Asterisk internal frame definitions. More...
#include <sys/time.h>#include "asterisk/endian.h"#include "asterisk/linkedlists.h"

Go to the source code of this file.
Data Structures | |
| struct | ast_codec_pref |
| struct | ast_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_AUDIO_UNDEFINED ((1 << 13) | (1 << 14)) |
| #define | AST_FORMAT_G722 (1 << 12) |
| #define | AST_FORMAT_G723_1 (1 << 0) |
| #define | AST_FORMAT_G726 (1 << 11) |
| #define | AST_FORMAT_G726_AAL2 (1 << 4) |
| #define | AST_FORMAT_G729A (1 << 8) |
| #define | AST_FORMAT_GSM (1 << 1) |
| #define | AST_FORMAT_H261 (1 << 18) |
| #define | AST_FORMAT_H263 (1 << 19) |
| #define | AST_FORMAT_H263_PLUS (1 << 20) |
| #define | AST_FORMAT_H264 (1 << 21) |
| #define | AST_FORMAT_ILBC (1 << 10) |
| #define | AST_FORMAT_JPEG (1 << 16) |
| #define | AST_FORMAT_LPC10 (1 << 7) |
| #define | AST_FORMAT_MAX_TEXT (1 << 28)) |
| #define | AST_FORMAT_MP4_VIDEO (1 << 22) |
| #define | AST_FORMAT_PNG (1 << 17) |
| #define | AST_FORMAT_SLINEAR (1 << 6) |
| #define | AST_FORMAT_SLINEAR16 (1 << 15) |
| #define | AST_FORMAT_SPEEX (1 << 9) |
| #define | AST_FORMAT_T140 (1 << 27) |
| #define | AST_FORMAT_T140RED (1 << 26) |
| #define | AST_FORMAT_TEXT_MASK (((1 << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK)) |
| #define | AST_FORMAT_ULAW (1 << 2) |
| #define | AST_FORMAT_VIDEO_MASK (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) |
| #define | ast_frame_byteswap_be(fr) do { ; } while(0) |
| #define | ast_frame_byteswap_le(fr) do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data.ptr, __f->data.ptr, __f->samples); } while(0) |
| #define | AST_FRAME_DTMF AST_FRAME_DTMF_END |
| #define | AST_FRAME_SET_BUFFER(fr, _base, _ofs, _datalen) |
| #define | ast_frfree(fr) ast_frame_free(fr, 1) |
| #define | AST_FRIENDLY_OFFSET 64 |
| Offset into a frame's data buffer. | |
| #define | AST_HTML_BEGIN 4 |
| #define | AST_HTML_DATA 2 |
| #define | AST_HTML_END 8 |
| #define | AST_HTML_LDCOMPLETE 16 |
| #define | AST_HTML_LINKREJECT 20 |
| #define | AST_HTML_LINKURL 18 |
| #define | AST_HTML_NOSUPPORT 17 |
| #define | AST_HTML_UNLINK 19 |
| #define | AST_HTML_URL 1 |
| #define | AST_MALLOCD_DATA (1 << 1) |
| #define | AST_MALLOCD_HDR (1 << 0) |
| #define | AST_MALLOCD_SRC (1 << 2) |
| #define | AST_MIN_OFFSET 32 |
| #define | AST_MODEM_T38 1 |
| #define | AST_MODEM_V150 2 |
| #define | AST_OPTION_AUDIO_MODE 4 |
| #define | AST_OPTION_ECHOCAN 8 |
| #define | AST_OPTION_FLAG_ACCEPT 1 |
| #define | AST_OPTION_FLAG_ANSWER 5 |
| #define | AST_OPTION_FLAG_QUERY 4 |
| #define | AST_OPTION_FLAG_REJECT 2 |
| #define | AST_OPTION_FLAG_REQUEST 0 |
| #define | AST_OPTION_FLAG_WTF 6 |
| #define | AST_OPTION_OPRMODE 7 |
| #define | AST_OPTION_RELAXDTMF 3 |
| #define | AST_OPTION_RXGAIN 6 |
| #define | AST_OPTION_T38_STATE 10 |
| #define | AST_OPTION_TDD 2 |
| #define | AST_OPTION_TONE_VERIFY 1 |
| #define | AST_OPTION_TXGAIN 5 |
| #define | AST_SMOOTHER_FLAG_BE (1 << 1) |
| #define | AST_SMOOTHER_FLAG_G729 (1 << 0) |
Enumerations | |
| enum | { AST_FRFLAG_HAS_TIMING_INFO = (1 << 0) } |
| 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 | |
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| |
| #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) |
Asterisk internal frame definitions.
Definition in file frame.h.
| #define AST_FORMAT_ADPCM (1 << 5) |
ADPCM (IMA)
Definition at line 243 of file frame.h.
Referenced by adpcmtolin_sample(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), convertcap(), vox_read(), and vox_write().
| #define AST_FORMAT_ALAW (1 << 3) |
Raw A-law data (G.711)
Definition at line 239 of file frame.h.
Referenced by alawtolin_sample(), alawtoulaw_sample(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), cb_events(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_new(), dahdi_read(), dahdi_write(), find_transcoders(), is_encoder(), misdn_read(), misdn_set_opt_exec(), oh323_rtp_read(), pcm_seek(), pcm_write(), read_config(), and start_rtp().
| #define AST_FORMAT_AUDIO_MASK ((1 << 16)-1) |
Maximum audio mask
Definition at line 263 of file frame.h.
Referenced by add_sdp(), ast_best_codec(), ast_channel_make_compatible_helper(), ast_closestream(), ast_codec_choose(), ast_filehelper(), ast_openstream_full(), ast_openvstream(), ast_parse_allow_disallow(), ast_playstream(), ast_request(), ast_rtp_read(), ast_translate_available_formats(), ast_translator_best_choice(), ast_writestream(), begin_dial_channel(), filestream_destructor(), func_channel_read(), generator_force(), gtalk_rtp_read(), jingle_rtp_read(), oh323_request(), phone_read(), process_sdp(), set_format(), sip_call(), sip_request_call(), sip_rtp_read(), sip_write(), skinny_request(), transmit_connect_with_sdp(), and transmit_modify_with_sdp().
| #define AST_FORMAT_AUDIO_UNDEFINED ((1 << 13) | (1 << 14)) |
| #define AST_FORMAT_G722 (1 << 12) |
G.722
Definition at line 257 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_raw_write(), ast_slinfactory_feed(), au_seek(), convertcap(), g722tolin16_sample(), g722tolin_sample(), pcm_read(), and rtp_get_rate().
| #define AST_FORMAT_G723_1 (1 << 0) |
G.723.1 compression
Definition at line 233 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 255 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_set_rtpmap_type(), g726_read(), g726_write(), and g726tolin_sample().
| #define AST_FORMAT_G726_AAL2 (1 << 4) |
ADPCM (G.726, 32kbps, AAL2 codeword packing)
Definition at line 241 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_lookup_mime_subtype(), ast_rtp_set_rtpmap_type(), codec_ast2skinny(), codec_skinny2ast(), and setup_rtp_connection().
| #define AST_FORMAT_G729A (1 << 8) |
G.729A audio
Definition at line 249 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 235 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), convertcap(), gsm_read(), gsm_write(), gsmtolin_sample(), wav_read(), and wav_write().
| #define AST_FORMAT_H261 (1 << 18) |
H.261 Video
Definition at line 269 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 271 of file frame.h.
