frame.h File Reference

Asterisk internal frame definitions. More...

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

Include dependency graph for frame.h:

This graph shows which files directly or indirectly include this file:

Data Structures
struct	ast_codec_pref
struct	ast_control_t38_parameters
struct	ast_format_list
	Definition of supported media formats (codecs). More...
struct	ast_frame
	Data structure associated with a single frame of data. More...
struct	ast_option_header
struct	oprmode
Defines
#define	AST_FORMAT_ADPCM (1 << 5)
#define	AST_FORMAT_ALAW (1 << 3)
#define	AST_FORMAT_AUDIO_MASK ((1 << 16)-1)
#define	AST_FORMAT_G722 (1 << 12)
#define	AST_FORMAT_G723_1 (1 << 0)
#define	AST_FORMAT_G726 (1 << 11)
#define	AST_FORMAT_G726_AAL2 (1 << 4)
#define	AST_FORMAT_G729A (1 << 8)
#define	AST_FORMAT_GSM (1 << 1)
#define	AST_FORMAT_H261 (1 << 18)
#define	AST_FORMAT_H263 (1 << 19)
#define	AST_FORMAT_H263_PLUS (1 << 20)
#define	AST_FORMAT_H264 (1 << 21)
#define	AST_FORMAT_ILBC (1 << 10)
#define	AST_FORMAT_JPEG (1 << 16)
#define	AST_FORMAT_LPC10 (1 << 7)
#define	AST_FORMAT_MAX_TEXT (1 << 28))
#define	AST_FORMAT_MP4_VIDEO (1 << 22)
#define	AST_FORMAT_PNG (1 << 17)
#define	AST_FORMAT_SIREN14 (1 << 14)
#define	AST_FORMAT_SIREN7 (1 << 13)
#define	AST_FORMAT_SLINEAR (1 << 6)
#define	AST_FORMAT_SLINEAR16 (1 << 15)
#define	AST_FORMAT_SPEEX (1 << 9)
#define	AST_FORMAT_T140 (1 << 27)
#define	AST_FORMAT_T140RED (1 << 26)
#define	AST_FORMAT_TEXT_MASK (((1 << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK))
#define	AST_FORMAT_ULAW (1 << 2)
#define	AST_FORMAT_VIDEO_MASK (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK))
#define	ast_frame_byteswap_be(fr) do { ; } while(0)
#define	ast_frame_byteswap_le(fr) do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data.ptr, __f->data.ptr, __f->samples); } while(0)
#define	AST_FRAME_DTMF AST_FRAME_DTMF_END
#define	AST_FRAME_SET_BUFFER(fr, _base, _ofs, _datalen)
#define	ast_frfree(fr) ast_frame_free(fr, 1)
#define	AST_FRIENDLY_OFFSET 64
	Offset into a frame's data buffer.
#define	AST_HTML_BEGIN 4
#define	AST_HTML_DATA 2
#define	AST_HTML_END 8
#define	AST_HTML_LDCOMPLETE 16
#define	AST_HTML_LINKREJECT 20
#define	AST_HTML_LINKURL 18
#define	AST_HTML_NOSUPPORT 17
#define	AST_HTML_UNLINK 19
#define	AST_HTML_URL 1
#define	AST_MALLOCD_DATA (1 << 1)
#define	AST_MALLOCD_HDR (1 << 0)
#define	AST_MALLOCD_SRC (1 << 2)
#define	AST_MIN_OFFSET 32
#define	AST_MODEM_T38 1
#define	AST_MODEM_V150 2
#define	AST_OPTION_AUDIO_MODE 4
#define	AST_OPTION_ECHOCAN 8
#define	AST_OPTION_FLAG_ACCEPT 1
#define	AST_OPTION_FLAG_ANSWER 5
#define	AST_OPTION_FLAG_QUERY 4
#define	AST_OPTION_FLAG_REJECT 2
#define	AST_OPTION_FLAG_REQUEST 0
#define	AST_OPTION_FLAG_WTF 6
#define	AST_OPTION_OPRMODE 7
#define	AST_OPTION_RELAXDTMF 3
#define	AST_OPTION_RXGAIN 6
#define	AST_OPTION_T38_STATE 10
#define	AST_OPTION_TDD 2
#define	AST_OPTION_TONE_VERIFY 1
#define	AST_OPTION_TXGAIN 5
#define	AST_SMOOTHER_FLAG_BE (1 << 1)
#define	AST_SMOOTHER_FLAG_G729 (1 << 0)
Enumerations
enum	{ AST_FRFLAG_HAS_TIMING_INFO = (1 << 0), AST_FRFLAG_FROM_TRANSLATOR = (1 << 1), AST_FRFLAG_FROM_DSP = (1 << 2) }
enum	ast_control_frame_type { AST_CONTROL_HANGUP = 1, AST_CONTROL_RING = 2, AST_CONTROL_RINGING = 3, AST_CONTROL_ANSWER = 4, AST_CONTROL_BUSY = 5, AST_CONTROL_TAKEOFFHOOK = 6, AST_CONTROL_OFFHOOK = 7, AST_CONTROL_CONGESTION = 8, AST_CONTROL_FLASH = 9, AST_CONTROL_WINK = 10, AST_CONTROL_OPTION = 11, AST_CONTROL_RADIO_KEY = 12, AST_CONTROL_RADIO_UNKEY = 13, AST_CONTROL_PROGRESS = 14, AST_CONTROL_PROCEEDING = 15, AST_CONTROL_HOLD = 16, AST_CONTROL_UNHOLD = 17, AST_CONTROL_VIDUPDATE = 18, _XXX_AST_CONTROL_T38 = 19, AST_CONTROL_SRCUPDATE = 20, AST_CONTROL_T38_PARAMETERS = 24 }
enum	ast_control_t38 { AST_T38_REQUEST_NEGOTIATE = 1, AST_T38_REQUEST_TERMINATE, AST_T38_NEGOTIATED, AST_T38_TERMINATED, AST_T38_REFUSED }
enum	ast_control_t38_rate { AST_T38_RATE_2400 = 0, AST_T38_RATE_4800, AST_T38_RATE_7200, AST_T38_RATE_9600, AST_T38_RATE_12000, AST_T38_RATE_14400 }
enum	ast_control_t38_rate_management { AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF = 0, AST_T38_RATE_MANAGEMENT_LOCAL_TCF }
enum	ast_frame_type { AST_FRAME_DTMF_END = 1, AST_FRAME_VOICE, AST_FRAME_VIDEO, AST_FRAME_CONTROL, AST_FRAME_NULL, AST_FRAME_IAX, AST_FRAME_TEXT, AST_FRAME_IMAGE, AST_FRAME_HTML, AST_FRAME_CNG, AST_FRAME_MODEM, AST_FRAME_DTMF_BEGIN }
	Frame types. More...
Functions
char *	ast_codec2str (int codec)
	Get a name from a format Gets a name from a format.
int	ast_codec_choose (struct ast_codec_pref *pref, int formats, int find_best)
	Select the best audio format according to preference list from supplied options. If "find_best" is non-zero then if nothing is found, the "Best" format of the format list is selected, otherwise 0 is returned.
int	ast_codec_get_len (int format, int samples)
	Returns the number of bytes for the number of samples of the given format.
int	ast_codec_get_samples (struct ast_frame *f)
	Returns the number of samples contained in the frame.
static int	ast_codec_interp_len (int format)
	Gets duration in ms of interpolation frame for a format.
int	ast_codec_pref_append (struct ast_codec_pref *pref, int format)
	Append a audio codec to a preference list, removing it first if it was already there.
void	ast_codec_pref_convert (struct ast_codec_pref pref, char buf, size_t size, int right)
	Shift an audio codec preference list up or down 65 bytes so that it becomes an ASCII string.
struct ast_format_list	ast_codec_pref_getsize (struct ast_codec_pref *pref, int format)
	Get packet size for codec.
int	ast_codec_pref_index (struct ast_codec_pref *pref, int index)
	Codec located at a particular place in the preference index.
void	ast_codec_pref_init (struct ast_codec_pref *pref)
	Initialize an audio codec preference to "no preference".
void	ast_codec_pref_prepend (struct ast_codec_pref *pref, int format, int only_if_existing)
	Prepend an audio codec to a preference list, removing it first if it was already there.
void	ast_codec_pref_remove (struct ast_codec_pref *pref, int format)
	Remove audio a codec from a preference list.
int	ast_codec_pref_setsize (struct ast_codec_pref *pref, int format, int framems)
	Set packet size for codec.
int	ast_codec_pref_string (struct ast_codec_pref pref, char buf, size_t size)
	Dump audio codec preference list into a string.
static force_inline int	ast_format_rate (int format)
	Get the sample rate for a given format.
int	ast_frame_adjust_volume (struct ast_frame *f, int adjustment)
	Adjusts the volume of the audio samples contained in a frame.
void	ast_frame_dump (const char name, struct ast_frame f, char *prefix)
struct ast_frame *	ast_frame_enqueue (struct ast_frame head, struct ast_frame f, int maxlen, int dupe)
	Appends a frame to the end of a list of frames, truncating the maximum length of the list.
void	ast_frame_free (struct ast_frame *fr, int cache)
	Requests a frame to be allocated.
int	ast_frame_slinear_sum (struct ast_frame f1, struct ast_frame f2)
	Sums two frames of audio samples.
struct ast_frame *	ast_frdup (const struct ast_frame *fr)
	Copies a frame.
struct ast_frame *	ast_frisolate (struct ast_frame *fr)
	Makes a frame independent of any static storage.
struct ast_format_list *	ast_get_format_list (size_t *size)
struct ast_format_list *	ast_get_format_list_index (int index)
int	ast_getformatbyname (const char *name)
	Gets a format from a name.
char *	ast_getformatname (int format)
	Get the name of a format.
char *	ast_getformatname_multiple (char *buf, size_t size, int format)
	Get the names of a set of formats.
int	ast_parse_allow_disallow (struct ast_codec_pref pref, int mask, const char *list, int allowing)
	Parse an "allow" or "deny" line in a channel or device configuration and update the capabilities mask and pref if provided. Video codecs are not added to codec preference lists, since we can not transcode.
void	ast_swapcopy_samples (void dst, const void src, int samples)
Variables
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_smoother *	ast_smoother_new (int bytes)
struct ast_frame *	ast_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.

