SLINEAR:
01495    case AST_FORMAT_SLINEAR16:
01496       samples = f->datalen / 2;
01497       break;
01498    case AST_FORMAT_LPC10:
01499       /* assumes that the RTP packet contains one LPC10 frame */
01500       samples = 22 * 8;
01501       samples += (((char *)(f->data.ptr))[7] & 0x1) * 8;
01502       break;
01503    case AST_FORMAT_ULAW:
01504    case AST_FORMAT_ALAW:
01505    case AST_FORMAT_TESTLAW:
01506       samples = f->datalen;
01507       break;
01508    case AST_FORMAT_G722:
01509    case AST_FORMAT_ADPCM:
01510    case AST_FORMAT_G726:
01511    case AST_FORMAT_G726_AAL2:
01512       samples = f->datalen * 2;
01513       break;
01514    case AST_FORMAT_SIREN7:
01515       /* 16,000 samples per second at 32kbps is 4,000 bytes per second */
01516       samples = f->datalen * (16000 / 4000);
01517       break;
01518    case AST_FORMAT_SIREN14:
01519       /* 32,000 samples per second at 48kbps is 6,000 bytes per second */
01520       samples = (int) f->datalen * ((float) 32000 / 6000);
01521       break;
01522    case AST_FORMAT_G719:
01523       /* 48,000 samples per second at 64kbps is 8,000 bytes per second */
01524       samples = (int) f->datalen * ((float) 48000 / 8000);
01525       break;
01526    default:
01527       ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), f->subclass.codec));
01528    }
01529    return samples;
01530 }

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

Gets duration in ms of interpolation frame for a format.

Definition at line 782 of file frame.h.

References AST_FORMAT_ILBC.

Referenced by __get_from_jb(), and jb_get_and_deliver().

00783 { 
00784    return (format == AST_FORMAT_ILBC) ? 30 : 20;
00785 }

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 1099 of file frame.c.

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

Referenced by ast_parse_allow_disallow().

01100 {
01101    int x, newindex = 0;
01102 
01103    ast_codec_pref_remove(pref, format);
01104 
01105    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01106       if (AST_FORMAT_LIST[x].bits == format) {
01107          newindex = x + 1;
01108          break;
01109       }
01110    }
01111 
01112    if (newindex) {
01113       for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01114          if (!pref->order[x]) {
01115             pref->order[x] = newindex;
01116             break;
01117          }
01118       }
01119    }
01120 
01121    return x;
01122 }

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 1002 of file frame.c.

References ast_codec_pref::order.

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

01003 {
01004    int x, differential = (int) 'A', mem;
01005    char *from, *to;
01006 
01007    if (right) {
01008       from = pref->order;
01009       to = buf;
01010       mem = size;
01011    } else {
01012       to = pref->order;
01013       from = buf;
01014       mem = sizeof(format_t) * 8;
01015    }
01016 
01017    memset(to, 0, mem);
01018    for (x = 0; x < sizeof(format_t) * 8; x++) {
01019       if (!from[x])
01020          break;
01021       to[x] = right ? (from[x] + differential) : (from[x] - differential);
01022    }
01023 }

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

Get packet size for codec.

Definition at line 1205 of file frame.c.

References ARRAY_LEN, ast_getformatname(), ast_log(), AST_LOG_WARNING, 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_instance_bridge(), ast_rtp_write(), handle_open_receive_channel_ack_message(), skinny_set_rtp_peer(), and transmit_connect().

01206 {
01207    int x, idx = -1, framems = 0;
01208    struct ast_format_list fmt = { 0, };
01209 
01210    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01211       if (AST_FORMAT_LIST[x].bits == format) {
01212          fmt = AST_FORMAT_LIST[x];
01213          idx = x;
01214          break;
01215       }
01216    }
01217 
01218    if (idx < 0) {
01219       ast_log(AST_LOG_WARNING, "Format %s unknown; unable to get preferred codec packet size\n", ast_getformatname(format));
01220       return fmt;
01221    }
01222 
01223    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01224       if (pref->order[x] == (idx + 1)) {
01225          framems = pref->framing[x];
01226          break;
01227       }
01228    }
01229 
01230    /* size validation */
01231    if (!framems)
01232       framems = AST_FORMAT_LIST[idx].def_ms;
01233 
01234    if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */
01235       framems -= framems % AST_FORMAT_LIST[idx].inc_ms;
01236 
01237    if (framems < AST_FORMAT_LIST[idx].min_ms)
01238       framems = AST_FORMAT_LIST[idx].min_ms;
01239 
01240    if (framems > AST_FORMAT_LIST[idx].max_ms)
01241       framems = AST_FORMAT_LIST[idx].max_ms;
01242 
01243    fmt.cur_ms = framems;
01244 
01245    return fmt;
01246 }

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 1061 of file frame.c.

