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), AST_FRFLAG_FROM_TRANSLATOR = (1 << 1), AST_FRFLAG_FROM_DSP = (1 << 2), AST_FRFLAG_FROM_FILESTREAM = (1 << 3) } |
| enum | ast_control_frame_type { AST_CONTROL_HANGUP = 1, AST_CONTROL_RING = 2, AST_CONTROL_RINGING = 3, AST_CONTROL_ANSWER = 4, AST_CONTROL_BUSY = 5, AST_CONTROL_TAKEOFFHOOK = 6, AST_CONTROL_OFFHOOK = 7, AST_CONTROL_CONGESTION = 8, AST_CONTROL_FLASH = 9, AST_CONTROL_WINK = 10, AST_CONTROL_OPTION = 11, AST_CONTROL_RADIO_KEY = 12, AST_CONTROL_RADIO_UNKEY = 13, AST_CONTROL_PROGRESS = 14, AST_CONTROL_PROCEEDING = 15, AST_CONTROL_HOLD = 16, AST_CONTROL_UNHOLD = 17, AST_CONTROL_VIDUPDATE = 18, _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) |
Asterisk internal frame definitions.
Definition in file frame.h.
| #define AST_FORMAT_ADPCM (1 << 5) |
ADPCM (IMA)
Definition at line 255 of file frame.h.
Referenced by adpcmtolin_sample(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), convertcap(), vox_read(), and vox_write().
| #define AST_FORMAT_ALAW (1 << 3) |
Raw A-law data (G.711)
Definition at line 251 of file frame.h.
Referenced by alawtolin_sample(), alawtoulaw_sample(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), cb_events(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_new(), dahdi_read(), dahdi_write(), find_transcoders(), is_encoder(), misdn_read(), misdn_set_opt_exec(), oh323_rtp_read(), pcm_seek(), pcm_write(), read_config(), and start_rtp().
| #define AST_FORMAT_AUDIO_MASK ((1 << 16)-1) |
Maximum audio mask
Definition at line 275 of file frame.h.
Referenced by add_sdp(), ast_best_codec(), ast_channel_make_compatible_helper(), ast_codec_choose(), ast_filehelper(), ast_openstream_full(), ast_openvstream(), ast_parse_allow_disallow(), ast_playstream(), ast_request(), ast_rtp_read(), ast_translate_available_formats(), ast_translator_best_choice(), ast_writestream(), begin_dial_channel(), filestream_destructor(), func_channel_read(), generator_force(), gtalk_rtp_read(), jingle_rtp_read(), oh323_request(), phone_read(), process_sdp(), set_format(), sip_call(), sip_request_call(), sip_rtp_read(), sip_write(), skinny_request(), transmit_connect_with_sdp(), and transmit_modify_with_sdp().
| #define AST_FORMAT_AUDIO_UNDEFINED ((1 << 13) | (1 << 14)) |
| #define AST_FORMAT_G722 (1 << 12) |
G.722
Definition at line 269 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_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 245 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_write(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_destroy(), dahdi_translate(), g723_read(), g723_write(), load_module(), phone_request(), phone_setup(), phone_write(), register_translator(), and start_rtp().
| #define AST_FORMAT_G726 (1 << 11) |
ADPCM (G.726, 32kbps, RFC3551 codeword packing)
Definition at line 267 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_set_rtpmap_type(), g726_read(), g726_write(), and g726tolin_sample().
| #define AST_FORMAT_G726_AAL2 (1 << 4) |
ADPCM (G.726, 32kbps, AAL2 codeword packing)
Definition at line 253 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_lookup_mime_subtype(), ast_rtp_set_rtpmap_type(), codec_ast2skinny(), codec_skinny2ast(), and setup_rtp_connection().
| #define AST_FORMAT_G729A (1 << 8) |
G.729A audio
Definition at line 261 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_destroy(), dahdi_translate(), g729_read(), g729_write(), load_module(), phone_request(), phone_setup(), phone_write(), and start_rtp().
| #define AST_FORMAT_GSM (1 << 1) |
GSM compression
Definition at line 247 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), convertcap(), gsm_read(), gsm_write(), gsmtolin_sample(), wav_read(), and wav_write().
| #define AST_FORMAT_H261 (1 << 18) |
H.261 Video
Definition at line 281 of file frame.h.
Referenced by codec_ast2skinny(), codec_skinny2ast(), and h261_encap().
| #define AST_FORMAT_H263 (1 << 19) |
H.263 Video
Definition at line 283 of file frame.h.