For an explanation of frames, see AST Multimedia and signalling frames
Frames are send of Asterisk channels, see Asterisk Channels

Definition in file frame.h.

Define Documentation

#define AST_FORMAT_ADPCM (1 << 5)

ADPCM (IMA)

Definition at line 251 of file frame.h.

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

#define AST_FORMAT_ALAW (1 << 3)

Raw A-law data (G.711)

Definition at line 247 of file frame.h.

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

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

#define AST_FORMAT_G722 (1 << 12)

G.722

Definition at line 265 of file frame.h.

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

#define AST_FORMAT_G723_1 (1 << 0)

G.723.1 compression

Definition at line 241 of file frame.h.

Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_write(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_destroy(), dahdi_translate(), g723_read(), g723_write(), load_module(), phone_request(), phone_setup(), phone_write(), register_translator(), and start_rtp().

#define AST_FORMAT_G726 (1 << 11)

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

Definition at line 263 of file frame.h.

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

#define AST_FORMAT_G726_AAL2 (1 << 4)

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

Definition at line 249 of file frame.h.

Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_lookup_mime_subtype(), ast_rtp_set_rtpmap_type_rate(), codec_ast2skinny(), codec_skinny2ast(), and setup_rtp_connection().

#define AST_FORMAT_G729A (1 << 8)

G.729A audio

Definition at line 257 of file frame.h.

Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_destroy(), dahdi_translate(), g729_read(), g729_write(), load_module(), phone_request(), phone_setup(), phone_write(), and start_rtp().

#define AST_FORMAT_GSM (1 << 1)

GSM compression

Definition at line 243 of file frame.h.

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

#define AST_FORMAT_H261 (1 << 18)

H.261 Video

Definition at line 279 of file frame.h.

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

#define AST_FORMAT_H263 (1 << 19)

H.263 Video

Definition at line 281 of file frame.h.

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

#define AST_FORMAT_H263_PLUS (1 << 20)

H.263+ Video

Definition at line 283 of file frame.h.

Referenced by h263p_encap().

#define AST_FORMAT_H264 (1 << 21)

H.264 Video

Definition at line 285 of file frame.h.

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

#define AST_FORMAT_ILBC (1 << 10)

iLBC Free Compression

Definition at line 261 of file frame.h.

Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_codec_interp_len(), convertcap(), ilbc_read(), ilbc_sample(), and ilbc_write().

#define AST_FORMAT_JPEG (1 << 16)

JPEG Images

Definition at line 275 of file frame.h.

Referenced by jpeg_read_image(), and jpeg_write_image().

#define AST_FORMAT_LPC10 (1 << 7)

LPC10, 180 samples/frame

Definition at line 255 of file frame.h.

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

#define AST_FORMAT_MAX_TEXT (1 << 28))

Maximum text mask

Definition at line 294 of file frame.h.

#define AST_FORMAT_MP4_VIDEO (1 << 22)

MPEG4 Video

Definition at line 287 of file frame.h.

Referenced by mpeg4_encap().

#define AST_FORMAT_PNG (1 << 17)

PNG Images

Definition at line 277 of file frame.h.

Referenced by phone_read().

#define AST_FORMAT_SIREN14 (1 << 14)

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

Definition at line 269 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 (1 << 13)

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

Definition at line 267 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 (1 << 6)

Raw 16-bit Signed Linear (8000 Hz) PCM

Definition at line 253 of file frame.h.

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

#define AST_FORMAT_SLINEAR16 (1 << 15)

Raw 16-bit Signed Linear (16000 Hz) PCM

Definition at line 271 of file frame.h.

Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_slinfactory_init_rate(), console_new(), slin16_sample(), slinear_read(), slinear_write(), softmix_bridge_join(), softmix_bridge_write(), and stream_monitor().

#define AST_FORMAT_SPEEX (1 << 9)

SpeeX Free Compression

Definition at line 259 of file frame.h.

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

#define AST_FORMAT_T140 (1 << 27)

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

Definition at line 292 of file frame.h.

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

#define AST_FORMAT_T140RED (1 << 26)

T.140 RED Text format RFC 4103

Definition at line 290 of file frame.h.

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

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

Definition at line 295 of file frame.h.

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

#define AST_FORMAT_ULAW (1 << 2)

Raw mu-law data (G.711)

Definition at line 245 of file frame.h.

Referenced by __adsi_transmit_messages(), _ast_adsi_transmit_message_full(), adsi_careful_send(), alarmreceiver_exec(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), calc_energy(), codec_ast2skinny(), codec_skinny2ast(), conf_run(), convertcap(), dahdi_new(), dahdi_read(), dahdi_translate(), dahdi_write(), find_transcoders(), is_encoder(), load_module(), milliwatt_generate(), oh323_rtp_read(), old_milliwatt_exec(), phone_request(), phone_setup(), phone_write(), pri_dchannel(), send_tone_burst(), start_rtp(), and ulaw_sample().

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

Definition at line 288 of file frame.h.

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

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

Definition at line 495 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 494 of file frame.h.

Referenced by phone_read().

#define AST_FRAME_DTMF AST_FRAME_DTMF_END

Definition at line 124 of file frame.h.

Referenced by __adsi_transmit_messages(), __ast_play_and_record(), action_atxfer(), action_dahdidialoffhook(), agent_ack_sleep(), ast_audiohook_write_list(), ast_bridge_call(), ast_dsp_process(), ast_generic_bridge(), ast_jb_put(), background_detect_exec(), cb_events(), channel_spy(), cli_console_dial(), conf_exec(), conf_run(), console_dial(), dahdi_bridge(), dahdi_read(), dictate_exec(), disa_exec(), do_immediate_setup(), echo_exec(), eivr_comm(), feature_request_and_dial(), gtalk_handle_dtmf(), handle_recordfile(), handle_request(), handle_request_info(), handle_speechrecognize(), iax2_bridge(), jingle_handle_dtmf(), mgcp_rtp_read(), misdn_bridge(), mp3_exec(), NBScat_exec(), oh323_rtp_read(), phone_exception(), pri_dchannel(), process_ast_dsp(), receive_dtmf_digits(), record_exec(), rpt(), rpt_call(), rpt_exec(), send_waveform_to_channel(), sip_rtp_read(), speech_background(), ss_thread(), unistim_do_senddigit(), unistim_senddigit_end(), volume_callback(), wait_for_answer(), and wait_for_winner().

#define AST_FRAME_SET_BUFFER	(	fr,
		_base,
		_ofs,
		_datalen	)

Value:

{              \
   (fr)->data.ptr = (char *)_base + (_ofs);  \
   (fr)->offset = (_ofs);        \
   (fr)->datalen = (_datalen);      \
   }

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

Definition at line 182 of file frame.h.

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

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

Definition at line 462 of file frame.h.