References 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().

01062 {
01063    int slot = 0;
01064 
01065    if ((idx >= 0) && (idx < sizeof(pref->order))) {
01066       slot = pref->order[idx];
01067    }
01068 
01069    return slot ? AST_FORMAT_LIST[slot - 1].bits : 0;
01070 }

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 1125 of file frame.c.

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

Referenced by create_addr().

01126 {
01127    int x, newindex = 0;
01128 
01129    /* First step is to get the codecs "index number" */
01130    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01131       if (AST_FORMAT_LIST[x].bits == format) {
01132          newindex = x + 1;
01133          break;
01134       }
01135    }
01136    /* Done if its unknown */
01137    if (!newindex)
01138       return;
01139 
01140    /* Now find any existing occurrence, or the end */
01141    for (x = 0; x < sizeof(format_t) * 8; x++) {
01142       if (!pref->order[x] || pref->order[x] == newindex)
01143          break;
01144    }
01145 
01146    /* If we failed to find any occurrence, set to the end */
01147    if (x == sizeof(format_t) * 8) {
01148       --x;
01149    }
01150 
01151    if (only_if_existing && !pref->order[x])
01152       return;
01153 
01154    /* Move down to make space to insert - either all the way to the end,
01155       or as far as the existing location (which will be overwritten) */
01156    for (; x > 0; x--) {
01157       pref->order[x] = pref->order[x - 1];
01158       pref->framing[x] = pref->framing[x - 1];
01159    }
01160 
01161    /* And insert the new entry */
01162    pref->order[0] = newindex;
01163    pref->framing[0] = 0; /* ? */
01164 }

void ast_codec_pref_remove ( struct ast_codec_pref pref,
format_t  format 
)

Remove audio a codec from a preference list.

Definition at line 1073 of file frame.c.

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

Referenced by ast_codec_pref_append(), and ast_parse_allow_disallow().

01074 {
01075    struct ast_codec_pref oldorder;
01076    int x, y = 0;
01077    int slot;
01078    int size;
01079 
01080    if (!pref->order[0])
01081       return;
01082 
01083    memcpy(&oldorder, pref, sizeof(oldorder));
01084    memset(pref, 0, sizeof(*pref));
01085 
01086    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01087       slot = oldorder.order[x];
01088       size = oldorder.framing[x];
01089       if (! slot)
01090          break;
01091       if (AST_FORMAT_LIST[slot-1].bits != format) {
01092          pref->order[y] = slot;
01093          pref->framing[y++] = size;
01094       }
01095    }
01096 }

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

Set packet size for codec.

Definition at line 1167 of file frame.c.

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

Referenced by ast_parse_allow_disallow(), and process_sdp_a_audio().

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

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 1025 of file frame.c.

References ast_codec_pref_index(), and ast_getformatname().

Referenced by dump_prefs(), and socket_process().

01026 {
01027    int x;
01028    format_t codec; 
01029    size_t total_len, slen;
01030    char *formatname;
01031    
01032    memset(buf, 0, size);
01033    total_len = size;
01034    buf[0] = '(';
01035    total_len--;
01036    for (x = 0; x < sizeof(format_t) * 8; x++) {
01037       if (total_len <= 0)
01038          break;
01039       if (!(codec = ast_codec_pref_index(pref,x)))
01040          break;
01041       if ((formatname = ast_getformatname(codec))) {
01042          slen = strlen(formatname);
01043          if (slen > total_len)
01044             break;
01045          strncat(buf, formatname, total_len - 1); /* safe */
01046          total_len -= slen;
01047       }
01048       if (total_len && x < sizeof(format_t) * 8 - 1 && ast_codec_pref_index(pref, x + 1)) {
01049          strncat(buf, "|", total_len - 1); /* safe */
01050          total_len--;
01051       }
01052    }
01053    if (total_len) {
01054       strncat(buf, ")", total_len - 1); /* safe */
01055       total_len--;
01056    }
01057 
01058    return size - total_len;
01059 }

static force_inline int ast_format_rate ( format_t  format  )  [static]
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 1584 of file frame.c.