Referenced by codec_ast2skinny(), codec_skinny2ast(), h263_encap(), h263_read(), and h263_write().
| #define AST_FORMAT_H263_PLUS (1 << 20) |
| #define AST_FORMAT_H264 (1 << 21) |
H.264 Video
Definition at line 275 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 253 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_codec_interp_len(), convertcap(), ilbc_read(), ilbc_write(), and ilbctolin_sample().
| #define AST_FORMAT_JPEG (1 << 16) |
JPEG Images
Definition at line 265 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 247 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_samples(), and lpc10tolin_sample().
| #define AST_FORMAT_MP4_VIDEO (1 << 22) |
| #define AST_FORMAT_PNG (1 << 17) |
| #define AST_FORMAT_SLINEAR (1 << 6) |
Raw 16-bit Signed Linear (8000 Hz) PCM
Definition at line 245 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_feed(), ast_speech_new(), attempt_reconnect(), audio_audiohook_write_list(), audiohook_read_frame_both(), audiohook_read_frame_single(), background_detect_exec(), build_conf(), chanspy_exec(), conf_run(), connect_link(), dahdi_read(), dahdi_translate(), dahdi_write(), dictate_exec(), do_waiting(), eagi_exec(), extenspy_exec(), fax_generator_generate(), find_transcoders(), handle_jack_audio(), handle_recordfile(), handle_speechcreate(), handle_speechrecognize(), iax_frame_wrap(), ices_exec(), init_outgoing(), is_encoder(), isAnsweringMachine(), jack_exec(), jack_hook_callback(), linear_alloc(), linear_generator(), lintoadpcm_sample(), lintoalaw_sample(), lintog722_sample(), lintog726_sample(), lintogsm_sample(), lintoilbc_sample(), lintolpc10_sample(), lintospeex_sample(), lintoulaw_sample(), load_module(), load_moh_classes(), local_ast_moh_start(), measurenoise(), misdn_set_opt_exec(), mixmonitor_thread(), mp3_exec(), nbs_request(), nbs_xwrite(), NBScat_exec(), ogg_vorbis_read(), ogg_vorbis_write(), oh323_rtp_read(), orig_app(), orig_exten(), oss_new(), oss_read(), oss_request(), parkandannounce_exec(), phone_new(), phone_read(), phone_request(), phone_setup(), phone_write(), playtones_alloc(), playtones_generator(), read_config(), record_exec(), rpt(), rpt_call(), rpt_exec(), rpt_tele_thread(), send_waveform_to_channel(), silence_generator_generate(), slin8_to_slin16_sample(), slinear_read(), slinear_write(), socket_process(), speech_background(), spy_generate(), tonepair_alloc(), 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 261 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_slinfactory_feed(), console_new(), lin16tog722_sample(), slin16_to_slin8_sample(), slinear_read(), slinear_write(), and stream_monitor().
| #define AST_FORMAT_SPEEX (1 << 9) |
SpeeX Free Compression
Definition at line 251 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_samples(), ast_rtp_write(), convertcap(), and speextolin_sample().
| #define AST_FORMAT_T140 (1 << 27) |
T.140 Text format - ITU T.140, RFC 4103
Definition at line 282 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 280 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 285 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 237 of file frame.h.
Referenced by __adsi_transmit_messages(), _ast_adsi_transmit_message_full(), adsi_careful_send(), alarmreceiver_exec(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), calc_energy(), codec_ast2skinny(), codec_skinny2ast(), conf_run(), convertcap(), dahdi_new(), dahdi_read(), dahdi_translate(), dahdi_write(), find_transcoders(), is_encoder(), load_module(), milliwatt_generate(), oh323_rtp_read(), old_milliwatt_exec(), phone_request(), phone_setup(), phone_write(), pri_dchannel(), send_tone_burst(), start_rtp(), ulawtoalaw_sample(), and ulawtolin_sample().
| #define AST_FORMAT_VIDEO_MASK (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) |
Definition at line 278 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 485 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 484 of file frame.h.
Referenced by phone_read().
| #define AST_FRAME_DTMF AST_FRAME_DTMF_END |
Definition at line 124 of file frame.h.
Referenced by __adsi_transmit_messages(), __ast_play_and_record(), action_atxfer(), action_dahdidialoffhook(), agent_ack_sleep(), ast_audiohook_write_list(), ast_bridge_call(), ast_dsp_process(), ast_feature_request_and_dial(), ast_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(), 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 | ) |
{ \
(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 174 of file frame.h.
Referenced by fax_generator_generate(), g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), ilbc_read(), ogg_vorbis_read(), pcm_read(), slinear_read(), t38_tx_packet_handler(), vox_read(), and wav_read().
| #define ast_frfree | ( | fr | ) | ast_frame_free(fr, 1) |
Definition at line 452 of file frame.h.
Referenced by __adsi_transmit_messages(), __ast_answer(), __ast_play_and_record(), __ast_queue_frame(), __ast_read(), __ast_request_and_dial(), adsi_careful_send(), agent_ack_sleep(), agent_read(), ast_audiohook_read_frame(), ast_autoservice_stop(), ast_bridge_call(), ast_channel_free(), ast_dsp_process(), ast_feature_request_and_dial(), ast_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_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(), 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(), process_ast_dsp(), 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 195 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(), slinear_read(), sms_generate(), tonepair_generator(), usbradio_read(), vox_read(), and wav_read().
| #define AST_HTML_BEGIN 4 |
| #define AST_HTML_DATA 2 |
| #define AST_HTML_END 8 |
| #define AST_HTML_LDCOMPLETE 16 |
Load is complete
Definition at line 221 of file frame.h.
Referenced by ast_frame_dump(), and sendurl_exec().
| #define AST_HTML_LINKREJECT 20 |
| #define AST_HTML_LINKURL 18 |
| #define AST_HTML_NOSUPPORT 17 |
Peer is unable to support HTML
Definition at line 223 of file frame.h.
Referenced by ast_frame_dump(), and sendurl_exec().
| #define AST_HTML_UNLINK 19 |
| #define AST_HTML_URL 1 |
Sending a URL
Definition at line 213 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 201 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 199 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 203 of file frame.h.
Referenced by __frame_free(), ast_frisolate(), and speex_callback().
| #define AST_MIN_OFFSET 32 |
Definition at line 196 of file frame.h.
Referenced by __ast_smoother_feed().
| #define AST_MODEM_T38 1 |
T.38 Fax-over-IP
Definition at line 207 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 |
| #define AST_OPTION_AUDIO_MODE 4 |
Set (or clear) Audio (Not-Clear) Mode
Definition at line 366 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 388 of file frame.h.
Referenced by dahdi_setoption().
| #define AST_OPTION_FLAG_REQUEST 0 |
Definition at line 348 of file frame.h.
Referenced by ast_bridge_call(), and iax2_setoption().
| #define AST_OPTION_OPRMODE 7 |
Definition at line 385 of file frame.h.
Referenced by dahdi_setoption(), and dial_exec_full().
| #define AST_OPTION_RELAXDTMF 3 |
Relax the parameters for DTMF reception (mainly for radio use)
Definition at line 363 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 382 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 394 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 360 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 357 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 374 of file frame.h.
Referenced by common_exec(), dahdi_setoption(), func_channel_write(), iax2_setoption(), reset_volumes(), and set_listen_volume().
Definition at line 555 of file frame.h.
Referenced by ast_rtp_write().
Definition at line 560 of file frame.h.
Referenced by ast_rtp_write().
| #define AST_SMOOTHER_FLAG_BE (1 << 1) |
Definition at line 345 of file frame.h.
Referenced by ast_rtp_write().
| #define AST_SMOOTHER_FLAG_G729 (1 << 0) |
Definition at line 344 of file frame.h.