Referenced by __adsi_transmit_messages(), __ast_answer(), __ast_play_and_record(), __ast_queue_frame(), __ast_read(), __ast_request_and_dial(), adsi_careful_send(), agent_ack_sleep(), agent_read(), ast_audiohook_read_frame(), ast_autoservice_stop(), ast_bridge_call(), ast_bridge_handle_trip(), ast_channel_free(), ast_dsp_process(), ast_generic_bridge(), ast_jb_destroy(), ast_jb_put(), ast_readaudio_callback(), ast_readvideo_callback(), ast_recvtext(), ast_rtp_write(), ast_safe_sleep_conditional(), ast_send_image(), ast_slinfactory_destroy(), ast_slinfactory_feed(), ast_slinfactory_flush(), ast_slinfactory_read(), ast_tonepair(), ast_translate(), ast_udptl_bridge(), ast_waitfordigit_full(), ast_write(), ast_writestream(), async_wait(), audio_audiohook_write_list(), autoservice_run(), background_detect_exec(), bridge_handle_dtmf(), bridge_native_loop(), bridge_p2p_loop(), builtin_atxfer(), calc_cost(), channel_spy(), check_goto_on_transfer(), conf_exec(), conf_flush(), conf_free(), conf_run(), create_jb(), dahdi_bridge(), dial_exec_full(), dictate_exec(), disa_exec(), do_idle_thread(), do_waiting(), echo_exec(), eivr_comm(), feature_request_and_dial(), find_cache(), gen_generate(), handle_cli_file_convert(), handle_recordfile(), handle_speechrecognize(), iax2_bridge(), iax_park_thread(), ices_exec(), isAnsweringMachine(), jack_exec(), jb_empty_and_reset_adaptive(), jb_empty_and_reset_fixed(), jb_get_and_deliver(), launch_asyncagi(), manage_parkinglot(), masq_park_call(), measurenoise(), moh_files_generator(), monitor_dial(), mp3_exec(), NBScat_exec(), read_frame(), receive_dtmf_digits(), record_exec(), recordthread(), rpt(), rpt_exec(), run_agi(), send_tone_burst(), send_waveform_to_channel(), sendurl_exec(), speech_background(), spy_generate(), ss_thread(), transmit_audio(), transmit_t38(), wait_for_answer(), wait_for_hangup(), wait_for_winner(), waitforring_exec(), and waitstream_core().

#define AST_FRIENDLY_OFFSET 64

Offset into a frame's data buffer.

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

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

Definition at line 203 of file frame.h.

Referenced by __get_from_jb(), alsa_read(), ast_frdup(), ast_frisolate(), ast_prod(), ast_rtcp_read(), ast_rtp_read(), ast_smoother_read(), ast_trans_frameout(), ast_udptl_read(), conf_run(), dahdi_decoder_frameout(), dahdi_encoder_frameout(), dahdi_read(), fax_generator_generate(), g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), iax_frame_wrap(), ilbc_read(), jb_get_and_deliver(), linear_generator(), milliwatt_generate(), moh_generate(), mohalloc(), mp3_exec(), NBScat_exec(), newpvt(), ogg_vorbis_read(), oss_read(), pcm_read(), phone_read(), playtones_generator(), process_rfc3389(), send_tone_burst(), send_waveform_to_channel(), siren14read(), siren7read(), slinear_read(), sms_generate(), tonepair_generator(), usbradio_read(), vox_read(), and wav_read().

#define AST_HTML_BEGIN 4

Beginning frame

Definition at line 225 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_DATA 2

Data frame

Definition at line 223 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_END 8

End frame

Definition at line 227 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_LDCOMPLETE 16

Load is complete

Definition at line 229 of file frame.h.

Referenced by ast_frame_dump(), and sendurl_exec().

#define AST_HTML_LINKREJECT 20

Reject link request

Definition at line 237 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_LINKURL 18

Send URL, and track

Definition at line 233 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_NOSUPPORT 17

Peer is unable to support HTML

Definition at line 231 of file frame.h.

Referenced by ast_frame_dump(), and sendurl_exec().

#define AST_HTML_UNLINK 19

No more HTML linkage

Definition at line 235 of file frame.h.

Referenced by ast_frame_dump().

#define AST_HTML_URL 1

Sending a URL

Definition at line 221 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 209 of file frame.h.

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

#define AST_MALLOCD_HDR (1 << 0)

Need the header be free'd?

Definition at line 207 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 211 of file frame.h.

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

#define AST_MIN_OFFSET 32

Definition at line 204 of file frame.h.

Referenced by __ast_smoother_feed().

#define AST_MODEM_T38 1

T.38 Fax-over-IP

Definition at line 215 of file frame.h.

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

#define AST_MODEM_V150 2

V.150 Modem-over-IP

Definition at line 217 of file frame.h.

Referenced by ast_frame_dump().

#define AST_OPTION_AUDIO_MODE 4

Set (or clear) Audio (Not-Clear) Mode

Definition at line 376 of file frame.h.

Referenced by dahdi_hangup(), and dahdi_setoption().

#define AST_OPTION_ECHOCAN 8

Explicitly enable or disable echo cancelation for the given channel

Definition at line 398 of file frame.h.

Referenced by dahdi_setoption().

#define AST_OPTION_FLAG_ACCEPT 1

Definition at line 359 of file frame.h.

#define AST_OPTION_FLAG_ANSWER 5

Definition at line 362 of file frame.h.

#define AST_OPTION_FLAG_QUERY 4

Definition at line 361 of file frame.h.

#define AST_OPTION_FLAG_REJECT 2

Definition at line 360 of file frame.h.

#define AST_OPTION_FLAG_REQUEST 0

Definition at line 358 of file frame.h.

Referenced by ast_bridge_call(), and iax2_setoption().

#define AST_OPTION_FLAG_WTF 6

Definition at line 363 of file frame.h.

#define AST_OPTION_OPRMODE 7

Definition at line 395 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)

Definition at line 373 of file frame.h.

Referenced by dahdi_setoption(), and rpt().

#define AST_OPTION_RXGAIN 6

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

Definition at line 392 of file frame.h.

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

#define AST_OPTION_T38_STATE 10

Definition at line 404 of file frame.h.

Referenced by ast_channel_get_t38_state(), and sip_queryoption().

#define AST_OPTION_TDD 2

Put a compatible channel into TDD (TTY for the hearing-impared) mode

Definition at line 370 of file frame.h.

Referenced by dahdi_hangup(), dahdi_setoption(), and handle_tddmode().

#define AST_OPTION_TONE_VERIFY 1

Verify touchtones by muting audio transmission (and reception) and verify the tone is still present

Definition at line 367 of file frame.h.

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

#define AST_OPTION_TXGAIN 5

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

Definition at line 384 of file frame.h.

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

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

Definition at line 565 of file frame.h.

Referenced by ast_rtp_write().

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

Definition at line 570 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 571 of file frame.h.

#define AST_SMOOTHER_FLAG_BE (1 << 1)

Definition at line 355 of file frame.h.

Referenced by ast_rtp_write().

#define AST_SMOOTHER_FLAG_G729 (1 << 0)

Definition at line 354 of file frame.h.

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

Enumeration Type Documentation

anonymous enum

Enumerator:

AST_FRFLAG_HAS_TIMING_INFO	This frame contains valid timing information
AST_FRFLAG_FROM_TRANSLATOR	This frame came from a translator and is still the original frame. The translator can not be free'd if the frame inside of it still has this flag set.
AST_FRFLAG_FROM_DSP	This frame came from a dsp and is still the original frame. The dsp cannot be free'd if the frame inside of it still has this flag set.

Definition at line 126 of file frame.h.

00126      {
00127    /*! This frame contains valid timing information */
00128    AST_FRFLAG_HAS_TIMING_INFO = (1 << 0),
00129    /*! This frame came from a translator and is still the original frame.
00130     *  The translator can not be free'd if the frame inside of it still has
00131     *  this flag set. */
00132    AST_FRFLAG_FROM_TRANSLATOR = (1 << 1),
00133    /*! This frame came from a dsp and is still the original frame.
00134     *  The dsp cannot be free'd if the frame inside of it still has
00135     *  this flag set. */
00136    AST_FRFLAG_FROM_DSP = (1 << 2),
00137 };

enum ast_control_frame_type

Enumerator:

AST_CONTROL_HANGUP	Other end has hungup
AST_CONTROL_RING	Local ring
AST_CONTROL_RINGING	Remote end is ringing
AST_CONTROL_ANSWER	Remote end has answered
AST_CONTROL_BUSY	Remote end is busy
AST_CONTROL_TAKEOFFHOOK	Make it go off hook
AST_CONTROL_OFFHOOK	Line is off hook
AST_CONTROL_CONGESTION	Congestion (circuits busy)
AST_CONTROL_FLASH	Flash hook
AST_CONTROL_WINK	Wink
AST_CONTROL_OPTION	Set a low-level option
AST_CONTROL_RADIO_KEY	Key Radio
AST_CONTROL_RADIO_UNKEY	Un-Key Radio
AST_CONTROL_PROGRESS	Indicate PROGRESS
AST_CONTROL_PROCEEDING	Indicate CALL PROCEEDING
AST_CONTROL_HOLD	Indicate call is placed on hold
AST_CONTROL_UNHOLD	Indicate call is left from hold
AST_CONTROL_VIDUPDATE	Indicate video frame update
_XXX_AST_CONTROL_T38	T38 state change request/notification Deprecated: This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead.
AST_CONTROL_SRCUPDATE	Indicate source of media has changed
AST_CONTROL_T38_PARAMETERS	T38 state change request/notification with parameters

Definition at line 297 of file frame.h.