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

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

01585 {
01586    int count;
01587    short *fdata = f->data.ptr;
01588    short adjust_value = abs(adjustment);
01589 
01590    if ((f->frametype != AST_FRAME_VOICE) || (f->subclass.codec != AST_FORMAT_SLINEAR))
01591       return -1;
01592 
01593    if (!adjustment)
01594       return 0;
01595 
01596    for (count = 0; count < f->samples; count++) {
01597       if (adjustment > 0) {
01598          ast_slinear_saturated_multiply(&fdata[count], &adjust_value);
01599       } else if (adjustment < 0) {
01600          ast_slinear_saturated_divide(&fdata[count], &adjust_value);
01601       }
01602    }
01603 
01604    return 0;
01605 }

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 1629 of file frame.c.

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

Referenced by ast_audiohook_write_frame(), and mute_callback().

01630 {
01631    struct ast_frame *next;
01632 
01633    for (next = AST_LIST_NEXT(frame, frame_list);
01634        frame;
01635        frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) {
01636       memset(frame->data.ptr, 0, frame->datalen);
01637    }
01638    return 0;
01639 }

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

Dump a frame for debugging purposes

Definition at line 778 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, ast_frame_subclass::codec, COLOR_BLACK, COLOR_BRCYAN, COLOR_BRGREEN, COLOR_BRMAGENTA, COLOR_BRRED, COLOR_YELLOW, ast_frame::data, ast_frame::datalen, ast_frame::frametype, ast_frame_subclass::integer, ast_frame::ptr, ast_control_t38_parameters::request_response, ast_frame::subclass, and term_color().

Referenced by __ast_read(), and ast_write().