Referenced by __ast_smoother_feed(), ast_smoother_read(), and smoother_frame_feed().
| anonymous enum |
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 };
| 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
|
| AST_CONTROL_SRCUPDATE |
Indicate source of media has changed |
| AST_CONTROL_T38_PARAMETERS |
T38 state change request/notification with parameters |
Definition at line 287 of file frame.h.
00287 { 00288 AST_CONTROL_HANGUP = 1, /*!< Other end has hungup */ 00289 AST_CONTROL_RING = 2, /*!< Local ring */ 00290 AST_CONTROL_RINGING = 3, /*!< Remote end is ringing */ 00291 AST_CONTROL_ANSWER = 4, /*!< Remote end has answered */ 00292 AST_CONTROL_BUSY = 5, /*!< Remote end is busy */ 00293 AST_CONTROL_TAKEOFFHOOK = 6, /*!< Make it go off hook */ 00294 AST_CONTROL_OFFHOOK = 7, /*!< Line is off hook */ 00295 AST_CONTROL_CONGESTION = 8, /*!< Congestion (circuits busy) */ 00296 AST_CONTROL_FLASH = 9, /*!< Flash hook */ 00297 AST_CONTROL_WINK = 10, /*!< Wink */ 00298 AST_CONTROL_OPTION = 11, /*!< Set a low-level option */ 00299 AST_CONTROL_RADIO_KEY = 12, /*!< Key Radio */ 00300 AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */ 00301 AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */ 00302 AST_CONTROL_PROCEEDING = 15, /*!< Indicate CALL PROCEEDING */ 00303 AST_CONTROL_HOLD = 16, /*!< Indicate call is placed on hold */ 00304 AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold */ 00305 AST_CONTROL_VIDUPDATE = 18, /*!< Indicate video frame update */ 00306 _XXX_AST_CONTROL_T38 = 19, /*!< T38 state change request/notification \deprecated This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead. */ 00307 AST_CONTROL_SRCUPDATE = 20, /*!< Indicate source of media has changed */ 00308 AST_CONTROL_T38_PARAMETERS = 24, /*!< T38 state change request/notification with parameters */ 00309 };
| enum ast_control_t38 |
Definition at line 311 of file frame.h.
00311 { 00312 AST_T38_REQUEST_NEGOTIATE = 1, /*!< Request T38 on a channel (voice to fax) */ 00313 AST_T38_REQUEST_TERMINATE, /*!< Terminate T38 on a channel (fax to voice) */ 00314 AST_T38_NEGOTIATED, /*!< T38 negotiated (fax mode) */ 00315 AST_T38_TERMINATED, /*!< T38 terminated (back to voice) */ 00316 AST_T38_REFUSED /*!< T38 refused for some reason (usually rejected by remote end) */ 00317 };
| enum ast_control_t38_rate |
| 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 319 of file frame.h.
00319 { 00320 AST_T38_RATE_2400 = 0, 00321 AST_T38_RATE_4800, 00322 AST_T38_RATE_7200, 00323 AST_T38_RATE_9600, 00324 AST_T38_RATE_12000, 00325 AST_T38_RATE_14400, 00326 };
Definition at line 328 of file frame.h.
00328 { 00329 AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF = 0, 00330 AST_T38_RATE_MANAGEMENT_LOCAL_TCF, 00331 };
| enum ast_frame_type |
Frame types.
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 };
| int __ast_smoother_feed | ( | struct ast_smoother * | s, | |
| struct ast_frame * | f, | |||
| int | swap | |||
| ) |
Definition at line 199 of file frame.c.
References AST_FRAME_VOICE, ast_log(), AST_MIN_OFFSET, AST_SMOOTHER_FLAG_G729, ast_swapcopy_samples(), 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.
00200 { 00201 if (f->frametype != AST_FRAME_VOICE) { 00202 ast_log(LOG_WARNING, "Huh? Can't smooth a non-voice frame!\n"); 00203 return -1; 00204 } 00205 if (!s->format) { 00206 s->format = f->subclass; 00207 s->samplesperbyte = (float)f->samples / (float)f->datalen; 00208 } else if (s->format != f->subclass) { 00209 ast_log(LOG_WARNING, "Smoother was working on %d format frames, now trying to feed %d?\n", s->format, f->subclass); 00210 return -1; 00211 } 00212 if (s->len + f->datalen > SMOOTHER_SIZE) { 00213 ast_log(LOG_WARNING, "Out of smoother space\n"); 00214 return -1; 00215 } 00216 if (((f->datalen == s->size) || 00217 ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) && 00218 !s->opt && 00219 !s->len && 00220 (f->offset >= AST_MIN_OFFSET)) { 00221 /* Optimize by sending the frame we just got 00222 on the next read, thus eliminating the douple 00223 copy */ 00224 if (swap) 00225 ast_swapcopy_samples(f->data.ptr, f->data.ptr, f->samples); 00226 s->opt = f; 00227 s->opt_needs_swap = swap ? 1 : 0; 00228 return 0; 00229 } 00230 00231 return smoother_frame_feed(s, f, swap); 00232 }
| char* ast_codec2str | ( | int | codec | ) |
Get a name from a format Gets a name from a format.
| codec | codec number (1,2,4,8,16,etc.) |
Definition at line 637 of file frame.c.
References ARRAY_LEN, and ast_format_list::desc.
Referenced by moh_alloc(), show_codec_n(), and show_codecs().
00638 { 00639 int x; 00640 char *ret = "unknown"; 00641 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00642 if (AST_FORMAT_LIST[x].bits == codec) { 00643 ret = AST_FORMAT_LIST[x].desc; 00644 break; 00645 } 00646 } 00647 return ret; 00648 }
| 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 1194 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().
01195 { 01196 int x, ret = 0, slot; 01197 01198 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01199 slot = pref->order[x]; 01200 01201 if (!slot) 01202 break; 01203 if (formats & AST_FORMAT_LIST[slot-1].bits) { 01204 ret = AST_FORMAT_LIST[slot-1].bits; 01205 break; 01206 } 01207 } 01208 if (ret & AST_FORMAT_AUDIO_MASK) 01209 return ret; 01210 01211 ast_debug(4, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec"); 01212 01213 return find_best ? ast_best_codec(formats) : 0; 01214 }
| 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 1458 of file frame.c.
References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G722, AST_FORMAT_G723_1, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_ILBC, AST_FORMAT_SLINEAR, AST_FORMAT_SLINEAR16, AST_FORMAT_ULAW, ast_getformatname(), ast_log(), len(), and LOG_WARNING.
Referenced by moh_generate(), and monmp3thread().
01459 { 01460 int len = 0; 01461 01462 /* XXX Still need speex, g723, and lpc10 XXX */ 01463 switch(format) { 01464 case AST_FORMAT_G723_1: 01465 len = (samples / 240) * 20; 01466 break; 01467 case AST_FORMAT_ILBC: 01468 len = (samples / 240) * 50; 01469 break; 01470 case AST_FORMAT_GSM: 01471 len = (samples / 160) * 33; 01472 break; 01473 case AST_FORMAT_G729A: 01474 len = samples / 8; 01475 break; 01476 case AST_FORMAT_SLINEAR: 01477 case AST_FORMAT_SLINEAR16: 01478 len = samples * 2; 01479 break; 01480 case AST_FORMAT_ULAW: 01481 case AST_FORMAT_ALAW: 01482 len = samples; 01483 break; 01484 case AST_FORMAT_G722: 01485 case AST_FORMAT_ADPCM: 01486 case AST_FORMAT_G726: 01487 case AST_FORMAT_G726_AAL2: 01488 len = samples / 2; 01489 break; 01490 default: 01491 ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format)); 01492 } 01493 01494 return len; 01495 }
| int ast_codec_get_samples | ( | struct ast_frame * | f | ) |
Returns the number of samples contained in the frame.