00298 00299 00300 00301 00302 00303 00304 00305 00306 00307 00308 00309 00310 00311 00312 00313 00314 00315 00316 00317 00318 00319 };

class="fragment">00297 { AST_CONTROL_HANGUP = 1, /*!< Other end has hungup */ AST_CONTROL_RING = 2, /*!< Local ring */ AST_CONTROL_RINGING = 3, /*!< Remote end is ringing */ AST_CONTROL_ANSWER = 4, /*!< Remote end has answered */ AST_CONTROL_BUSY = 5, /*!< Remote end is busy */ AST_CONTROL_TAKEOFFHOOK = 6, /*!< Make it go off hook */ AST_CONTROL_OFFHOOK = 7, /*!< Line is off hook */ AST_CONTROL_CONGESTION = 8, /*!< Congestion (circuits busy) */ AST_CONTROL_FLASH = 9, /*!< Flash hook */ AST_CONTROL_WINK = 10, /*!< Wink */ AST_CONTROL_OPTION = 11, /*!< Set a low-level option */ AST_CONTROL_RADIO_KEY = 12, /*!< Key Radio */ AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */ AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */ AST_CONTROL_PROCEEDING = 15, /*!< Indicate CALL PROCEEDING */ AST_CONTROL_HOLD = 16, /*!< Indicate call is placed on hold */ AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold */ AST_CONTROL_VIDUPDATE = 18, /*!< Indicate video frame update */ _XXX_AST_CONTROL_T38 = 19, /*!< T38 state change request/notification \deprecated This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead. */ AST_CONTROL_SRCUPDATE = 20, /*!< Indicate source of media has changed */ AST_CONTROL_T38_PARAMETERS = 24, /*!< T38 state change request/notification with parameters */

enum ast_control_t38

Enumerator:

AST_T38_REQUEST_NEGOTIATE	Request T38 on a channel (voice to fax)
AST_T38_REQUEST_TERMINATE	Terminate T38 on a channel (fax to voice)
AST_T38_NEGOTIATED	T38 negotiated (fax mode)
AST_T38_TERMINATED	T38 terminated (back to voice)
AST_T38_REFUSED	T38 refused for some reason (usually rejected by remote end)

Definition at line 321 of file frame.h.

00321                      {
00322    AST_T38_REQUEST_NEGOTIATE = 1,   /*!< Request T38 on a channel (voice to fax) */
00323    AST_T38_REQUEST_TERMINATE, /*!< Terminate T38 on a channel (fax to voice) */
00324    AST_T38_NEGOTIATED,     /*!< T38 negotiated (fax mode) */
00325    AST_T38_TERMINATED,     /*!< T38 terminated (back to voice) */
00326    AST_T38_REFUSED         /*!< T38 refused for some reason (usually rejected by remote end) */
00327 };

enum ast_control_t38_rate

Enumerator:

AST_T38_RATE_2400
AST_T38_RATE_4800
AST_T38_RATE_7200
AST_T38_RATE_9600
AST_T38_RATE_12000
AST_T38_RATE_14400

Definition at line 329 of file frame.h.

00329                           {
00330    AST_T38_RATE_2400 = 0,
00331    AST_T38_RATE_4800,
00332    AST_T38_RATE_7200,
00333    AST_T38_RATE_9600,
00334    AST_T38_RATE_12000,
00335    AST_T38_RATE_14400,
00336 };

enum ast_control_t38_rate_management

Enumerator:

AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF
AST_T38_RATE_MANAGEMENT_LOCAL_TCF

Definition at line 338 of file frame.h.

00338                                      {
00339    AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF = 0,
00340    AST_T38_RATE_MANAGEMENT_LOCAL_TCF,
00341 };

enum ast_frame_type

Frame types.

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

Enumerator:

AST_FRAME_DTMF_END	DTMF end event, subclass is the digit
AST_FRAME_VOICE	Voice data, subclass is AST_FORMAT_*
AST_FRAME_VIDEO	Video frame, maybe?? :)
AST_FRAME_CONTROL	A control frame, subclass is AST_CONTROL_*
AST_FRAME_NULL	An empty, useless frame
AST_FRAME_IAX	Inter Asterisk Exchange private frame type
AST_FRAME_TEXT	Text messages
AST_FRAME_IMAGE	Image Frames
AST_FRAME_HTML	HTML Frame
AST_FRAME_CNG	Comfort Noise frame (subclass is level of CNG in -dBov), body may include zero or more 8-bit quantization coefficients
AST_FRAME_MODEM	Modem-over-IP data streams
AST_FRAME_DTMF_BEGIN	DTMF begin event, subclass is the digit

Definition at line 97 of file frame.h.

00097                     {
00098    /*! DTMF end event, subclass is the digit */
00099    AST_FRAME_DTMF_END = 1,
00100    /*! Voice data, subclass is AST_FORMAT_* */
00101    AST_FRAME_VOICE,
00102    /*! Video frame, maybe?? :) */
00103    AST_FRAME_VIDEO,
00104    /*! A control frame, subclass is AST_CONTROL_* */
00105    AST_FRAME_CONTROL,
00106    /*! An empty, useless frame */
00107    AST_FRAME_NULL,
00108    /*! Inter Asterisk Exchange private frame type */
00109    AST_FRAME_IAX,
00110    /*! Text messages */
00111    AST_FRAME_TEXT,
00112    /*! Image Frames */
00113    AST_FRAME_IMAGE,
00114    /*! HTML Frame */
00115    AST_FRAME_HTML,
00116    /*! Comfort Noise frame (subclass is level of CNG in -dBov), 
00117        body may include zero or more 8-bit quantization coefficients */
00118    AST_FRAME_CNG,
00119    /*! Modem-over-IP data streams */
00120    AST_FRAME_MODEM,  
00121    /*! DTMF begin event, subclass is the digit */
00122    AST_FRAME_DTMF_BEGIN,
00123 };

Function Documentation

int __ast_smoother_feed	(	struct ast_smoother *	s,
		struct ast_frame *	f,
		int	swap
	)

Definition at line 201 of file frame.c.

References AST_FRAME_VOICE, ast_log(), AST_MIN_OFFSET, AST_SMOOTHER_FLAG_G729, ast_swapcopy_samples(), 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.

00202 {
00203    if (f->frametype != AST_FRAME_VOICE) {
00204       ast_log(LOG_WARNING, "Huh?  Can't smooth a non-voice frame!\n");
00205       return -1;
00206    }
00207    if (!s->format) {
00208       s->format = f->subclass;
00209       s->samplesperbyte = (float)f->samples / (float)f->datalen;
00210    } else if (s->format != f->subclass) {
00211       ast_log(LOG_WARNING, "Smoother was working on %d format frames, now trying to feed %d?\n", s->format, f->subclass);
00212       return -1;
00213    }
00214    if (s->len + f->datalen > SMOOTHER_SIZE) {
00215       ast_log(LOG_WARNING, "Out of smoother space\n");
00216       return -1;
00217    }
00218    if (((f->datalen == s->size) ||
00219         ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) &&
00220        !s->opt &&
00221        !s->len &&
00222        (f->offset >= AST_MIN_OFFSET)) {
00223       /* Optimize by sending the frame we just got
00224          on the next read, thus eliminating the douple
00225          copy */
00226       if (swap)
00227          ast_swapcopy_samples(f->data.ptr, f->data.ptr, f->samples);
00228       s->opt = f;
00229       s->opt_needs_swap = swap ? 1 : 0;
00230       return 0;
00231    }
00232 
00233    return smoother_frame_feed(s, f, swap);
00234 }

char* ast_codec2str ( int codec )

Get a name from a format Gets a name from a format.

Parameters:

codec

codec number (1,2,4,8,16,etc.)

Returns:: This returns a static string identifying the format on success, 0 on error.

Definition at line 654 of file frame.c.

References ARRAY_LEN, and ast_format_list::desc.

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

00655 {
00656    int x;
00657    char *ret = "unknown";
00658    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00659       if (AST_FORMAT_LIST[x].bits == codec) {
00660          ret = AST_FORMAT_LIST[x].desc;
00661          break;
00662       }
00663    }
00664    return ret;
00665 }

int ast_codec_choose	(	struct ast_codec_pref *	pref,
		int	formats,
		int	find_best
	)

Select the best audio format according to preference list from supplied options. If "find_best" is non-zero then if nothing is found, the "Best" format of the format list is selected, otherwise 0 is returned.

Definition at line 1211 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().

01212 {
01213    int x, ret = 0, slot;
01214 
01215    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01216       slot = pref->order[x];
01217 
01218       if (!slot)
01219          break;
01220       if (formats & AST_FORMAT_LIST[slot-1].bits) {
01221          ret = AST_FORMAT_LIST[slot-1].bits;
01222          break;
01223       }
01224    }
01225    if (ret & AST_FORMAT_AUDIO_MASK)
01226       return ret;
01227 
01228    ast_debug(4, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec");
01229 
01230       return find_best ? ast_best_codec(formats) : 0;
01231 }

int ast_codec_get_len	(	int	format,
		int	samples
	)

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

Definition at line 1484 of file frame.c.

References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G722, AST_FORMAT_G723_1, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_ILBC, AST_FORMAT_SIREN14, AST_FORMAT_SIREN7, AST_FORMAT_SLINEAR, AST_FORMAT_SLINEAR16, AST_FORMAT_ULAW, ast_getformatname(), ast_log(), len(), and LOG_WARNING.

Referenced by moh_generate(), and monmp3thread().