00779 {
00780    const char noname[] = "unknown";
00781    char ftype[40] = "Unknown Frametype";
00782    char cft[80];
00783    char subclass[40] = "Unknown Subclass";
00784    char csub[80];
00785    char moreinfo[40] = "";
00786    char cn[60];
00787    char cp[40];
00788    char cmn[40];
00789    const char *message = "Unknown";
00790 
00791    if (!name)
00792       name = noname;
00793 
00794 
00795    if (!f) {
00796       ast_verbose("%s [ %s (NULL) ] [%s]\n", 
00797          term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00798          term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 
00799          term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00800       return;
00801    }
00802    /* XXX We should probably print one each of voice and video when the format changes XXX */
00803    if (f->frametype == AST_FRAME_VOICE)
00804       return;
00805    if (f->frametype == AST_FRAME_VIDEO)
00806       return;
00807    switch(f->frametype) {
00808    case AST_FRAME_DTMF_BEGIN:
00809       strcpy(ftype, "DTMF Begin");
00810       subclass[0] = f->subclass.integer;
00811       subclass[1] = '\0';
00812       break;
00813    case AST_FRAME_DTMF_END:
00814       strcpy(ftype, "DTMF End");
00815       subclass[0] = f->subclass.integer;
00816       subclass[1] = '\0';
00817       break;
00818    case AST_FRAME_CONTROL:
00819       strcpy(ftype, "Control");
00820       switch (f->subclass.integer) {
00821       case AST_CONTROL_HANGUP:
00822          strcpy(subclass, "Hangup");
00823          break;
00824       case AST_CONTROL_RING:
00825          strcpy(subclass, "Ring");
00826          break;
00827       case AST_CONTROL_RINGING:
00828          strcpy(subclass, "Ringing");
00829          break;
00830       case AST_CONTROL_ANSWER:
00831          strcpy(subclass, "Answer");
00832          break;
00833       case AST_CONTROL_BUSY:
00834          strcpy(subclass, "Busy");
00835          break;
00836       case AST_CONTROL_TAKEOFFHOOK:
00837          strcpy(subclass, "Take Off Hook");
00838          break;
00839       case AST_CONTROL_OFFHOOK:
00840          strcpy(subclass, "Line Off Hook");
00841          break;
00842       case AST_CONTROL_CONGESTION:
00843          strcpy(subclass, "Congestion");
00844          break;
00845       case AST_CONTROL_FLASH:
00846          strcpy(subclass, "Flash");
00847          break;
00848       case AST_CONTROL_WINK:
00849          strcpy(subclass, "Wink");
00850          break;
00851       case AST_CONTROL_OPTION:
00852          strcpy(subclass, "Option");
00853          break;
00854       case AST_CONTROL_RADIO_KEY:
00855          strcpy(subclass, "Key Radio");
00856          break;
00857       case AST_CONTROL_RADIO_UNKEY:
00858          strcpy(subclass, "Unkey Radio");
00859          break;
00860       case AST_CONTROL_HOLD:
00861          strcpy(subclass, "Hold");
00862          break;
00863       case AST_CONTROL_UNHOLD:
00864          strcpy(subclass, "Unhold");
00865          break;
00866       case AST_CONTROL_T38_PARAMETERS:
00867          if (f->datalen != sizeof(struct ast_control_t38_parameters)) {
00868             message = "Invalid";
00869          } else {
00870             struct ast_control_t38_parameters *parameters = f->data.ptr;
00871             enum ast_control_t38 state = parameters->request_response;
00872             if (state == AST_T38_REQUEST_NEGOTIATE)
00873                message = "Negotiation Requested";
00874             else if (state == AST_T38_REQUEST_TERMINATE)
00875                message = "Negotiation Request Terminated";
00876             else if (state == AST_T38_NEGOTIATED)
00877                message = "Negotiated";
00878             else if (state == AST_T38_TERMINATED)
00879                message = "Terminated";
00880             else if (state == AST_T38_REFUSED)
00881                message = "Refused";
00882          }
00883          snprintf(subclass, sizeof(subclass), "T38_Parameters/%s", message);
00884          break;
00885       case -1:
00886          strcpy(subclass, "Stop generators");
00887          break;
00888       default:
00889          snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass.integer);
00890       }
00891       break;
00892    case AST_FRAME_NULL:
00893       strcpy(ftype, "Null Frame");
00894       strcpy(subclass, "N/A");
00895       break;
00896    case AST_FRAME_IAX:
00897       /* Should never happen */
00898       strcpy(ftype, "IAX Specific");
00899       snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass.integer);
00900       break;
00901    case AST_FRAME_TEXT:
00902       strcpy(ftype, "Text");
00903       strcpy(subclass, "N/A");
00904       ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo));
00905       break;
00906    case AST_FRAME_IMAGE:
00907       strcpy(ftype, "Image");
00908       snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass.codec));
00909       break;
00910    case AST_FRAME_HTML:
00911       strcpy(ftype, "HTML");
00912       switch (f->subclass.integer) {
00913       case AST_HTML_URL:
00914          strcpy(subclass, "URL");
00915          ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo));
00916          break;
00917       case AST_HTML_DATA:
00918          strcpy(subclass, "Data");
00919          break;
00920       case AST_HTML_BEGIN:
00921          strcpy(subclass, "Begin");
00922          break;
00923       case AST_HTML_END:
00924          strcpy(subclass, "End");
00925          break;
00926       case AST_HTML_LDCOMPLETE:
00927          strcpy(subclass, "Load Complete");
00928          break;
00929       case AST_HTML_NOSUPPORT:
00930          strcpy(subclass, "No Support");
00931          break;
00932       case AST_HTML_LINKURL:
00933          strcpy(subclass, "Link URL");
00934          ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo));
00935          break;
00936       case AST_HTML_UNLINK:
00937          strcpy(subclass, "Unlink");
00938          break;
00939       case AST_HTML_LINKREJECT:
00940          strcpy(subclass, "Link Reject");
00941          break;
00942       default:
00943          snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass.integer);
00944          break;
00945       }
00946       break;
00947    case AST_FRAME_MODEM:
00948       strcpy(ftype, "Modem");
00949       switch (f->subclass.integer) {
00950       case AST_MODEM_T38:
00951          strcpy(subclass, "T.38");
00952          break;
00953       case AST_MODEM_V150:
00954          strcpy(subclass, "V.150");
00955          break;
00956       default:
00957          snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass.integer);
00958          break;
00959       }
00960       break;
00961    default:
00962       snprintf(ftype, sizeof(ftype), "Unknown Frametype '%u'", f->frametype);
00963    }
00964    if (!ast_strlen_zero(moreinfo))
00965       ast_verbose("%s [ TYPE: %s (%u) SUBCLASS: %s (%d) '%s' ] [%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(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)),
00972              term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00973    else
00974       ast_verbose("%s [ TYPE: %s (%u) SUBCLASS: %s (%d) ] [%s]\n",  
00975              term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00976              term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)),
00977              f->frametype, 
00978              term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)),
00979              f->subclass.integer, 
00980              term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00981 }

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

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

void ast_frame_free ( struct ast_frame fr,
int  cache 
)

Requests a frame to be allocated.