Referenced by codec_ast2skinny(), codec_skinny2ast(), h263_encap(), h263_read(), and h263_write().
| #define AST_FORMAT_H263_PLUS (1 << 20) |
| #define AST_FORMAT_H264 (1 << 21) |
H.264 Video
Definition at line 287 of file frame.h.
Referenced by h264_encap(), h264_read(), and h264_write().
| #define AST_FORMAT_ILBC (1 << 10) |
iLBC Free Compression
Definition at line 265 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_codec_interp_len(), convertcap(), ilbc_read(), ilbc_write(), and ilbctolin_sample().
| #define AST_FORMAT_JPEG (1 << 16) |
JPEG Images
Definition at line 277 of file frame.h.
Referenced by jpeg_read_image(), and jpeg_write_image().
| #define AST_FORMAT_LPC10 (1 << 7) |
LPC10, 180 samples/frame
Definition at line 259 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_samples(), and lpc10tolin_sample().
| #define AST_FORMAT_MP4_VIDEO (1 << 22) |
| #define AST_FORMAT_PNG (1 << 17) |
| #define AST_FORMAT_SLINEAR (1 << 6) |
Raw 16-bit Signed Linear (8000 Hz) PCM
Definition at line 257 of file frame.h.
Referenced by __ast_play_and_record(), __ast_register_translator(), action_originate(), agent_new(), alsa_new(), alsa_read(), alsa_request(), ast_audiohook_read_frame(), ast_best_codec(), ast_channel_make_compatible_helper(), ast_channel_start_silence_generator(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_call_progress(), ast_dsp_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(), moh_class_malloc(), mp3_exec(), nbs_request(), nbs_xwrite(), NBScat_exec(), ogg_vorbis_read(), ogg_vorbis_write(), oh323_rtp_read(), orig_app(), orig_exten(), oss_new(), oss_read(), oss_request(), parkandannounce_exec(), phone_new(), phone_read(), phone_request(), phone_setup(), phone_write(), playtones_alloc(), 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 273 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_slinfactory_feed(), console_new(), lin16tog722_sample(), slin16_to_slin8_sample(), slinear_read(), slinear_write(), and stream_monitor().
| #define AST_FORMAT_SPEEX (1 << 9) |
SpeeX Free Compression
Definition at line 263 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_samples(), ast_rtp_write(), convertcap(), and speextolin_sample().
| #define AST_FORMAT_T140 (1 << 27) |
T.140 Text format - ITU T.140, RFC 4103
Definition at line 294 of file frame.h.
Referenced by add_tcodec_to_sdp(), ast_rtp_read(), and ast_write().
| #define AST_FORMAT_T140RED (1 << 26) |
T.140 RED Text format RFC 4103
Definition at line 292 of file frame.h.
Referenced by add_tcodec_to_sdp(), ast_rtp_read(), process_sdp(), and rtp_red_init().
| #define AST_FORMAT_TEXT_MASK (((1 << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK)) |
Definition at line 297 of file frame.h.
Referenced by add_sdp(), ast_request(), check_peer_ok(), sip_new(), and sip_rtp_read().
| #define AST_FORMAT_ULAW (1 << 2) |
Raw mu-law data (G.711)
Definition at line 249 of file frame.h.
Referenced by __adsi_transmit_messages(), _ast_adsi_transmit_message_full(), adsi_careful_send(), alarmreceiver_exec(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), calc_energy(), codec_ast2skinny(), codec_skinny2ast(), conf_run(), convertcap(), dahdi_new(), dahdi_read(), dahdi_translate(), dahdi_write(), find_transcoders(), is_encoder(), load_module(), milliwatt_generate(), oh323_rtp_read(), old_milliwatt_exec(), phone_request(), phone_setup(), phone_write(), pri_dchannel(), send_tone_burst(), start_rtp(), ulawtoalaw_sample(), and ulawtolin_sample().
| #define AST_FORMAT_VIDEO_MASK (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) |
Definition at line 290 of file frame.h.
Referenced by add_sdp(), ast_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 497 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 496 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 186 of file frame.h.
Referenced by fax_generator_generate(), g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), ilbc_read(), ogg_vorbis_read(), pcm_read(), slinear_read(), t38_tx_packet_handler(), vox_read(), and wav_read().
| #define ast_frfree | ( | fr | ) | ast_frame_free(fr, 1) |
Definition at line 464 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(), dictate_exec(), disa_exec(), do_idle_thread(), do_waiting(), echo_exec(), eivr_comm(), find_cache(), gen_generate(), handle_cli_file_convert(), handle_invite_replaces(), 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(), 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 207 of file frame.h.