01485 {
01486    int len = 0;
01487 
01488    /* XXX Still need speex, and lpc10 XXX */ 
01489    switch(format) {
01490    case AST_FORMAT_G723_1:
01491       len = (samples / 240) * 20;
01492       break;
01493    case AST_FORMAT_ILBC:
01494       len = (samples / 240) * 50;
01495       break;
01496    case AST_FORMAT_GSM:
01497       len = (samples / 160) * 33;
01498       break;
01499    case AST_FORMAT_G729A:
01500       len = samples / 8;
01501       break;
01502    case AST_FORMAT_SLINEAR:
01503    case AST_FORMAT_SLINEAR16:
01504       len = samples * 2;
01505       break;
01506    case AST_FORMAT_ULAW:
01507    case AST_FORMAT_ALAW:
01508       len = samples;
01509       break;
01510    case AST_FORMAT_G722:
01511    case AST_FORMAT_ADPCM:
01512    case AST_FORMAT_G726:
01513    case AST_FORMAT_G726_AAL2:
01514       len = samples / 2;
01515       break;
01516    case AST_FORMAT_SIREN7:
01517       /* 16,000 samples per second at 32kbps is 4,000 bytes per second */
01518       len = samples / (16000 / 4000);
01519       break;
01520    case AST_FORMAT_SIREN14:
01521       /* 32,000 samples per second at 48kbps is 6,000 bytes per second */
01522       len = (int) samples / ((float) 32000 / 6000);
01523       break;
01524    default:
01525       ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format));
01526    }
01527 
01528    return len;
01529 }

int ast_codec_get_samples ( struct ast_frame * f )

Returns the number of samples contained in the frame.

Definition at line 1431 of file frame.c.

References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G722, AST_FORMAT_G723_1, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_ILBC, AST_FORMAT_LPC10, AST_FORMAT_SIREN14, AST_FORMAT_SIREN7, AST_FORMAT_SLINEAR, AST_FORMAT_SLINEAR16, AST_FORMAT_SPEEX, AST_FORMAT_ULAW, ast_getformatname(), ast_log(), ast_frame::data, ast_frame::datalen, g723_samples(), LOG_WARNING, ast_frame::ptr, speex_samples(), and ast_frame::subclass.

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

01432 {
01433    int samples = 0;
01434 
01435    switch(f->subclass) {
01436    case AST_FORMAT_SPEEX:
01437       samples = speex_samples(f->data.ptr, f->datalen);
01438       break;
01439    case AST_FORMAT_G723_1:
01440       samples = g723_samples(f->data.ptr, f->datalen);
01441       break;
01442    case AST_FORMAT_ILBC:
01443       samples = 240 * (f->datalen / 50);
01444       break;
01445    case AST_FORMAT_GSM:
01446       samples = 160 * (f->datalen / 33);
01447       break;
01448    case AST_FORMAT_G729A:
01449       samples = f->datalen * 8;
01450       break;
01451    case AST_FORMAT_SLINEAR:
01452    case AST_FORMAT_SLINEAR16:
01453       samples = f->datalen / 2;
01454       break;
01455    case AST_FORMAT_LPC10:
01456       /* assumes that the RTP packet contains one LPC10 frame */
01457       samples = 22 * 8;
01458       samples += (((char *)(f->data.ptr))[7] & 0x1) * 8;
01459       break;
01460    case AST_FORMAT_ULAW:
01461    case AST_FORMAT_ALAW:
01462       samples = f->datalen;
01463       break;
01464    case AST_FORMAT_G722:
01465    case AST_FORMAT_ADPCM:
01466    case AST_FORMAT_G726:
01467    case AST_FORMAT_G726_AAL2:
01468       samples = f->datalen * 2;
01469       break;
01470    case AST_FORMAT_SIREN7:
01471       /* 16,000 samples per second at 32kbps is 4,000 bytes per second */
01472       samples = f->datalen * (16000 / 4000);
01473       break;
01474    case AST_FORMAT_SIREN14:
01475       /* 32,000 samples per second at 48kbps is 6,000 bytes per second */
01476       samples = (int) f->datalen * ((float) 32000 / 6000);
01477       break;
01478    default:
01479       ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass));
01480    }
01481    return samples;
01482 }

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

Gets duration in ms of interpolation frame for a format.

Definition at line 653 of file frame.h.

References AST_FORMAT_ILBC.

Referenced by __get_from_jb(), and jb_get_and_deliver().

00654 { 
00655    return (format == AST_FORMAT_ILBC) ? 30 : 20;
00656 }

int ast_codec_pref_append	(	struct ast_codec_pref *	pref,
		int	format
	)

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

Definition at line 1071 of file frame.c.

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

Referenced by ast_parse_allow_disallow().

01072 {
01073    int x, newindex = 0;
01074 
01075    ast_codec_pref_remove(pref, format);
01076 
01077    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01078       if (AST_FORMAT_LIST[x].bits == format) {
01079          newindex = x + 1;
01080          break;
01081       }
01082    }
01083 
01084    if (newindex) {
01085       for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01086          if (!pref->order[x]) {
01087             pref->order[x] = newindex;
01088             break;
01089          }
01090       }
01091    }
01092 
01093    return x;
01094 }

void ast_codec_pref_convert	(	struct ast_codec_pref *	pref,
		char *	buf,
		size_t	size,
		int	right
	)

Shift an audio codec preference list up or down 65 bytes so that it becomes an ASCII string.

Definition at line 974 of file frame.c.

References ast_codec_pref::order.

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

00975 {
00976    int x, differential = (int) 'A', mem;
00977    char *from, *to;
00978 
00979    if (right) {
00980       from = pref->order;
00981       to = buf;
00982       mem = size;
00983    } else {
00984       to = pref->order;
00985       from = buf;
00986       mem = 32;
00987    }
00988 
00989    memset(to, 0, mem);
00990    for (x = 0; x < 32 ; x++) {
00991       if (!from[x])
00992          break;
00993       to[x] = right ? (from[x] + differential) : (from[x] - differential);
00994    }
00995 }

struct ast_format_list ast_codec_pref_getsize	(	struct ast_codec_pref *	pref,
		int	format
	)			`[read]`

Get packet size for codec.

Definition at line 1172 of file frame.c.

References ARRAY_LEN, ast_format_list::bits, ast_format_list::cur_ms, ast_format_list::def_ms, format, ast_format_list::inc_ms, ast_format_list::max_ms, and ast_format_list::min_ms.

Referenced by add_codec_to_sdp(), ast_rtp_bridge(), ast_rtp_codec_setpref(), ast_rtp_write(), handle_open_receive_channel_ack_message(), skinny_set_rtp_peer(), and transmit_connect().

01173 {
01174    int x, idx = -1, framems = 0;
01175    struct ast_format_list fmt = { 0, };
01176 
01177    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01178       if (AST_FORMAT_LIST[x].bits == format) {
01179          fmt = AST_FORMAT_LIST[x];
01180          idx = x;
01181          break;
01182       }
01183    }
01184 
01185    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01186       if (pref->order[x] == (idx + 1)) {
01187          framems = pref->framing[x];
01188          break;
01189       }
01190    }
01191 
01192    /* size validation */
01193    if (!framems)
01194       framems = AST_FORMAT_LIST[idx].def_ms;
01195 
01196    if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */
01197       framems -= framems % AST_FORMAT_LIST[idx].inc_ms;
01198 
01199    if (framems < AST_FORMAT_LIST[idx].min_ms)
01200       framems = AST_FORMAT_LIST[idx].min_ms;
01201 
01202    if (framems > AST_FORMAT_LIST[idx].max_ms)
01203       framems = AST_FORMAT_LIST[idx].max_ms;
01204 
01205    fmt.cur_ms = framems;
01206 
01207    return fmt;
01208 }

int ast_codec_pref_index	(	struct ast_codec_pref *	pref,
		int	index
	)

Codec located at a particular place in the preference index.

Audio Codec Preferences

Definition at line 1032 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().

01033 {
01034    int slot = 0;
01035 
01036    if ((idx >= 0) && (idx < sizeof(pref->order))) {
01037       slot = pref->order[idx];
01038    }
01039 
01040    return slot ? AST_FORMAT_LIST[slot - 1].bits : 0;
01041 }

void ast_codec_pref_init ( struct ast_codec_pref * pref )

Initialize an audio codec preference to "no preference".

Audio Codec Preferences

void ast_codec_pref_prepend	(	struct ast_codec_pref *	pref,
		int	format,
		int	only_if_existing
	)

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

Definition at line 1097 of file frame.c.

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

Referenced by create_addr().