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

Frees a frame or list of frames

Parameters:
fr Frame to free, or head of list to free
cache Whether to consider this frame for frame caching

Definition at line 375 of file frame.c.

References __frame_free(), and AST_LIST_NEXT.

Referenced by mixmonitor_thread().

00376 {
00377    struct ast_frame *next;
00378 
00379    for (next = AST_LIST_NEXT(frame, frame_list);
00380         frame;
00381         frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) {
00382       __frame_free(frame, cache);
00383    }
00384 }

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 1607 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.

01608 {
01609    int count;
01610    short *data1, *data2;
01611 
01612    if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass.codec != AST_FORMAT_SLINEAR))
01613       return -1;
01614 
01615    if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass.codec != AST_FORMAT_SLINEAR))
01616       return -1;
01617 
01618    if (f1->samples != f2->samples)
01619       return -1;
01620 
01621    for (count = 0, data1 = f1->data.ptr, data2 = f2->data.ptr;
01622         count < f1->samples;
01623         count++, data1++, data2++)
01624       ast_slinear_saturated_add(data1, data2);
01625 
01626    return 0;
01627 }

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

Copies a frame.

Parameters:
fr frame to copy Duplicates a frame -- should only rarely be used, typically frisolate is good enough
Returns:
Returns a frame on success, NULL on error

Definition at line 474 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, frame_cache, frames, ast_frame::frametype, ast_frame::len, len(), ast_frame_cache::list, ast_frame::mallocd, ast_frame::mallocd_hdr_len, ast_frame::offset, ast_frame::ptr, ast_frame::samples, ast_frame::seqno, ast_frame_cache::size, ast_frame::src, ast_frame::subclass, ast_frame::ts, and ast_frame::uint32.

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

00475 {
00476    struct ast_frame *out = NULL;
00477    int len, srclen = 0;
00478    void *buf = NULL;
00479 
00480 #if !defined(LOW_MEMORY)
00481    struct ast_frame_cache *frames;
00482 #endif
00483 
00484    /* Start with standard stuff */
00485    len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen;
00486    /* If we have a source, add space for it */
00487    /*
00488     * XXX Watch out here - if we receive a src which is not terminated
00489     * properly, we can be easily attacked. Should limit the size we deal with.
00490     */
00491    if (f->src)
00492       srclen = strlen(f->src);
00493    if (srclen > 0)
00494       len += srclen + 1;
00495    
00496 #if !defined(LOW_MEMORY)
00497    if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) {
00498       AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) {
00499          if (out->mallocd_hdr_len >= len) {
00500             size_t mallocd_len = out->mallocd_hdr_len;
00501 
00502             AST_LIST_REMOVE_CURRENT(frame_list);
00503             memset(out, 0, sizeof(*out));
00504             out->mallocd_hdr_len = mallocd_len;
00505             buf = out;
00506             frames->size--;
00507             break;
00508          }
00509       }
00510       AST_LIST_TRAVERSE_SAFE_END;
00511    }
00512 #endif
00513 
00514    if (!buf) {
00515       if (!(buf = ast_calloc_cache(1, len)))
00516          return NULL;
00517       out = buf;
00518       out->mallocd_hdr_len = len;
00519    }
00520 
00521    out->frametype = f->frametype;
00522    out->subclass.codec = f->subclass.codec;
00523    out->datalen = f->datalen;
00524    out->samples = f->samples;
00525    out->delivery = f->delivery;
00526    /* Even though this new frame was allocated from the heap, we can't mark it
00527     * with AST_MALLOCD_HDR, AST_MALLOCD_DATA and AST_MALLOCD_SRC, because that
00528     * would cause ast_frfree() to attempt to individually free each of those
00529     * under the assumption that they were separately allocated. Since this frame
00530     * was allocated in a single allocation, we'll only mark it as if the header
00531     * was heap-allocated; this will result in the entire frame being properly freed.
00532     */
00533    out->mallocd = AST_MALLOCD_HDR;
00534    out->offset = AST_FRIENDLY_OFFSET;
00535    if (out->datalen) {
00536       out->data.ptr = buf + sizeof(*out) + AST_FRIENDLY_OFFSET;
00537       memcpy(out->data.ptr, f->data.ptr, out->datalen);
00538    } else {
00539       out->data.uint32 = f->data.uint32;
00540    }
00541    if (srclen > 0) {
00542       /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */
00543       char *src;
00544       out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen;
00545       src = (char *) out->src;
00546       /* Must have space since we allocated for it */
00547       strcpy(src, f->src);
00548    }
00549    ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO);
00550    out->ts = f->ts;
00551    out->len = f->len;
00552    out->seqno = f->seqno;
00553    return out;
00554 }

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

Makes a frame independent of any static storage.