Definition at line 1414 of file frame.c.
References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G722, AST_FORMAT_G723_1, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_ILBC, AST_FORMAT_LPC10, AST_FORMAT_SLINEAR, AST_FORMAT_SLINEAR16, AST_FORMAT_SPEEX, AST_FORMAT_ULAW, ast_getformatname(), ast_log(), 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().
01415 { 01416 int samples=0; 01417 switch(f->subclass) { 01418 case AST_FORMAT_SPEEX: 01419 samples = speex_samples(f->data.ptr, f->datalen); 01420 break; 01421 case AST_FORMAT_G723_1: 01422 samples = g723_samples(f->data.ptr, f->datalen); 01423 break; 01424 case AST_FORMAT_ILBC: 01425 samples = 240 * (f->datalen / 50); 01426 break; 01427 case AST_FORMAT_GSM: 01428 samples = 160 * (f->datalen / 33); 01429 break; 01430 case AST_FORMAT_G729A: 01431 samples = f->datalen * 8; 01432 break; 01433 case AST_FORMAT_SLINEAR: 01434 case AST_FORMAT_SLINEAR16: 01435 samples = f->datalen / 2; 01436 break; 01437 case AST_FORMAT_LPC10: 01438 /* assumes that the RTP packet contains one LPC10 frame */ 01439 samples = 22 * 8; 01440 samples += (((char *)(f->data.ptr))[7] & 0x1) * 8; 01441 break; 01442 case AST_FORMAT_ULAW: 01443 case AST_FORMAT_ALAW: 01444 samples = f->datalen; 01445 break; 01446 case AST_FORMAT_G722: 01447 case AST_FORMAT_ADPCM: 01448 case AST_FORMAT_G726: 01449 case AST_FORMAT_G726_AAL2: 01450 samples = f->datalen * 2; 01451 break; 01452 default: 01453 ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass)); 01454 } 01455 return samples; 01456 }
| static int ast_codec_interp_len | ( | int | format | ) | [inline, static] |
Gets duration in ms of interpolation frame for a format.
Definition at line 643 of file frame.h.
References AST_FORMAT_ILBC.
Referenced by __get_from_jb(), and jb_get_and_deliver().
00644 { 00645 return (format == AST_FORMAT_ILBC) ? 30 : 20; 00646 }
| 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 1054 of file frame.c.
References ARRAY_LEN, ast_codec_pref_remove(), and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow().
01055 { 01056 int x, newindex = 0; 01057 01058 ast_codec_pref_remove(pref, format); 01059 01060 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01061 if (AST_FORMAT_LIST[x].bits == format) { 01062 newindex = x + 1; 01063 break; 01064 } 01065 } 01066 01067 if (newindex) { 01068 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01069 if (!pref->order[x]) { 01070 pref->order[x] = newindex; 01071 break; 01072 } 01073 } 01074 } 01075 01076 return x; 01077 }
| 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 957 of file frame.c.
References ast_codec_pref::order.
Referenced by check_access(), create_addr(), dump_prefs(), and socket_process().
00958 { 00959 int x, differential = (int) 'A', mem; 00960 char *from, *to; 00961 00962 if (right) { 00963 from = pref->order; 00964 to = buf; 00965 mem = size; 00966 } else { 00967 to = pref->order; 00968 from = buf; 00969 mem = 32; 00970 } 00971 00972 memset(to, 0, mem); 00973 for (x = 0; x < 32 ; x++) { 00974 if (!from[x]) 00975 break; 00976 to[x] = right ? (from[x] + differential) : (from[x] - differential); 00977 } 00978 }
| struct ast_format_list ast_codec_pref_getsize | ( | struct ast_codec_pref * | pref, | |
| int | format | |||
| ) | [read] |
Get packet size for codec.
Definition at line 1155 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().
01156 { 01157 int x, idx = -1, framems = 0; 01158 struct ast_format_list fmt = { 0, }; 01159 01160 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01161 if (AST_FORMAT_LIST[x].bits == format) { 01162 fmt = AST_FORMAT_LIST[x]; 01163 idx = x; 01164 break; 01165 } 01166 } 01167 01168 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01169 if (pref->order[x] == (idx + 1)) { 01170 framems = pref->framing[x]; 01171 break; 01172 } 01173 } 01174 01175 /* size validation */ 01176 if (!framems) 01177 framems = AST_FORMAT_LIST[idx].def_ms; 01178 01179 if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */ 01180 framems -= framems % AST_FORMAT_LIST[idx].inc_ms; 01181 01182 if (framems < AST_FORMAT_LIST[idx].min_ms) 01183 framems = AST_FORMAT_LIST[idx].min_ms; 01184 01185 if (framems > AST_FORMAT_LIST[idx].max_ms) 01186 framems = AST_FORMAT_LIST[idx].max_ms; 01187 01188 fmt.cur_ms = framems; 01189 01190 return fmt; 01191 }
| int ast_codec_pref_index | ( | struct ast_codec_pref * | pref, | |
| int | index | |||
| ) |
Codec located at a particular place in the preference index.
Definition at line 1015 of file frame.c.
References ast_format_list::bits, and ast_codec_pref::order.
Referenced by _sip_show_peer(), add_sdp(), ast_codec_pref_string(), function_iaxpeer(), function_sippeer(), gtalk_invite(), handle_cli_iax2_show_peer(), jingle_accept_call(), print_codec_to_cli(), and socket_process().
01016 { 01017 int slot = 0; 01018 01019 if ((idx >= 0) && (idx < sizeof(pref->order))) { 01020 slot = pref->order[idx]; 01021 } 01022 01023 return slot ? AST_FORMAT_LIST[slot - 1].bits : 0; 01024 }
| void ast_codec_pref_init | ( | struct ast_codec_pref * | pref | ) |
Initialize an audio codec preference to "no preference".
| void ast_codec_pref_prepend | ( | struct ast_codec_pref * | pref, | |
| int | format, | |||
| int | only_if_existing | |||
| ) |
Prepend an audio codec to a preference list, removing it first if it was already there.
Definition at line 1080 of file frame.c.
References ARRAY_LEN, ast_codec_pref::framing, and ast_codec_pref::order.
Referenced by create_addr().
01081 { 01082 int x, newindex = 0; 01083 01084 /* First step is to get the codecs "index number" */ 01085 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01086 if (AST_FORMAT_LIST[x].bits == format) { 01087 newindex = x + 1; 01088 break; 01089 } 01090 } 01091 /* Done if its unknown */ 01092 if (!newindex) 01093 return; 01094 01095 /* Now find any existing occurrence, or the end */ 01096 for (x = 0; x < 32; x++) { 01097 if (!pref->order[x] || pref->order[x] == newindex) 01098 break; 01099 } 01100 01101 if (only_if_existing && !pref->order[x]) 01102 return; 01103 01104 /* Move down to make space to insert - either all the way to the end, 01105 or as far as the existing location (which will be overwritten) */ 01106 for (; x > 0; x--) { 01107 pref->order[x] = pref->order[x - 1]; 01108 pref->framing[x] = pref->framing[x - 1]; 01109 } 01110 01111 /* And insert the new entry */ 01112 pref->order[0] = newindex; 01113 pref->framing[0] = 0; /* ? */ 01114 }
| void ast_codec_pref_remove | ( | struct ast_codec_pref * | pref, | |
| int | format | |||
| ) |
Remove audio a codec from a preference list.
Definition at line 1027 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().