01098 {
01099    int x, newindex = 0;
01100 
01101    /* First step is to get the codecs "index number" */
01102    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01103       if (AST_FORMAT_LIST[x].bits == format) {
01104          newindex = x + 1;
01105          break;
01106       }
01107    }
01108    /* Done if its unknown */
01109    if (!newindex)
01110       return;
01111 
01112    /* Now find any existing occurrence, or the end */
01113    for (x = 0; x < 32; x++) {
01114       if (!pref->order[x] || pref->order[x] == newindex)
01115          break;
01116    }
01117 
01118    if (only_if_existing && !pref->order[x])
01119       return;
01120 
01121    /* Move down to make space to insert - either all the way to the end,
01122       or as far as the existing location (which will be overwritten) */
01123    for (; x > 0; x--) {
01124       pref->order[x] = pref->order[x - 1];
01125       pref->framing[x] = pref->framing[x - 1];
01126    }
01127 
01128    /* And insert the new entry */
01129    pref->order[0] = newindex;
01130    pref->framing[0] = 0; /* ? */
01131 }

void ast_codec_pref_remove	(	struct ast_codec_pref *	pref,
		int	format
	)

Remove audio a codec from a preference list.

Definition at line 1044 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().

01045 {
01046    struct ast_codec_pref oldorder;
01047    int x, y = 0;
01048    int slot;
01049    int size;
01050 
01051    if (!pref->order[0])
01052       return;
01053 
01054    memcpy(&oldorder, pref, sizeof(oldorder));
01055    memset(pref, 0, sizeof(*pref));
01056 
01057    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01058       slot = oldorder.order[x];
01059       size = oldorder.framing[x];
01060       if (! slot)
01061          break;
01062       if (AST_FORMAT_LIST[slot-1].bits != format) {
01063          pref->order[y] = slot;
01064          pref->framing[y++] = size;
01065       }
01066    }
01067    
01068 }

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

Set packet size for codec.

Definition at line 1134 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().

01135 {
01136    int x, idx = -1;
01137 
01138    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01139       if (AST_FORMAT_LIST[x].bits == format) {
01140          idx = x;
01141          break;
01142       }
01143    }
01144 
01145    if (idx < 0)
01146       return -1;
01147 
01148    /* size validation */
01149    if (!framems)
01150       framems = AST_FORMAT_LIST[idx].def_ms;
01151 
01152    if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */
01153       framems -= framems % AST_FORMAT_LIST[idx].inc_ms;
01154 
01155    if (framems < AST_FORMAT_LIST[idx].min_ms)
01156       framems = AST_FORMAT_LIST[idx].min_ms;
01157 
01158    if (framems > AST_FORMAT_LIST[idx].max_ms)
01159       framems = AST_FORMAT_LIST[idx].max_ms;
01160 
01161    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
01162       if (pref->order[x] == (idx + 1)) {
01163          pref->framing[x] = framems;
01164          break;
01165       }
01166    }
01167 
01168    return x;
01169 }

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 997 of file frame.c.

References ast_codec_pref_index(), and ast_getformatname().

Referenced by dump_prefs(), and socket_process().

00998 {
00999    int x, codec; 
01000    size_t total_len, slen;
01001    char *formatname;
01002    
01003    memset(buf,0,size);
01004    total_len = size;
01005    buf[0] = '(';
01006    total_len--;
01007    for(x = 0; x < 32 ; x++) {
01008       if (total_len <= 0)
01009          break;
01010       if (!(codec = ast_codec_pref_index(pref,x)))
01011          break;
01012       if ((formatname = ast_getformatname(codec))) {
01013          slen = strlen(formatname);
01014          if (slen > total_len)
01015             break;
01016          strncat(buf, formatname, total_len - 1); /* safe */
01017          total_len -= slen;
01018       }
01019       if (total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) {
01020          strncat(buf, "|", total_len - 1); /* safe */
01021          total_len--;
01022       }
01023    }
01024    if (total_len) {
01025       strncat(buf, ")", total_len - 1); /* safe */
01026       total_len--;
01027    }
01028 
01029    return size - total_len;
01030 }

static force_inline int ast_format_rate ( int format ) [static]

Get the sample rate for a given format.

Definition at line 680 of file frame.h.

References AST_FORMAT_G722, AST_FORMAT_SIREN14, AST_FORMAT_SIREN7, and AST_FORMAT_SLINEAR16.

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

00681 {
00682    switch (format) {
00683    case AST_FORMAT_G722:
00684    case AST_FORMAT_SLINEAR16:
00685    case AST_FORMAT_SIREN7:
00686       return 16000;
00687    case AST_FORMAT_SIREN14:
00688       return 32000;
00689    default:
00690       return 8000;
00691    }
00692 }

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 1531 of file frame.c.

References AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_slinear_saturated_divide(), ast_slinear_saturated_multiply(), 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().

01532 {
01533    int count;
01534    short *fdata = f->data.ptr;
01535    short adjust_value = abs(adjustment);
01536 
01537    if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR))
01538       return -1;
01539 
01540    if (!adjustment)
01541       return 0;
01542 
01543    for (count = 0; count < f->samples; count++) {
01544       if (adjustment > 0) {
01545          ast_slinear_saturated_multiply(&fdata[count], &adjust_value);
01546       } else if (adjustment < 0) {
01547          ast_slinear_saturated_divide(&fdata[count], &adjust_value);
01548       }
01549    }
01550 
01551    return 0;
01552 }

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

Dump a frame for debugging purposes

Definition at line 756 of file frame.c.

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

Referenced by __ast_read(), and ast_write().

00757 {
00758    const char noname[] = "unknown";
00759    char ftype[40] = "Unknown Frametype";
00760    char cft[80];
00761    char subclass[40] = "Unknown Subclass";
00762    char csub[80];
00763    char moreinfo[40] = "";
00764    char cn[60];
00765    char cp[40];
00766    char cmn[40];
00767    const char *message = "Unknown";
00768 
00769    if (!name)
00770       name = noname;
00771 
00772 
00773    if (!f) {
00774       ast_verbose("%s [ %s (NULL) ] [%s]\n", 
00775          term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00776          term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 
00777          term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00778       return;
00779    }
00780    /* XXX We should probably print one each of voice and video when the format changes XXX */
00781    if (f->frametype == AST_FRAME_VOICE)
00782       return;
00783    if (f->frametype == AST_FRAME_VIDEO)
00784       return;
00785    switch(f->frametype) {
00786    case AST_FRAME_DTMF_BEGIN:
00787       strcpy(ftype, "DTMF Begin");
00788       subclass[0] = f->subclass;
00789       subclass[1] = '\0';
00790       break;
00791    case AST_FRAME_DTMF_END:
00792       strcpy(ftype, "DTMF End");
00793       subclass[0] = f->subclass;
00794       subclass[1] = '\0';
00795       break;
00796    case AST_FRAME_CONTROL:
00797       strcpy(ftype, "Control");
00798       switch(f->subclass) {
00799       case AST_CONTROL_HANGUP:
00800          strcpy(subclass, "Hangup");
00801          break;
00802       case AST_CONTROL_RING:
00803          strcpy(subclass, "Ring");
00804          break;
00805       case AST_CONTROL_RINGING:
00806          strcpy(subclass, "Ringing");
00807          break;
00808       case AST_CONTROL_ANSWER:
00809          strcpy(subclass, "Answer");
00810          break;
00811       case AST_CONTROL_BUSY:
00812          strcpy(subclass, "Busy");
00813          break;
00814       case AST_CONTROL_TAKEOFFHOOK:
00815          strcpy(subclass, "Take Off Hook");
00816          break;
00817       case AST_CONTROL_OFFHOOK:
00818          strcpy(subclass, "Line Off Hook");
00819          break;
00820       case AST_CONTROL_CONGESTION:
00821          strcpy(subclass, "Congestion");
00822          break;
00823       case AST_CONTROL_FLASH:
00824          strcpy(subclass, "Flash");
00825          break;
00826       case AST_CONTROL_WINK:
00827          strcpy(subclass, "Wink");
00828          break;
00829       case AST_CONTROL_OPTION:
00830          strcpy(subclass, "Option");
00831          break;
00832       case AST_CONTROL_RADIO_KEY:
00833          strcpy(subclass, "Key Radio");
00834          break;
00835       case AST_CONTROL_RADIO_UNKEY:
00836          strcpy(subclass, "Unkey Radio");
00837          break;
00838       case AST_CONTROL_HOLD:
00839          strcpy(subclass, "Hold");
00840          break;
00841       case AST_CONTROL_UNHOLD:
00842          strcpy(subclass, "Unhold");
00843          break;
00844       case AST_CONTROL_T38_PARAMETERS:
00845          if (f->datalen != sizeof(struct ast_control_t38_parameters)) {
00846             message = "Invalid";
00847          } else {
00848             struct ast_control_t38_parameters *parameters = f->data.ptr;
00849             enum ast_control_t38 state = parameters->request_response;
00850             if (state == AST_T38_REQUEST_NEGOTIATE)
00851                message = "Negotiation Requested";
00852             else if (state == AST_T38_REQUEST_TERMINATE)
00853                message = "Negotiation Request Terminated";
00854             else if (state == AST_T38_NEGOTIATED)
00855                message = "Negotiated";
00856             else if (state == AST_T38_TERMINATED)
00857                message = "Terminated";
00858             else if (state == AST_T38_REFUSED)
00859                message = "Refused";
00860          }
00861          snprintf(subclass, sizeof(subclass), "T38_Parameters/%s", message);
00862          break;
00863       case -1:
00864          strcpy(subclass, "Stop generators");
00865          break;
00866       default:
00867          snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass);
00868       }
00869       break;
00870    case AST_FRAME_NULL:
00871       strcpy(ftype, "Null Frame");
00872       strcpy(subclass, "N/A");
00873       break;
00874    case AST_FRAME_IAX:
00875       /* Should never happen */
00876       strcpy(ftype, "IAX Specific");
00877       snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass);
00878       break;
00879    case AST_FRAME_TEXT:
00880       strcpy(ftype, "Text");
00881       strcpy(subclass, "N/A");
00882       ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo));
00883       break;
00884    case AST_FRAME_IMAGE:
00885       strcpy(ftype, "Image");
00886       snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass));
00887       break;
00888    case AST_FRAME_HTML:
00889       strcpy(ftype, "HTML");
00890       switch(f->subclass) {
00891       case AST_HTML_URL:
00892          strcpy(subclass, "URL");
00893          ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo));
00894          break;
00895       case AST_HTML_DATA:
00896          strcpy(subclass, "Data");
00897          break;
00898       case AST_HTML_BEGIN:
00899          strcpy(subclass, "Begin");
00900          break;
00901       case AST_HTML_END:
00902          strcpy(subclass, "End");
00903          break;
00904       case AST_HTML_LDCOMPLETE:
00905          strcpy(subclass, "Load Complete");
00906          break;
00907       case AST_HTML_NOSUPPORT:
00908          strcpy(subclass, "No Support");
00909          break;
00910       case AST_HTML_LINKURL:
00911          strcpy(subclass, "Link URL");
00912          ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo));
00913          break;
00914       case AST_HTML_UNLINK:
00915          strcpy(subclass, "Unlink");
00916          break;
00917       case AST_HTML_LINKREJECT:
00918          strcpy(subclass, "Link Reject");
00919          break;
00920       default:
00921          snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass);
00922          break;
00923       }
00924       break;
00925    case AST_FRAME_MODEM:
00926       strcpy(ftype, "Modem");
00927       switch (f->subclass) {
00928       case AST_MODEM_T38:
00929          strcpy(subclass, "T.38");
00930          break;
00931       case AST_MODEM_V150:
00932          strcpy(subclass, "V.150");
00933          break;
00934       default:
00935          snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass);
00936          break;
00937       }
00938       break;
00939    default:
00940       snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype);
00941    }
00942    if (!ast_strlen_zero(moreinfo))
00943       ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n",  
00944              term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00945              term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)),
00946              f->frametype, 
00947              term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)),
00948              f->subclass, 
00949              term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)),
00950              term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00951    else
00952       ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n",  
00953              term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)),
00954              term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)),
00955              f->frametype, 
00956              term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)),
00957              f->subclass, 
00958              term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn)));
00959 }