Parameters:
fr frame to act upon Take a frame, and if it's not been malloc'd, make a malloc'd copy and if the data hasn't been malloced then make the data malloc'd. If you need to store frames, say for queueing, then you should call this function.
Returns:
Returns a frame on success, NULL on error
Note:
This function may modify the frame passed to it, so you must not assume the frame will be intact after the isolated frame has been produced. In other words, calling this function on a frame should be the last operation you do with that frame before freeing it (or exiting the block, if the frame is on the stack.)

Definition at line 391 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_dsp_process(), 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().

00392 {
00393    struct ast_frame *out;
00394    void *newdata;
00395 
00396    /* if none of the existing frame is malloc'd, let ast_frdup() do it
00397       since it is more efficient
00398    */
00399    if (fr->mallocd == 0) {
00400       return ast_frdup(fr);
00401    }
00402 
00403    /* if everything is already malloc'd, we are done */
00404    if ((fr->mallocd & (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) ==
00405        (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) {
00406       return fr;
00407    }
00408 
00409    if (!(fr->mallocd & AST_MALLOCD_HDR)) {
00410       /* Allocate a new header if needed */
00411       if (!(out = ast_frame_header_new())) {
00412          return NULL;
00413       }
00414       out->frametype = fr->frametype;
00415       out->subclass.codec = fr->subclass.codec;
00416       out->datalen = fr->datalen;
00417       out->samples = fr->samples;
00418       out->offset = fr->offset;
00419       /* Copy the timing data */
00420       ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO);
00421       if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) {
00422          out->ts = fr->ts;
00423          out->len = fr->len;
00424          out->seqno = fr->seqno;
00425       }
00426    } else {
00427       out = fr;
00428    }
00429    
00430    if (!(fr->mallocd & AST_MALLOCD_SRC) && fr->src) {
00431       if (!(out->src = ast_strdup(fr->src))) {
00432          if (out != fr) {
00433             ast_free(out);
00434          }
00435          return NULL;
00436       }
00437    } else {
00438       out->src = fr->src;
00439       fr->src = NULL;
00440       fr->mallocd &= ~AST_MALLOCD_SRC;
00441    }
00442    
00443    if (!(fr->mallocd & AST_MALLOCD_DATA))  {
00444       if (!fr->datalen) {
00445          out->data.uint32 = fr->data.uint32;
00446          out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC;
00447          return out;
00448       }
00449       if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) {
00450          if (out->src != fr->src) {
00451             ast_free((void *) out->src);
00452          }
00453          if (out != fr) {
00454             ast_free(out);
00455          }
00456          return NULL;
00457       }
00458       newdata += AST_FRIENDLY_OFFSET;
00459       out->offset = AST_FRIENDLY_OFFSET;
00460       out->datalen = fr->datalen;
00461       memcpy(newdata, fr->data.ptr, fr->datalen);
00462       out->data.ptr = newdata;
00463    } else {
00464       out->data = fr->data;
00465       memset(&fr->data, 0, sizeof(fr->data));
00466       fr->mallocd &= ~AST_MALLOCD_DATA;
00467    }
00468 
00469    out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA;
00470    
00471    return out;
00472 }

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

Definition at line 572 of file frame.c.

References ARRAY_LEN.

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

00573 {
00574    *size = ARRAY_LEN(AST_FORMAT_LIST);
00575    return AST_FORMAT_LIST;
00576 }

struct ast_format_list* ast_get_format_list_index ( int  index  )  [read]

Definition at line 567 of file frame.c.

00568 {
00569    return &AST_FORMAT_LIST[idx];
00570 }

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 641 of file frame.c.

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

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

00642 {
00643    int x, all;
00644    format_t format = 0;
00645 
00646    all = strcasecmp(name, "all") ? 0 : 1;
00647    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00648       if (all || 
00649            !strcasecmp(AST_FORMAT_LIST[x].name,name) ||
00650            !strcasecmp(AST_FORMAT_LIST[x].name, ast_expand_codec_alias(name))) {
00651          format |= AST_FORMAT_LIST[x].bits;
00652          if (!all)
00653             break;
00654       }
00655    }
00656 
00657    return format;
00658 }

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 578 of file frame.c.