01028 { 01029 struct ast_codec_pref oldorder; 01030 int x, y = 0; 01031 int slot; 01032 int size; 01033 01034 if (!pref->order[0]) 01035 return; 01036 01037 memcpy(&oldorder, pref, sizeof(oldorder)); 01038 memset(pref, 0, sizeof(*pref)); 01039 01040 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01041 slot = oldorder.order[x]; 01042 size = oldorder.framing[x]; 01043 if (! slot) 01044 break; 01045 if (AST_FORMAT_LIST[slot-1].bits != format) { 01046 pref->order[y] = slot; 01047 pref->framing[y++] = size; 01048 } 01049 } 01050 01051 }
| int ast_codec_pref_setsize | ( | struct ast_codec_pref * | pref, | |
| int | format, | |||
| int | framems | |||
| ) |
Set packet size for codec.
Definition at line 1117 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().
01118 { 01119 int x, idx = -1; 01120 01121 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01122 if (AST_FORMAT_LIST[x].bits == format) { 01123 idx = x; 01124 break; 01125 } 01126 } 01127 01128 if (idx < 0) 01129 return -1; 01130 01131 /* size validation */ 01132 if (!framems) 01133 framems = AST_FORMAT_LIST[idx].def_ms; 01134 01135 if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */ 01136 framems -= framems % AST_FORMAT_LIST[idx].inc_ms; 01137 01138 if (framems < AST_FORMAT_LIST[idx].min_ms) 01139 framems = AST_FORMAT_LIST[idx].min_ms; 01140 01141 if (framems > AST_FORMAT_LIST[idx].max_ms) 01142 framems = AST_FORMAT_LIST[idx].max_ms; 01143 01144 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01145 if (pref->order[x] == (idx + 1)) { 01146 pref->framing[x] = framems; 01147 break; 01148 } 01149 } 01150 01151 return x; 01152 }
| 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 980 of file frame.c.
References ast_codec_pref_index(), and ast_getformatname().
Referenced by dump_prefs(), and socket_process().
00981 { 00982 int x, codec; 00983 size_t total_len, slen; 00984 char *formatname; 00985 00986 memset(buf,0,size); 00987 total_len = size; 00988 buf[0] = '('; 00989 total_len--; 00990 for(x = 0; x < 32 ; x++) { 00991 if (total_len <= 0) 00992 break; 00993 if (!(codec = ast_codec_pref_index(pref,x))) 00994 break; 00995 if ((formatname = ast_getformatname(codec))) { 00996 slen = strlen(formatname); 00997 if (slen > total_len) 00998 break; 00999 strncat(buf, formatname, total_len - 1); /* safe */ 01000 total_len -= slen; 01001 } 01002 if (total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) { 01003 strncat(buf, "|", total_len - 1); /* safe */ 01004 total_len--; 01005 } 01006 } 01007 if (total_len) { 01008 strncat(buf, ")", total_len - 1); /* safe */ 01009 total_len--; 01010 } 01011 01012 return size - total_len; 01013 }
| static force_inline int ast_format_rate | ( | int | format | ) | [static] |
Get the sample rate for a given format.
Definition at line 670 of file frame.h.
References AST_FORMAT_G722, and AST_FORMAT_SLINEAR16.
Referenced by __get_from_jb(), ast_read_generator_actions(), ast_readaudio_callback(), ast_readvideo_callback(), ast_rtp_read(), ast_smoother_read(), ast_translate(), calc_cost(), calc_timestamp(), generator_force(), rtp_get_rate(), and schedule_delivery().
00671 { 00672 if (format == AST_FORMAT_G722 || format == AST_FORMAT_SLINEAR16) 00673 return 16000; 00674 00675 return 8000; 00676 }
| int ast_frame_adjust_volume | ( | struct ast_frame * | f, | |
| int | adjustment | |||
| ) |
Adjusts the volume of the audio samples contained in a frame.
| 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. |
Definition at line 1497 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().
01498 { 01499 int count; 01500 short *fdata = f->data.ptr; 01501 short adjust_value = abs(adjustment); 01502 01503 if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR)) 01504 return -1; 01505 01506 if (!adjustment) 01507 return 0; 01508 01509 for (count = 0; count < f->samples; count++) { 01510 if (adjustment > 0) { 01511 ast_slinear_saturated_multiply(&fdata[count], &adjust_value); 01512 } else if (adjustment < 0) { 01513 ast_slinear_saturated_divide(&fdata[count], &adjust_value); 01514 } 01515 } 01516 01517 return 0; 01518 }
| void ast_frame_dump | ( | const char * | name, | |
| struct ast_frame * | f, | |||
| char * | prefix | |||
| ) |
Dump a frame for debugging purposes
Definition at line 739 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().
00740 { 00741 const char noname[] = "unknown"; 00742 char ftype[40] = "Unknown Frametype"; 00743 char cft[80]; 00744 char subclass[40] = "Unknown Subclass"; 00745 char csub[80]; 00746 char moreinfo[40] = ""; 00747 char cn[60]; 00748 char cp[40]; 00749 char cmn[40]; 00750 const char *message = "Unknown"; 00751 00752 if (!name) 00753 name = noname; 00754 00755 00756 if (!f) { 00757 ast_verbose("%s [ %s (NULL) ] [%s]\n", 00758 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00759 term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00760 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00761 return; 00762 } 00763 /* XXX We should probably print one each of voice and video when the format changes XXX */ 00764 if (f->frametype == AST_FRAME_VOICE) 00765 return; 00766 if (f->frametype == AST_FRAME_VIDEO) 00767 return; 00768 switch(f->frametype) { 00769 case AST_FRAME_DTMF_BEGIN: 00770 strcpy(ftype, "DTMF Begin"); 00771 subclass[0] = f->subclass; 00772 subclass[1] = '\0'; 00773 break; 00774 case AST_FRAME_DTMF_END: 00775 strcpy(ftype, "DTMF End"); 00776 subclass[0] = f->subclass; 00777 subclass[1] = '\0'; 00778 break; 00779 case AST_FRAME_CONTROL: 00780 strcpy(ftype, "Control"); 00781 switch(f->subclass) { 00782 case AST_CONTROL_HANGUP: 00783 strcpy(subclass, "Hangup"); 00784 break; 00785 case AST_CONTROL_RING: 00786 strcpy(subclass, "Ring"); 00787 break; 00788 case AST_CONTROL_RINGING: 00789 strcpy(subclass, "Ringing"); 00790 break; 00791 case AST_CONTROL_ANSWER: 00792 strcpy(subclass, "Answer"); 00793 break; 00794 case AST_CONTROL_BUSY: 00795 strcpy(subclass, "Busy"); 00796 break; 00797 case AST_CONTROL_TAKEOFFHOOK: 00798 strcpy(subclass, "Take Off Hook"); 00799 break; 00800 case AST_CONTROL_OFFHOOK: 00801 strcpy(subclass, "Line Off Hook"); 00802 break; 00803 case AST_CONTROL_CONGESTION: 00804 strcpy(subclass, "Congestion"); 00805 break; 00806 case AST_CONTROL_FLASH: 00807 strcpy(subclass, "Flash"); 00808 break; 00809 case AST_CONTROL_WINK: 00810 strcpy(subclass, "Wink"); 00811 break; 00812 case AST_CONTROL_OPTION: 00813 strcpy(subclass, "Option"); 00814 break; 00815 case AST_CONTROL_RADIO_KEY: 00816 strcpy(subclass, "Key Radio"); 00817 break; 00818 case AST_CONTROL_RADIO_UNKEY: 00819 strcpy(subclass, "Unkey Radio"); 00820 break; 00821 case AST_CONTROL_HOLD: 00822 strcpy(subclass, "Hold"); 00823 break; 00824 case AST_CONTROL_UNHOLD: 00825 strcpy(subclass, "Unhold"); 00826 break; 00827 case AST_CONTROL_T38_PARAMETERS: 00828 if (f->datalen != sizeof(struct ast_control_t38_parameters)) { 00829 message = "Invalid"; 00830 } else { 00831 struct ast_control_t38_parameters *parameters = f->data.ptr; 00832 enum ast_control_t38 state = parameters->request_response; 00833 if (state == AST_T38_REQUEST_NEGOTIATE) 00834 message = "Negotiation Requested"; 00835 else if (state == AST_T38_REQUEST_TERMINATE) 00836 message = "Negotiation Request Terminated"; 00837 else if (state == AST_T38_NEGOTIATED) 00838 message = "Negotiated"; 00839 else if (state == AST_T38_TERMINATED) 00840 