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 373 of file frame.c.

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

Referenced by mixmonitor_thread().

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

int ast_frame_slinear_sum	(	struct ast_frame *	f1,
		struct ast_frame *	f2
	)

Sums two frames of audio samples.

Parameters:

	f1	The first frame (which will contain the result)
	f2	The second frame

Returns:: 0 for success, non-zero for an error

The frames must be AST_FRAME_VOICE and must contain AST_FORMAT_SLINEAR samples, and must contain the same number of samples.

Definition at line 1554 of file frame.c.

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

01555 {
01556    int count;
01557    short *data1, *data2;
01558 
01559    if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR))
01560       return -1;
01561 
01562    if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR))
01563       return -1;
01564 
01565    if (f1->samples != f2->samples)
01566       return -1;
01567 
01568    for (count = 0, data1 = f1->data.ptr, data2 = f2->data.ptr;
01569         count < f1->samples;
01570         count++, data1++, data2++)
01571       ast_slinear_saturated_add(data1, data2);
01572 
01573    return 0;
01574 }

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(), buf, 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_jb_put(), ast_rtp_write(), ast_slinfactory_feed(), audiohook_read_frame_both(), audiohook_read_frame_single(), autoservice_run(), recordthread(), rpt(), and rpt_exec().

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 = f->subclass;
00523    out->datalen = f->datalen;
00524    out->samples = f->samples;
00525    out->delivery = f->delivery;
00526    /* Set us as having malloc'd header only, so it will eventually
00527       get freed. */
00528    out->mallocd = AST_MALLOCD_HDR;
00529    out->offset = AST_FRIENDLY_OFFSET;
00530    if (out->datalen) {
00531       out->data.ptr = buf + sizeof(*out) + AST_FRIENDLY_OFFSET;
00532       memcpy(out->data.ptr, f->data.ptr, out->datalen);  
00533    } else {
00534       out->data.uint32 = f->data.uint32;
00535    }
00536    if (srclen > 0) {
00537       /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */
00538       char *src;
00539       out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen;
00540       src = (char *) out->src;
00541       /* Must have space since we allocated for it */
00542       strcpy(src, f->src);
00543    }
00544    ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO);
00545    out->ts = f->ts;
00546    out->len = f->len;
00547    out->seqno = f->seqno;
00548    return out;
00549 }

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 389 of file frame.c.

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

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

00390 {
00391    struct ast_frame *out;
00392    void *newdata;
00393 
00394    /* if none of the existing frame is malloc'd, let ast_frdup() do it
00395       since it is more efficient
00396    */
00397    if (fr->mallocd == 0) {
00398       return ast_frdup(fr);
00399    }
00400 
00401    /* if everything is already malloc'd, we are done */
00402    if ((fr->mallocd & (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) ==
00403        (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) {
00404       return fr;
00405    }
00406 
00407    if (!(fr->mallocd & AST_MALLOCD_HDR)) {
00408       /* Allocate a new header if needed */
00409       if (!(out = ast_frame_header_new())) {
00410          return NULL;
00411       }
00412       out->frametype = fr->frametype;
00413       out->subclass = fr->subclass;
00414       out->datalen = fr->datalen;
00415       out->samples = fr->samples;
00416       out->offset = fr->offset;
00417       /* Copy the timing data */
00418       ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO);
00419       if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) {
00420          out->ts = fr->ts;
00421          out->len = fr->len;
00422          out->seqno = fr->seqno;
00423       }
00424    } else {
00425       ast_clear_flag(fr, AST_FRFLAG_FROM_TRANSLATOR);
00426       ast_clear_flag(fr, AST_FRFLAG_FROM_DSP);
00427       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 567 of file frame.c.

References ARRAY_LEN.

00568 {
00569    *size = ARRAY_LEN(AST_FORMAT_LIST);
00570    return AST_FORMAT_LIST;
00571 }

struct ast_format_list* ast_get_format_list_index ( int index ) [read]

Definition at line 562 of file frame.c.

00563 {
00564    return &AST_FORMAT_LIST[idx];
00565 }

int ast_getformatbyname ( const char * name )

Gets a format from a name.

Parameters:

name

string of format

Returns:: This returns the form of the format in binary on success, 0 on error.

Definition at line 636 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().

00637 {
00638    int x, all, format = 0;
00639 
00640    all = strcasecmp(name, "all") ? 0 : 1;
00641    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00642       if (all || 
00643            !strcasecmp(AST_FORMAT_LIST[x].name,name) ||
00644            !strcasecmp(AST_FORMAT_LIST[x].name, ast_expand_codec_alias(name))) {
00645          format |= AST_FORMAT_LIST[x].bits;
00646          if (!all)
00647             break;
00648       }
00649    }
00650 
00651    return format;
00652 }

char* ast_getformatname ( int format )

Get the name of a format.

Parameters:

format

id of format

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

Definition at line 573 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(), _sip_show_peer(), _skinny_show_line(), add_codec_to_answer(), add_codec_to_sdp(), add_tcodec_to_sdp(), add_vcodec_to_sdp(), agent_call(), ast_codec_get_len(), ast_codec_get_samples(), ast_codec_pref_string(), ast_dsp_process(), ast_frame_dump(), ast_openvstream(), ast_rtp_write(), ast_slinfactory_feed(), ast_streamfile(), ast_translator_build_path(), ast_unregister_translator(), ast_writestream(), background_detect_exec(), bridge_channel_join(), bridge_make_compatible(), dahdi_read(), do_waiting(), eagi_exec(), func_channel_read(), function_iaxpeer(), function_sippeer(), gtalk_show_channels(), handle_cli_core_show_file_formats(), handle_cli_core_show_translation(), handle_cli_iax2_show_channels(), handle_cli_iax2_show_peer(), handle_cli_moh_show_classes(), handle_core_show_image_formats(), iax2_request(), iax_show_provisioning(), jingle_show_channels(), login_exec(), moh_release(), oh323_rtp_read(), phone_setup(), print_codec_to_cli(), rebuild_matrix(), register_translator(), set_format(), set_local_capabilities(), set_peer_capabilities(), show_codecs(), sip_request_call(), sip_rtp_read(), socket_process(), start_rtp(), unistim_request(), unistim_rtp_read(), and unistim_write().