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

Referenced by __ast_play_and_record(), __ast_read(), __ast_register_translator(), __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_getsize(), 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_write(), 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(), show_codecs(), 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().

00579 {
00580    int x;
00581    char *ret = "unknown";
00582    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00583       if (AST_FORMAT_LIST[x].bits == format) {
00584          ret = AST_FORMAT_LIST[x].name;
00585          break;
00586       }
00587    }
00588    return ret;
00589 }

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 591 of file frame.c.

References ARRAY_LEN, ast_copy_string(), 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_codec_get_samples(), ast_request(), ast_streamfile(), ast_write(), bridge_make_compatible(), console_request(), 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(), iax2_bridge(), 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(), socket_process(), start_rtp(), unistim_new(), unistim_request(), and unistim_write().

00592 {
00593    int x;
00594    unsigned len;
00595    char *start, *end = buf;
00596 
00597    if (!size)
00598       return buf;
00599    snprintf(end, size, "0x%llx (", (unsigned long long) format);
00600    len = strlen(end);
00601    end += len;
00602    size -= len;
00603    start = end;
00604    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00605       if (AST_FORMAT_LIST[x].bits & format) {
00606          snprintf(end, size, "%s|", AST_FORMAT_LIST[x].name);
00607          len = strlen(end);
00608          end += len;
00609          size -= len;
00610       }
00611    }
00612    if (start == end)
00613       ast_copy_string(start, "nothing)", size);
00614    else if (size > 1)
00615       *(end - 1) = ')';
00616    return buf;
00617 }

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 1272 of file frame.c.

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

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(), reload_config(), set_config(), skinny_unregister(), and update_common_options().

01273 {
01274    int errors = 0, framems = 0;
01275    char *parse = NULL, *this = NULL, *psize = NULL;
01276    format_t format = 0;
01277 
01278    parse = ast_strdupa(list);
01279    while ((this = strsep(&parse, ","))) {
01280       framems = 0;
01281       if ((psize = strrchr(this, ':'))) {
01282          *psize++ = '\0';
01283          ast_debug(1, "Packetization for codec: %s is %s\n", this, psize);
01284          framems = atoi(psize);
01285          if (framems < 0) {
01286             framems = 0;
01287             errors++;
01288             ast_log(LOG_WARNING, "Bad packetization value for codec %s\n", this);
01289          }
01290       }
01291       if (!(format = ast_getformatbyname(this))) {
01292          ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this);
01293          errors++;
01294          continue;
01295       }
01296 
01297       if (mask) {
01298          if (allowing)
01299             *mask |= format;
01300          else
01301             *mask &= ~format;
01302       }
01303 
01304       /* Set up a preference list for audio. Do not include video in preferences 
01305          since we can not transcode video and have to use whatever is offered
01306        */
01307       if (pref && (format & AST_FORMAT_AUDIO_MASK)) {
01308          if (strcasecmp(this, "all")) {
01309             if (allowing) {
01310                ast_codec_pref_append(pref, format);
01311                ast_codec_pref_setsize(pref, format, framems);
01312             }
01313             else
01314                ast_codec_pref_remove(pref, format);
01315          } else if (!allowing) {
01316             memset(pref, 0, sizeof(*pref));
01317          }
01318       }
01319    }
01320    return errors;
01321 }

void ast_smoother_free ( struct ast_smoother s  ) 

Definition at line 294 of file frame.c.

References ast_free.

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

00295 {
00296    ast_free(s);
00297 }

int ast_smoother_get_flags ( struct ast_smoother smoother  ) 

Definition at line 193 of file frame.c.

References ast_smoother::flags.

00194 {
00195    return s->flags;
00196 }

struct ast_smoother* ast_smoother_new ( int  bytes  )  [read]

Definition at line 183 of file frame.c.

References ast_malloc, and ast_smoother_reset().

Referenced by ast_rtp_write(), and generic_fax_exec().

00184 {
00185    struct ast_smoother *s;
00186    if (size < 1)
00187       return NULL;
00188    if ((s = ast_malloc(sizeof(*s))))
00189       ast_smoother_reset(s, size);
00190    return s;
00191 }

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

Definition at line 244 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_smoother::data, ast_frame::data, ast_frame::datalen, ast_smoother::delivery, ast_frame::delivery, ast_smoother::f, ast_smoother::flags, ast_smoother::format, ast_smoother::framedata, ast_frame::frametype, ast_smoother::len, len(), LOG_WARNING, ast_frame::offset, ast_smoother::opt, ast_frame::ptr, ast_frame::samples, ast_smoother::samplesperbyte, ast_smoother::size, and ast_frame::subclass.