message = "Terminated"; 00841 else if (state == AST_T38_REFUSED) 00842 message = "Refused"; 00843 } 00844 snprintf(subclass, sizeof(subclass), "T38_Parameters/%s", message); 00845 break; 00846 case -1: 00847 strcpy(subclass, "Stop generators"); 00848 break; 00849 default: 00850 snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass); 00851 } 00852 break; 00853 case AST_FRAME_NULL: 00854 strcpy(ftype, "Null Frame"); 00855 strcpy(subclass, "N/A"); 00856 break; 00857 case AST_FRAME_IAX: 00858 /* Should never happen */ 00859 strcpy(ftype, "IAX Specific"); 00860 snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass); 00861 break; 00862 case AST_FRAME_TEXT: 00863 strcpy(ftype, "Text"); 00864 strcpy(subclass, "N/A"); 00865 ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); 00866 break; 00867 case AST_FRAME_IMAGE: 00868 strcpy(ftype, "Image"); 00869 snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass)); 00870 break; 00871 case AST_FRAME_HTML: 00872 strcpy(ftype, "HTML"); 00873 switch(f->subclass) { 00874 case AST_HTML_URL: 00875 strcpy(subclass, "URL"); 00876 ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); 00877 break; 00878 case AST_HTML_DATA: 00879 strcpy(subclass, "Data"); 00880 break; 00881 case AST_HTML_BEGIN: 00882 strcpy(subclass, "Begin"); 00883 break; 00884 case AST_HTML_END: 00885 strcpy(subclass, "End"); 00886 break; 00887 case AST_HTML_LDCOMPLETE: 00888 strcpy(subclass, "Load Complete"); 00889 break; 00890 case AST_HTML_NOSUPPORT: 00891 strcpy(subclass, "No Support"); 00892 break; 00893 case AST_HTML_LINKURL: 00894 strcpy(subclass, "Link URL"); 00895 ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); 00896 break; 00897 case AST_HTML_UNLINK: 00898 strcpy(subclass, "Unlink"); 00899 break; 00900 case AST_HTML_LINKREJECT: 00901 strcpy(subclass, "Link Reject"); 00902 break; 00903 default: 00904 snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass); 00905 break; 00906 } 00907 break; 00908 case AST_FRAME_MODEM: 00909 strcpy(ftype, "Modem"); 00910 switch (f->subclass) { 00911 case AST_MODEM_T38: 00912 strcpy(subclass, "T.38"); 00913 break; 00914 case AST_MODEM_V150: 00915 strcpy(subclass, "V.150"); 00916 break; 00917 default: 00918 snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass); 00919 break; 00920 } 00921 break; 00922 default: 00923 snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype); 00924 } 00925 if (!ast_strlen_zero(moreinfo)) 00926 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n", 00927 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00928 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00929 f->frametype, 00930 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00931 f->subclass, 00932 term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)), 00933 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00934 else 00935 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n", 00936 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00937 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00938 f->frametype, 00939 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00940 f->subclass, 00941 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00942 }
| 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.
| 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
| fr | Frame to free, or head of list to free | |
| cache | Whether to consider this frame for frame caching |
Definition at line 365 of file frame.c.
References __frame_free(), and AST_LIST_NEXT.
Referenced by mixmonitor_thread().
00366 { 00367 struct ast_frame *next; 00368 00369 for (next = AST_LIST_NEXT(frame, frame_list); 00370 frame; 00371 frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) { 00372 __frame_free(frame, cache); 00373 } 00374 }
Sums two frames of audio samples.
| f1 | The first frame (which will contain the result) | |
| f2 | The second frame |
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 1520 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.
01521 { 01522 int count; 01523 short *data1, *data2; 01524 01525 if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR)) 01526 return -1; 01527 01528 if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR)) 01529 return -1; 01530 01531 if (f1->samples != f2->samples) 01532 return -1; 01533 01534 for (count = 0, data1 = f1->data.ptr, data2 = f2->data.ptr; 01535 count < f1->samples; 01536 count++, data1++, data2++) 01537 ast_slinear_saturated_add(data1, data2); 01538 01539 return 0; 01540 }
Copies a frame.
| fr | frame to copy Duplicates a frame -- should only rarely be used, typically frisolate is good enough |
Definition at line 459 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().
00460 { 00461 struct ast_frame *out = NULL; 00462 int len, srclen = 0; 00463 void *buf = NULL; 00464 00465 #if !defined(LOW_MEMORY) 00466 struct ast_frame_cache *frames; 00467 #endif 00468 00469 /* Start with standard stuff */ 00470 len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00471 /* If we have a source, add space for it */ 00472 /* 00473 * XXX Watch out here - if we receive a src which is not terminated 00474 * properly, we can be easily attacked. Should limit the size we deal with. 00475 */ 00476 if (f->src) 00477 srclen = strlen(f->src); 00478 if (srclen > 0) 00479 len += srclen + 1; 00480 00481 #if !defined(LOW_MEMORY) 00482 if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) { 00483 AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) { 00484 if (out->mallocd_hdr_len >= len) { 00485 size_t mallocd_len = out->mallocd_hdr_len; 00486 00487 AST_LIST_REMOVE_CURRENT(frame_list); 00488 memset(out, 0, sizeof(*out)); 00489 out->mallocd_hdr_len = mallocd_len; 00490 buf = out; 00491 frames->size--; 00492 break; 00493 } 00494 } 00495 AST_LIST_TRAVERSE_SAFE_END; 00496 } 00497 #endif 00498 00499 if (!buf) { 00500 if (!(buf = ast_calloc_cache(1, len))) 00501 return NULL; 00502 out = buf; 00503 out->mallocd_hdr_len = len; 00504 } 00505 00506 out->frametype = f->frametype; 00507 out->subclass = f->subclass; 00508 out->datalen = f->datalen; 00509 out->samples = f->samples; 00510 out->delivery = f->delivery; 00511 /* Set us as having malloc'd header only, so it will eventually 00512 get freed. */ 00513 out->mallocd = AST_MALLOCD_HDR; 00514 out->offset = AST_FRIENDLY_OFFSET; 00515 if (out->datalen) { 00516 out->data.ptr = buf + sizeof(*out) + AST_FRIENDLY_OFFSET; 00517 memcpy(out->data.ptr, f->data.ptr, out->datalen); 00518 } else { 00519 out->data.uint32 = f->data.uint32; 00520 } 00521 if (srclen > 0) { 00522 /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */ 00523 char *src; 00524 out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00525 src = (char *) out->src; 00526 /* Must have space since we allocated for it */ 00527 strcpy(src, f->src); 00528 } 00529 ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO); 00530 out->ts = f->ts; 00531 out->len = f->len; 00532 out->seqno = f->seqno; 00533 return out; 00534 }
Makes a frame independent of any static storage.
| 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. |
Definition at line 381 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::data, ast_frame::datalen, ast_frame::frametype, ast_frame::len, ast_frame::mallocd, ast_frame::offset, ast_frame::ptr, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, and ast_frame::ts.