00574 {
00575    int x;
00576    char *ret = "unknown";
00577    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00578       if (AST_FORMAT_LIST[x].bits == format) {
00579          ret = AST_FORMAT_LIST[x].name;
00580          break;
00581       }
00582    }
00583    return ret;
00584 }

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

Get the names of a set of formats.

Parameters:

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

Returns:: The return value is buf.

Definition at line 586 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(), ast_streamfile(), bridge_make_compatible(), function_iaxpeer(), function_sippeer(), gtalk_is_answered(), gtalk_newcall(), handle_cli_iax2_show_peer(), handle_showchan(), iax2_bridge(), process_sdp(), serialize_showchan(), set_format(), show_channels_cb(), sip_new(), sip_request_call(), sip_show_channel(), sip_show_settings(), and sip_write().

00587 {
00588    int x;
00589    unsigned len;
00590    char *start, *end = buf;
00591 
00592    if (!size)
00593       return buf;
00594    snprintf(end, size, "0x%x (", format);
00595    len = strlen(end);
00596    end += len;
00597    size -= len;
00598    start = end;
00599    for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) {
00600       if (AST_FORMAT_LIST[x].bits & format) {
00601          snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name);
00602          len = strlen(end);
00603          end += len;
00604          size -= len;
00605       }
00606    }
00607    if (start == end)
00608       ast_copy_string(start, "nothing)", size);
00609    else if (size > 1)
00610       *(end -1) = ')';
00611    return buf;
00612 }

int ast_parse_allow_disallow	(	struct ast_codec_pref *	pref,
		int *	mask,
		const char *	list,
		int	allowing
	)

Parse an "allow" or "deny" line in a channel or device configuration and update the capabilities mask and pref if provided. Video codecs are not added to codec preference lists, since we can not transcode.

Returns:: Returns number of errors encountered during parsing

Definition at line 1233 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(), 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().

01234 {
01235    int errors = 0;
01236    char *parse = NULL, *this = NULL, *psize = NULL;
01237    int format = 0, framems = 0;
01238 
01239    parse = ast_strdupa(list);
01240    while ((this = strsep(&parse, ","))) {
01241       framems = 0;
01242       if ((psize = strrchr(this, ':'))) {
01243          *psize++ = '\0';
01244          ast_debug(1, "Packetization for codec: %s is %s\n", this, psize);
01245          framems = atoi(psize);
01246          if (framems < 0) {
01247             framems = 0;
01248             errors++;
01249             ast_log(LOG_WARNING, "Bad packetization value for codec %s\n", this);
01250          }
01251       }
01252       if (!(format = ast_getformatbyname(this))) {
01253          ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this);
01254          errors++;
01255          continue;
01256       }
01257 
01258       if (mask) {
01259          if (allowing)
01260             *mask |= format;
01261          else
01262             *mask &= ~format;
01263       }
01264 
01265       /* Set up a preference list for audio. Do not include video in preferences 
01266          since we can not transcode video and have to use whatever is offered
01267        */
01268       if (pref && (format & AST_FORMAT_AUDIO_MASK)) {
01269          if (strcasecmp(this, "all")) {
01270             if (allowing) {
01271                ast_codec_pref_append(pref, format);
01272                ast_codec_pref_setsize(pref, format, framems);
01273             }
01274             else
01275                ast_codec_pref_remove(pref, format);
01276          } else if (!allowing) {
01277             memset(pref, 0, sizeof(*pref));
01278          }
01279       }
01280    }
01281    return errors;
01282 }

void ast_smoother_free ( struct ast_smoother * s )

Definition at line 286 of file frame.c.

References ast_free.

Referenced by ast_rtp_destroy(), and ast_rtp_write().

00287 {
00288    ast_free(s);
00289 }

int ast_smoother_get_flags ( struct ast_smoother * smoother )

Definition at line 186 of file frame.c.

References ast_smoother::flags.

00187 {
00188    return s->flags;
00189 }

struct ast_smoother* ast_smoother_new ( int bytes ) [read]

Definition at line 176 of file frame.c.

References ast_malloc, ast_smoother_reset(), and s.

Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().

00177 {
00178    struct ast_smoother *s;
00179    if (size < 1)
00180       return NULL;
00181    if ((s = ast_malloc(sizeof(*s))))
00182       ast_smoother_reset(s, size);
00183    return s;
00184 }

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

Definition at line 236 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_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().

00237 {
00238    struct ast_frame *opt;
00239    int len;
00240 
00241    /* IF we have an optimization frame, send it */
00242    if (s->opt) {
00243       if (s->opt->offset < AST_FRIENDLY_OFFSET)
00244          ast_log(LOG_WARNING, "Returning a frame of inappropriate offset (%d).\n",
00245                      s->opt->offset);
00246       opt = s->opt;
00247       s->opt = NULL;
00248       return opt;
00249    }
00250 
00251    /* Make sure we have enough data */
00252    if (s->len < s->size) {
00253       /* Or, if this is a G.729 frame with VAD on it, send it immediately anyway */
00254       if (!((s->flags & AST_SMOOTHER_FLAG_G729) && (s->len % 10)))
00255          return NULL;
00256    }
00257    len = s->size;
00258    if (len > s->len)
00259       len = s->len;
00260    /* Make frame */
00261    s->f.frametype = AST_FRAME_VOICE;
00262    s->f.subclass = s->format;
00263    s->f.data.ptr = s->framedata + AST_FRIENDLY_OFFSET;
00264    s->f.offset = AST_FRIENDLY_OFFSET;
00265    s->f.datalen = len;
00266    /* Samples will be improper given VAD, but with VAD the concept really doesn't even exist */
00267    s->f.samples = len * s->samplesperbyte;   /* XXX rounding */
00268    s->f.delivery = s->delivery;
00269    /* Fill Data */
00270    memcpy(s->f.data.ptr, s->data, len);
00271    s->len -= len;
00272    /* Move remaining data to the front if applicable */
00273    if (s->len) {
00274       /* In principle this should all be fine because if we are sending
00275          G.729 VAD, the next timestamp will take over anyawy */
00276       memmove(s->data, s->data + len, s->len);
00277       if (!ast_tvzero(s->delivery)) {
00278          /* If we have delivery time, increment it, otherwise, leave it at 0 */
00279          s->delivery = ast_tvadd(s->delivery, ast_samp2tv(s->f.samples, ast_format_rate(s->format)));
00280       }
00281    }
00282    /* Return frame */
00283    return &s->f;
00284 }

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 154 of file frame.c.

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

Referenced by ast_rtp_codec_setpref().

00155 {
00156    /* if there is no change, then nothing to do */
00157    if (s->size == bytes) {
00158       return;
00159    }
00160    /* set the new desired output size */
00161    s->size = bytes;
00162    /* if there is no 'optimized' frame in the smoother,
00163     *   then there is nothing left to do
00164     */
00165    if (!s->opt) {
00166       return;
00167    }
00168    /* there is an 'optimized' frame here at the old size,
00169     * but it must now be put into the buffer so the data
00170     * can be extracted at the new size
00171     */
00172    smoother_frame_feed(s, s->opt, s->opt_needs_swap);
00173    s->opt = NULL;
00174 }

void ast_smoother_reset	(	struct ast_smoother *	s,
		int	bytes
	)

Definition at line 148 of file frame.c.

References ast_smoother::size.

Referenced by ast_smoother_new().

00149 {
00150    memset(s, 0, sizeof(*s));
00151    s->size = bytes;
00152 }

void ast_smoother_set_flags	(	struct ast_smoother *	smoother,
		int	flags
	)

Definition at line 191 of file frame.c.

References ast_smoother::flags.

Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().

00192 {
00193    s->flags = flags;
00194 }

int ast_smoother_test_flag	(	struct ast_smoother *	s,
		int	flag
	)

Definition at line 196 of file frame.c.

References ast_smoother::flags.

Referenced by ast_rtp_write().

00197 {
00198    return (s->flags & flag);
00199 }

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

Definition at line 551 of file frame.c.

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

00552 {
00553    int i;
00554    unsigned short *dst_s = dst;
00555    const unsigned short *src_s = src;
00556 
00557    for (i = 0; i < samples; i++)
00558       dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8);
00559 }

Variable Documentation

struct ast_frame ast_null_frame

Queueing a null frame is fairly common, so we declare a global null frame object for this purpose instead of having to declare one on the stack

Definition at line 124 of file frame.c.

Referenced by __ast_read(), __oh323_rtp_create(), __oh323_update_info(), agent_new(), agent_read(), ast_channel_masquerade(), ast_channel_setwhentohangup_tv(), ast_do_masquerade(), ast_rtcp_read(), ast_rtp_read(), ast_softhangup_nolock(), ast_udptl_read(), bridge_read(), conf_run(), console_read(), dahdi_read(), gtalk_rtp_read(), handle_request_invite(), handle_response_invite(), iax2_read(), jingle_rtp_read(), local_read(), mgcp_rtp_read(), oh323_read(), oh323_rtp_read(), process_rfc2833(), process_sdp(), send_dtmf(), sip_read(), sip_rtp_read(), skinny_rtp_read(), unistim_rtp_read(), and wakeup_sub().

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