Referenced by __get_from_jb(), alsa_read(), ast_frdup(), ast_frisolate(), ast_prod(), ast_rtcp_read(), ast_rtp_read(), ast_smoother_read(), ast_trans_frameout(), ast_udptl_read(), conf_run(), dahdi_decoder_frameout(), dahdi_encoder_frameout(), dahdi_read(), fax_generator_generate(), g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), iax_frame_wrap(), ilbc_read(), jb_get_and_deliver(), linear_generator(), milliwatt_generate(), moh_generate(), mohalloc(), mp3_exec(), NBScat_exec(), newpvt(), ogg_vorbis_read(), oss_read(), pcm_read(), phone_read(), 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 233 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 235 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 225 of file frame.h.
Referenced by ast_channel_sendurl(), ast_frame_dump(), and sip_sendhtml().
| #define AST_MALLOCD_DATA (1 << 1) |
Need the data be free'd?
Definition at line 213 of file frame.h.
Referenced by __frame_free(), ast_frisolate(), and create_video_frame().
| #define AST_MALLOCD_HDR (1 << 0) |
Need the header be free'd?
Definition at line 211 of file frame.h.
Referenced by __frame_free(), ast_frame_header_new(), ast_frdup(), ast_frisolate(), and create_video_frame().
| #define AST_MALLOCD_SRC (1 << 2) |
Need the source be free'd? (haha!)
Definition at line 215 of file frame.h.
Referenced by __frame_free(), and ast_frisolate().
| #define AST_MIN_OFFSET 32 |
Definition at line 208 of file frame.h.
Referenced by __ast_smoother_feed().
| #define AST_MODEM_T38 1 |
T.38 Fax-over-IP
Definition at line 219 of file frame.h.
Referenced by ast_frame_dump(), 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 378 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 400 of file frame.h.
Referenced by dahdi_setoption().
| #define AST_OPTION_FLAG_REQUEST 0 |
Definition at line 360 of file frame.h.
Referenced by ast_bridge_call(), and iax2_setoption().
| #define AST_OPTION_OPRMODE 7 |
Definition at line 397 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 375 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 394 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 406 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 372 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 369 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 386 of file frame.h.
Referenced by common_exec(), dahdi_setoption(), func_channel_write(), iax2_setoption(), reset_volumes(), and set_listen_volume().
Definition at line 567 of file frame.h.
Referenced by ast_rtp_write().
Definition at line 572 of file frame.h.
Referenced by ast_rtp_write().
| #define AST_SMOOTHER_FLAG_BE (1 << 1) |
Definition at line 357 of file frame.h.
Referenced by ast_rtp_write().
| #define AST_SMOOTHER_FLAG_G729 (1 << 0) |
Definition at line 356 of file frame.h.
Referenced by __ast_smoother_feed(), ast_smoother_read(), and smoother_frame_feed().
| anonymous enum |
| AST_FRFLAG_HAS_TIMING_INFO |
This frame contains valid timing information |
| AST_FRFLAG_FROM_TRANSLATOR |
This frame came from a translator and is still the original frame. The translator can not be free'd if the frame inside of it still has this flag set. |
| AST_FRFLAG_FROM_DSP |
This frame came from a dsp and is still the original frame. The dsp cannot be free'd if the frame inside of it still has this flag set. |
| AST_FRFLAG_FROM_FILESTREAM |
This frame came from a filestream and is still the original frame. The filestream cannot be free'd if the frame inside of it still has this flag set. |
Definition at line 126 of file frame.h.
00126 { 00127 /*! This frame contains valid timing information */ 00128 AST_FRFLAG_HAS_TIMING_INFO = (1 << 0), 00129 /*! This frame came from a translator and is still the original frame. 00130 * The translator can not be free'd if the frame inside of it still has 00131 * this flag set. */ 00132 AST_FRFLAG_FROM_TRANSLATOR = (1 << 1), 00133 /*! This frame came from a dsp and is still the original frame. 00134 * The dsp cannot be free'd if the frame inside of it still has 00135 * this flag set. */ 00136 AST_FRFLAG_FROM_DSP = (1 << 2), 00137 /*! This frame came from a filestream and is still the original frame. 00138 * The filestream cannot be free'd if the frame inside of it still has 00139 * this flag set. */ 00140 AST_FRFLAG_FROM_FILESTREAM = (1 << 3), 00141 };
| 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 299 of file frame.h.