Referenced by __ast_answer(), ast_dsp_process(), ast_slinfactory_feed(), ast_trans_frameout(), ast_write(), autoservice_run(), dahdi_decoder_frameout(), dahdi_encoder_frameout(), jpeg_read_image(), and read_frame().
00382 { 00383 struct ast_frame *out; 00384 void *newdata; 00385 00386 /* if none of the existing frame is malloc'd, let ast_frdup() do it 00387 since it is more efficient 00388 */ 00389 if (fr->mallocd == 0) { 00390 return ast_frdup(fr); 00391 } 00392 00393 /* if everything is already malloc'd, we are done */ 00394 if ((fr->mallocd & (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) == 00395 (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) { 00396 return fr; 00397 } 00398 00399 if (!(fr->mallocd & AST_MALLOCD_HDR)) { 00400 /* Allocate a new header if needed */ 00401 if (!(out = ast_frame_header_new())) { 00402 return NULL; 00403 } 00404 out->frametype = fr->frametype; 00405 out->subclass = fr->subclass; 00406 out->datalen = fr->datalen; 00407 out->samples = fr->samples; 00408 out->offset = fr->offset; 00409 /* Copy the timing data */ 00410 ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO); 00411 if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) { 00412 out->ts = fr->ts; 00413 out->len = fr->len; 00414 out->seqno = fr->seqno; 00415 } 00416 } else { 00417 out = fr; 00418 } 00419 00420 if (!(fr->mallocd & AST_MALLOCD_SRC) && fr->src) { 00421 if (!(out->src = ast_strdup(fr->src))) { 00422 if (out != fr) { 00423 ast_free(out); 00424 } 00425 return NULL; 00426 } 00427 } else { 00428 out->src = fr->src; 00429 fr->src = NULL; 00430 fr->mallocd &= ~AST_MALLOCD_SRC; 00431 } 00432 00433 if (!(fr->mallocd & AST_MALLOCD_DATA)) { 00434 if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) { 00435 if (out->src != fr->src) { 00436 ast_free((void *) out->src); 00437 } 00438 if (out != fr) { 00439 ast_free(out); 00440 } 00441 return NULL; 00442 } 00443 newdata += AST_FRIENDLY_OFFSET; 00444 out->offset = AST_FRIENDLY_OFFSET; 00445 out->datalen = fr->datalen; 00446 memcpy(newdata, fr->data.ptr, fr->datalen); 00447 out->data.ptr = newdata; 00448 } else { 00449 out->data = fr->data; 00450 memset(&fr->data, 0, sizeof(fr->data)); 00451 fr->mallocd &= ~AST_MALLOCD_DATA; 00452 } 00453 00454 out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA; 00455 00456 return out; 00457 }
| struct ast_format_list* ast_get_format_list | ( | size_t * | size | ) | [read] |
Definition at line 552 of file frame.c.
References ARRAY_LEN.
00553 { 00554 *size = ARRAY_LEN(AST_FORMAT_LIST); 00555 return AST_FORMAT_LIST; 00556 }
| struct ast_format_list* ast_get_format_list_index | ( | int | index | ) | [read] |
Definition at line 547 of file frame.c.
00548 { 00549 return &AST_FORMAT_LIST[idx]; 00550 }
| int ast_getformatbyname | ( | const char * | name | ) |
Gets a format from a name.
| name | string of format |
Definition at line 619 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().
00620 { 00621 int x, all, format = 0; 00622 00623 all = strcasecmp(name, "all") ? 0 : 1; 00624 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00625 if (all || 00626 !strcasecmp(AST_FORMAT_LIST[x].name,name) || 00627 !strcasecmp(AST_FORMAT_LIST[x].name, ast_expand_codec_alias(name))) { 00628 format |= AST_FORMAT_LIST[x].bits; 00629 if (!all) 00630 break; 00631 } 00632 } 00633 00634 return format; 00635 }
| char* ast_getformatname | ( | int | format | ) |
Get the name of a format.
| format | id of format |
Definition at line 558 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(), add_codec_to_answer(), add_codec_to_sdp(), add_tcodec_to_sdp(), add_vcodec_to_sdp(), agent_call(), ast_codec_get_len(), ast_codec_get_samples(), ast_codec_pref_string(), ast_dsp_process(), ast_frame_dump(), ast_openvstream(), ast_rtp_write(), ast_slinfactory_feed(), ast_streamfile(), ast_translator_build_path(), ast_unregister_translator(), ast_writestream(), background_detect_exec(), dahdi_read(), do_waiting(), eagi_exec(), func_channel_read(), function_iaxpeer(), function_sippeer(), gtalk_show_channels(), handle_cli_core_show_file_formats(), handle_cli_core_show_translation(), handle_cli_iax2_show_channels(), handle_cli_iax2_show_peer(), handle_cli_moh_show_classes(), handle_core_show_image_formats(), iax2_request(), iax_show_provisioning(), jingle_show_channels(), login_exec(), moh_release(), oh323_rtp_read(), phone_setup(), print_codec_to_cli(), rebuild_matrix(), register_translator(), set_format(), set_local_capabilities(), set_peer_capabilities(), show_codecs(), sip_request_call(), sip_rtp_read(), socket_process(), start_rtp(), unistim_request(), unistim_rtp_read(), and unistim_write().
00559 { 00560 int x; 00561 char *ret = "unknown"; 00562 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00563 if (AST_FORMAT_LIST[x].bits == format) { 00564 ret = AST_FORMAT_LIST[x].name; 00565 break; 00566 } 00567 } 00568 return ret; 00569 }
| char* ast_getformatname_multiple | ( | char * | buf, | |
| size_t | size, | |||
| int | format | |||
| ) |
Get the names of a set of formats.
| 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)" |
Definition at line 571 of file frame.c.
References ARRAY_LEN, ast_copy_string(), ast_format_list::bits, len(), and name.
Referenced by __ast_read(), _sip_show_peer(), add_sdp(), ast_streamfile(), function_iaxpeer(), function_sippeer(), gtalk_is_answered(), gtalk_newcall(), handle_cli_iax2_show_peer(), handle_showchan(), handle_skinny_show_line(), 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().
00572 { 00573 int x; 00574 unsigned len; 00575 char *start, *end = buf; 00576 00577 if (!size) 00578 return buf; 00579 snprintf(end, size, "0x%x (", format); 00580 len = strlen(end); 00581 end += len; 00582 size -= len; 00583 start = end; 00584 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00585 if (AST_FORMAT_LIST[x].bits & format) { 00586 snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name); 00587 len = strlen(end); 00588 end += len; 00589 size -= len; 00590 } 00591 } 00592 if (start == end) 00593 ast_copy_string(start, "nothing)", size); 00594 else if (size > 1) 00595 *(end -1) = ')'; 00596 return buf; 00597 }
| 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.
Definition at line 1216 of file frame.c.
References ast_codec_pref_append(), ast_codec_pref_remove(), ast_codec_pref_setsize(), ast_debug, AST_FORMAT_AUDIO_MASK, ast_getformatbyname(), ast_log(), ast_strdupa, format, LOG_WARNING, parse(), and strsep().
Referenced by action_originate(), apply_outgoing(), build_device(), build_peer(), build_user(), gtalk_create_member(), gtalk_load_config(), jingle_create_member(), jingle_load_config(), reload_config(), set_config(), and update_common_options().