Referenced by ast_rtp_write(), and generic_fax_exec().

00245 {
00246    struct ast_frame *opt;
00247    int len;
00248 
00249    /* IF we have an optimization frame, send it */
00250    if (s->opt) {
00251       if (s->opt->offset < AST_FRIENDLY_OFFSET)
00252          ast_log(LOG_WARNING, "Returning a frame of inappropriate offset (%d).\n",
00253                      s->opt->offset);
00254       opt = s->opt;
00255       s->opt = NULL;
00256       return opt;
00257    }
00258 
00259    /* Make sure we have enough data */
00260    if (s->len < s->size) {
00261       /* Or, if this is a G.729 frame with VAD on it, send it immediately anyway */
00262       if (!((s->flags & AST_SMOOTHER_FLAG_G729) && (s->len % 10)))
00263          return NULL;
00264    }
00265    len = s->size;
00266    if (len > s->len)
00267       len = s->len;
00268    /* Make frame */
00269    s->f.frametype = AST_FRAME_VOICE;
00270    s->f.subclass.codec = s->format;
00271    s->f.data.ptr = s->framedata + AST_FRIENDLY_OFFSET;
00272    s->f.offset = AST_FRIENDLY_OFFSET;
00273    s->f.datalen = len;
00274    /* Samples will be improper given VAD, but with VAD the concept really doesn't even exist */
00275    s->f.samples = len * s->samplesperbyte;   /* XXX rounding */
00276    s->f.delivery = s->delivery;
00277    /* Fill Data */
00278    memcpy(s->f.data.ptr, s->data, len);
00279    s->len -= len;
00280    /* Move remaining data to the front if applicable */
00281    if (s->len) {
00282       /* In principle this should all be fine because if we are sending
00283          G.729 VAD, the next timestamp will take over anyawy */
00284       memmove(s->data, s->data + len, s->len);
00285       if (!ast_tvzero(s->delivery)) {
00286          /* If we have delivery time, increment it, otherwise, leave it at 0 */
00287          s->delivery = ast_tvadd(s->delivery, ast_samp2tv(s->f.samples, ast_format_rate(s->format)));
00288       }
00289    }
00290    /* Return frame */
00291    return &s->f;
00292 }

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 161 of file frame.c.

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

00162 {
00163    /* if there is no change, then nothing to do */
00164    if (s->size == bytes) {
00165       return;
00166    }
00167    /* set the new desired output size */
00168    s->size = bytes;
00169    /* if there is no 'optimized' frame in the smoother,
00170     *   then there is nothing left to do
00171     */
00172    if (!s->opt) {
00173       return;
00174    }
00175    /* there is an 'optimized' frame here at the old size,
00176     * but it must now be put into the buffer so the data
00177     * can be extracted at the new size
00178     */
00179    smoother_frame_feed(s, s->opt, s->opt_needs_swap);
00180    s->opt = NULL;
00181 }

void ast_smoother_reset ( struct ast_smoother s,
int  bytes 
)

Definition at line 155 of file frame.c.

References ast_smoother::size.

Referenced by ast_smoother_new().

00156 {
00157    memset(s, 0, sizeof(*s));
00158    s->size = bytes;
00159 }

void ast_smoother_set_flags ( struct ast_smoother smoother,
int  flags 
)

Definition at line 198 of file frame.c.

References ast_smoother::flags.

Referenced by ast_rtp_write().

00199 {
00200    s->flags = flags;
00201 }

int ast_smoother_test_flag ( struct ast_smoother s,
int  flag 
)

Definition at line 203 of file frame.c.

References ast_smoother::flags.

Referenced by ast_rtp_write().

00204 {
00205    return (s->flags & flag);
00206 }

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

Definition at line 556 of file frame.c.

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

00557 {
00558    int i;
00559    unsigned short *dst_s = dst;
00560    const unsigned short *src_s = src;
00561 
00562    for (i = 0; i < samples; i++)
00563       dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8);
00564 }


Variable Documentation


Generated on 7 Aug 2019 for Asterisk - The Open Source Telephony Project by  doxygen 1.6.1