00299 { 00300 AST_CONTROL_HANGUP = 1, /*!< Other end has hungup */ 00301 AST_CONTROL_RING = 2, /*!< Local ring */ 00302 AST_CONTROL_RINGING = 3, /*!< Remote end is ringing */ 00303 AST_CONTROL_ANSWER = 4, /*!< Remote end has answered */ 00304 AST_CONTROL_BUSY = 5, /*!< Remote end is busy */ 00305 AST_CONTROL_TAKEOFFHOOK = 6, /*!< Make it go off hook */ 00306 AST_CONTROL_OFFHOOK = 7, /*!< Line is off hook */ 00307 AST_CONTROL_CONGESTION = 8, /*!< Congestion (circuits busy) */ 00308 AST_CONTROL_FLASH = 9, /*!< Flash hook */ 00309 AST_CONTROL_WINK = 10, /*!< Wink */ 00310 AST_CONTROL_OPTION = 11, /*!< Set a low-level option */ 00311 AST_CONTROL_RADIO_KEY = 12, /*!< Key Radio */ 00312 AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */ 00313 AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */ 00314 AST_CONTROL_PROCEEDING = 15, /*!< Indicate CALL PROCEEDING */ 00315 AST_CONTROL_HOLD = 16, /*!< Indicate call is placed on hold */ 00316 AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold */ 00317 AST_CONTROL_VIDUPDATE = 18, /*!< Indicate video frame update */ 00318 _XXX_AST_CONTROL_T38 = 19, /*!< T38 state change request/notification \deprecated This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead. */ 00319 AST_CONTROL_SRCUPDATE = 20, /*!< Indicate source of media has changed */ 00320 AST_CONTROL_T38_PARAMETERS = 24, /*!< T38 state change request/notification with parameters */ 00321 };
| enum ast_control_t38 |
Definition at line 323 of file frame.h.
00323 { 00324 AST_T38_REQUEST_NEGOTIATE = 1, /*!< Request T38 on a channel (voice to fax) */ 00325 AST_T38_REQUEST_TERMINATE, /*!< Terminate T38 on a channel (fax to voice) */ 00326 AST_T38_NEGOTIATED, /*!< T38 negotiated (fax mode) */ 00327 AST_T38_TERMINATED, /*!< T38 terminated (back to voice) */ 00328 AST_T38_REFUSED /*!< T38 refused for some reason (usually rejected by remote end) */ 00329 };
| 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 331 of file frame.h.
00331 { 00332 AST_T38_RATE_2400 = 0, 00333 AST_T38_RATE_4800, 00334 AST_T38_RATE_7200, 00335 AST_T38_RATE_9600, 00336 AST_T38_RATE_12000, 00337 AST_T38_RATE_14400, 00338 };
Definition at line 340 of file frame.h.
00340 { 00341 AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF = 0, 00342 AST_T38_RATE_MANAGEMENT_LOCAL_TCF, 00343 };
| 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 648 of file frame.c.
References ARRAY_LEN, and ast_format_list::desc.
Referenced by moh_alloc(), show_codec_n(), and show_codecs().
00649 { 00650 int x; 00651 char *ret = "unknown"; 00652 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00653 if (AST_FORMAT_LIST[x].bits == codec) { 00654 ret = AST_FORMAT_LIST[x].desc; 00655 break; 00656 } 00657 } 00658 return ret; 00659 }
| 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 1205 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().
01206 { 01207 int x, ret = 0, slot; 01208 01209 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01210 slot = pref->order[x]; 01211 01212 if (!slot) 01213 break; 01214 if (formats & AST_FORMAT_LIST[slot-1].bits) { 01215 ret = AST_FORMAT_LIST[slot-1].bits; 01216 break; 01217 } 01218 } 01219 if (ret & AST_FORMAT_AUDIO_MASK) 01220 return ret; 01221 01222 ast_debug(4, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec"); 01223 01224 return find_best ? ast_best_codec(formats) : 0; 01225 }
| 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 1469 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().