01217 { 01218 int errors = 0; 01219 char *parse = NULL, *this = NULL, *psize = NULL; 01220 int format = 0, framems = 0; 01221 01222 parse = ast_strdupa(list); 01223 while ((this = strsep(&parse, ","))) { 01224 framems = 0; 01225 if ((psize = strrchr(this, ':'))) { 01226 *psize++ = '\0'; 01227 ast_debug(1, "Packetization for codec: %s is %s\n", this, psize); 01228 framems = atoi(psize); 01229 if (framems < 0) { 01230 framems = 0; 01231 errors++; 01232 ast_log(LOG_WARNING, "Bad packetization value for codec %s\n", this); 01233 } 01234 } 01235 if (!(format = ast_getformatbyname(this))) { 01236 ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this); 01237 errors++; 01238 continue; 01239 } 01240 01241 if (mask) { 01242 if (allowing) 01243 *mask |= format; 01244 else 01245 *mask &= ~format; 01246 } 01247 01248 /* Set up a preference list for audio. Do not include video in preferences 01249 since we can not transcode video and have to use whatever is offered 01250 */ 01251 if (pref && (format & AST_FORMAT_AUDIO_MASK)) { 01252 if (strcasecmp(this, "all")) { 01253 if (allowing) { 01254 ast_codec_pref_append(pref, format); 01255 ast_codec_pref_setsize(pref, format, framems); 01256 } 01257 else 01258 ast_codec_pref_remove(pref, format); 01259 } else if (!allowing) { 01260 memset(pref, 0, sizeof(*pref)); 01261 } 01262 } 01263 } 01264 return errors; 01265 }
| void ast_smoother_free | ( | struct ast_smoother * | s | ) |
Definition at line 284 of file frame.c.
References ast_free.
Referenced by ast_rtp_destroy(), and ast_rtp_write().
00285 { 00286 ast_free(s); 00287 }
| int ast_smoother_get_flags | ( | struct ast_smoother * | smoother | ) |
Definition at line 184 of file frame.c.
References ast_smoother::flags.
00185 { 00186 return s->flags; 00187 }
| struct ast_smoother* ast_smoother_new | ( | int | bytes | ) | [read] |
Definition at line 174 of file frame.c.
References ast_malloc, ast_smoother_reset(), and s.
Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().
00175 { 00176 struct ast_smoother *s; 00177 if (size < 1) 00178 return NULL; 00179 if ((s = ast_malloc(sizeof(*s)))) 00180 ast_smoother_reset(s, size); 00181 return s; 00182 }
| struct ast_frame* ast_smoother_read | ( | struct ast_smoother * | s | ) | [read] |
Definition at line 234 of file frame.c.
References ast_format_rate(), AST_FRAME_VOICE, AST_FRIENDLY_OFFSET, ast_log(), ast_samp2tv(), AST_SMOOTHER_FLAG_G729, ast_tvadd(), ast_tvzero(), 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().
00235 { 00236 struct ast_frame *opt; 00237 int len; 00238 00239 /* IF we have an optimization frame, send it */ 00240 if (s->opt) { 00241 if (s->opt->offset < AST_FRIENDLY_OFFSET) 00242 ast_log(LOG_WARNING, "Returning a frame of inappropriate offset (%d).\n", 00243 s->opt->offset); 00244 opt = s->opt; 00245 s->opt = NULL; 00246 return opt; 00247 } 00248 00249 /* Make sure we have enough data */ 00250 if (s->len < s->size) { 00251 /* Or, if this is a G.729 frame with VAD on it, send it immediately anyway */ 00252 if (!((s->flags & AST_SMOOTHER_FLAG_G729) && (s->len % 10))) 00253 return NULL; 00254 } 00255 len = s->size; 00256 if (len > s->len) 00257 len = s->len; 00258 /* Make frame */ 00259 s->f.frametype = AST_FRAME_VOICE; 00260 s->f.subclass = s->format; 00261 s->f.data.ptr = s->framedata + AST_FRIENDLY_OFFSET; 00262 s->f.offset = AST_FRIENDLY_OFFSET; 00263 s->f.datalen = len; 00264 /* Samples will be improper given VAD, but with VAD the concept really doesn't even exist */ 00265 s->f.samples = len * s->samplesperbyte; /* XXX rounding */ 00266 s->f.delivery = s->delivery; 00267 /* Fill Data */ 00268 memcpy(s->f.data.ptr, s->data, len); 00269 s->len -= len; 00270 /* Move remaining data to the front if applicable */ 00271 if (s->len) { 00272 /* In principle this should all be fine because if we are sending 00273 G.729 VAD, the next timestamp will take over anyawy */ 00274 memmove(s->data, s->data + len, s->len); 00275 if (!ast_tvzero(s->delivery)) { 00276 /* If we have delivery time, increment it, otherwise, leave it at 0 */ 00277 s->delivery = ast_tvadd(s->delivery, ast_samp2tv(s->f.samples, ast_format_rate(s->format))); 00278 } 00279 } 00280 /* Return frame */ 00281 return &s->f; 00282 }
| void ast_smoother_reconfigure | ( | struct ast_smoother * | s, | |
| int | bytes | |||
| ) |
Reconfigure an existing smoother to output a different number of bytes per frame.
| s | the smoother to reconfigure | |
| bytes | the desired number of bytes per output frame |
Definition at line 152 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().
00153 { 00154 /* if there is no change, then nothing to do */ 00155 if (s->size == bytes) { 00156 return; 00157 } 00158 /* set the new desired output size */ 00159 s->size = bytes; 00160 /* if there is no 'optimized' frame in the smoother, 00161 * then there is nothing left to do 00162 */ 00163 if (!s->opt) { 00164 return; 00165 } 00166 /* there is an 'optimized' frame here at the old size, 00167 * but it must now be put into the buffer so the data 00168 * can be extracted at the new size 00169 */ 00170 smoother_frame_feed(s, s->opt, s->opt_needs_swap); 00171 s->opt = NULL; 00172 }
| void ast_smoother_reset | ( | struct ast_smoother * | s, | |
| int | bytes | |||
| ) |
Definition at line 146 of file frame.c.
References ast_smoother::size.
Referenced by ast_smoother_new().
00147 { 00148 memset(s, 0, sizeof(*s)); 00149 s->size = bytes; 00150 }
| void ast_smoother_set_flags | ( | struct ast_smoother * | smoother, | |
| int | flags | |||
| ) |
Definition at line 189 of file frame.c.
References ast_smoother::flags.
Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().
| int ast_smoother_test_flag | ( | struct ast_smoother * | s, | |
| int | flag | |||
| ) |
Definition at line 194 of file frame.c.
References ast_smoother::flags.
Referenced by ast_rtp_write().
00195 { 00196 return (s->flags & flag); 00197 }
| void ast_swapcopy_samples | ( | void * | dst, | |
| const void * | src, | |||
| int | samples | |||
| ) |
Definition at line 536 of file frame.c.
Referenced by __ast_smoother_feed(), iax_frame_wrap(), phone_write_buf(), and smoother_frame_feed().
| struct ast_frame ast_null_frame |
Queueing a null frame is fairly common, so we declare a global null frame object for this purpose instead of having to declare one on the stack
Definition at line 122 of file frame.c.
Referenced by __ast_read(), __oh323_rtp_create(), __oh323_update_info(), agent_new(), agent_read(), ast_channel_masquerade(), ast_channel_setwhentohangup_tv(), ast_do_masquerade(), ast_rtcp_read(), ast_rtp_read(), ast_softhangup_nolock(), ast_udptl_read(), conf_run(), console_read(), gtalk_rtp_read(), handle_request_invite(), handle_response_invite(), iax2_read(), jingle_rtp_read(), local_read(), mgcp_rtp_read(), oh323_read(), oh323_rtp_read(), process_rfc2833(), process_sdp(), send_dtmf(), sip_read(), sip_rtp_read(), skinny_rtp_read(), unistim_rtp_read(), and wakeup_sub().
1.6.1