01470 { 01471 int len = 0; 01472 01473 /* XXX Still need speex, g723, and lpc10 XXX */ 01474 switch(format) { 01475 case AST_FORMAT_G723_1: 01476 len = (samples / 240) * 20; 01477 break; 01478 case AST_FORMAT_ILBC: 01479 len = (samples / 240) * 50; 01480 break; 01481 case AST_FORMAT_GSM: 01482 len = (samples / 160) * 33; 01483 break; 01484 case AST_FORMAT_G729A: 01485 len = samples / 8; 01486 break; 01487 case AST_FORMAT_SLINEAR: 01488 case AST_FORMAT_SLINEAR16: 01489 len = samples * 2; 01490 break; 01491 case AST_FORMAT_ULAW: 01492 case AST_FORMAT_ALAW: 01493 len = samples; 01494 break; 01495 case AST_FORMAT_G722: 01496 case AST_FORMAT_ADPCM: 01497 case AST_FORMAT_G726: 01498 case AST_FORMAT_G726_AAL2: 01499 len = samples / 2; 01500 break; 01501 default: 01502 ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format)); 01503 } 01504 01505 return len; 01506 }
| int ast_codec_get_samples | ( | struct ast_frame * | f | ) |
Returns the number of samples contained in the frame.
Definition at line 1425 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().
01426 { 01427 int samples=0; 01428 switch(f->subclass) { 01429 case AST_FORMAT_SPEEX: 01430 samples = speex_samples(f->data.ptr, f->datalen); 01431 break; 01432 case AST_FORMAT_G723_1: 01433 samples = g723_samples(f->data.ptr, f->datalen); 01434 break; 01435 case AST_FORMAT_ILBC: 01436 samples = 240 * (f->datalen / 50); 01437 break; 01438 case AST_FORMAT_GSM: 01439 samples = 160 * (f->datalen / 33); 01440 break; 01441 case AST_FORMAT_G729A: 01442 samples = f->datalen * 8; 01443 break; 01444 case AST_FORMAT_SLINEAR: 01445 case AST_FORMAT_SLINEAR16: 01446 samples = f->datalen / 2; 01447 break; 01448 case AST_FORMAT_LPC10: 01449 /* assumes that the RTP packet contains one LPC10 frame */ 01450 samples = 22 * 8; 01451 samples += (((char *)(f->data.ptr))[7] & 0x1) * 8; 01452 break; 01453 case AST_FORMAT_ULAW: 01454 case AST_FORMAT_ALAW: 01455 samples = f->datalen; 01456 break; 01457 case AST_FORMAT_G722: 01458 case AST_FORMAT_ADPCM: 01459 case AST_FORMAT_G726: 01460 case AST_FORMAT_G726_AAL2: 01461 samples = f->datalen * 2; 01462 break; 01463 default: 01464 ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass)); 01465 } 01466 return samples; 01467 }
| static int ast_codec_interp_len | ( | int | format | ) | [inline, static] |
Gets duration in ms of interpolation frame for a format.
Definition at line 655 of file frame.h.
References AST_FORMAT_ILBC.
Referenced by __get_from_jb(), and jb_get_and_deliver().
00656 { 00657 return (format == AST_FORMAT_ILBC) ? 30 : 20; 00658 }
| 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 1065 of file frame.c.
References ARRAY_LEN, ast_codec_pref_remove(), and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow().
01066 { 01067 int x, newindex = 0; 01068 01069 ast_codec_pref_remove(pref, format); 01070 01071 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01072 if (AST_FORMAT_LIST[x].bits == format) { 01073 newindex = x + 1; 01074 break; 01075 } 01076 } 01077 01078 if (newindex) { 01079 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01080 if (!pref->order[x]) { 01081 pref->order[x] = newindex; 01082 break; 01083 } 01084 } 01085 } 01086 01087 return x; 01088 }
| 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 968 of file frame.c.
References ast_codec_pref::order.
Referenced by check_access(), create_addr(), dump_prefs(), and socket_process().
00969 { 00970 int x, differential = (int) 'A', mem; 00971 char *from, *to; 00972 00973 if (right) { 00974 from = pref->order; 00975 to = buf; 00976 mem = size; 00977 } else { 00978 to = pref->order; 00979 from = buf; 00980 mem = 32; 00981 } 00982 00983 memset(to, 0, mem); 00984 for (x = 0; x < 32 ; x++) { 00985 if (!from[x]) 00986 break; 00987 to[x] = right ? (from[x] + differential) : (from[x] - differential); 00988 } 00989 }
| struct ast_format_list ast_codec_pref_getsize | ( | struct ast_codec_pref * | pref, | |
| int | format | |||
| ) | [read] |
Get packet size for codec.
Definition at line 1166 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().
01167 { 01168 int x, idx = -1, framems = 0; 01169 struct ast_format_list fmt = { 0, }; 01170 01171 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01172 if (AST_FORMAT_LIST[x].bits == format) { 01173 fmt = AST_FORMAT_LIST[x]; 01174 idx = x; 01175 break; 01176 } 01177 } 01178 01179 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01180 if (pref->order[x] == (idx + 1)) { 01181 framems = pref->framing[x]; 01182 break; 01183 } 01184 } 01185 01186 /* size validation */ 01187 if (!framems) 01188 framems = AST_FORMAT_LIST[idx].def_ms; 01189 01190 if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */ 01191 framems -= framems % AST_FORMAT_LIST[idx].inc_ms; 01192 01193 if (framems < AST_FORMAT_LIST[idx].min_ms) 01194 framems = AST_FORMAT_LIST[idx].min_ms; 01195 01196 if (framems > AST_FORMAT_LIST[idx].max_ms) 01197 framems = AST_FORMAT_LIST[idx].max_ms; 01198 01199 fmt.cur_ms = framems; 01200 01201 return fmt; 01202 }
| 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 1026 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().
01027 { 01028 int slot = 0; 01029 01030 if ((idx >= 0) && (idx < sizeof(pref->order))) { 01031 slot = pref->order[idx]; 01032 } 01033 01034 return slot ? AST_FORMAT_LIST[slot - 1].bits : 0; 01035 }
| 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 1091 of file frame.c.
References ARRAY_LEN, ast_codec_pref::framing, and ast_codec_pref::order.
Referenced by create_addr().
01092 { 01093 int x, newindex = 0; 01094 01095 /* First step is to get the codecs "index number" */ 01096 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01097 if (AST_FORMAT_LIST[x].bits == format) { 01098 newindex = x + 1; 01099 break; 01100 } 01101 } 01102 /* Done if its unknown */ 01103 if (!newindex) 01104 return; 01105 01106 /* Now find any existing occurrence, or the end */ 01107 for (x = 0; x < 32; x++) { 01108 if (!pref->order[x] || pref->order[x] == newindex) 01109 break; 01110 } 01111 01112 if (only_if_existing && !pref->order[x]) 01113 return; 01114 01115 /* Move down to make space to insert - either all the way to the end, 01116 or as far as the existing location (which will be overwritten) */ 01117 for (; x > 0; x--) { 01118 pref->order[x] = pref->order[x - 1]; 01119 pref->framing[x] = pref->framing[x - 1]; 01120 } 01121 01122 /* And insert the new entry */ 01123 pref->order[0] = newindex; 01124 pref->framing[0] = 0; /* ? */ 01125 }
| void ast_codec_pref_remove | ( | struct ast_codec_pref * | pref, | |
| int | format | |||
| ) |
Remove audio a codec from a preference list.
Definition at line 1038 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().
01039 { 01040 struct ast_codec_pref oldorder; 01041 int x, y = 0; 01042 int slot; 01043 int size; 01044 01045 if (!pref->order[0]) 01046 return; 01047 01048 memcpy(&oldorder, pref, sizeof(oldorder)); 01049 memset(pref, 0, sizeof(*pref)); 01050 01051 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01052 slot = oldorder.order[x]; 01053 size = oldorder.framing[x]; 01054 if (! slot) 01055 break; 01056 if (AST_FORMAT_LIST[slot-1].bits != format) { 01057 pref->order[y] = slot; 01058 pref->framing[y++] = size; 01059 } 01060 } 01061 01062 }
| int ast_codec_pref_setsize | ( | struct ast_codec_pref * | pref, | |
| int | format, | |||
| int | framems | |||
| ) |
Set packet size for codec.
Definition at line 1128 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().
01129 { 01130 int x, idx = -1; 01131 01132 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01133 if (AST_FORMAT_LIST[x].bits == format) { 01134 idx = x; 01135 break; 01136 } 01137 } 01138 01139 if (idx < 0) 01140 return -1; 01141 01142 /* size validation */ 01143 if (!framems) 01144 framems = AST_FORMAT_LIST[idx].def_ms; 01145 01146 if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */ 01147 framems -= framems % AST_FORMAT_LIST[idx].inc_ms; 01148 01149 if (framems < AST_FORMAT_LIST[idx].min_ms) 01150 framems = AST_FORMAT_LIST[idx].min_ms; 01151 01152 if (framems > AST_FORMAT_LIST[idx].max_ms) 01153 framems = AST_FORMAT_LIST[idx].max_ms; 01154 01155 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01156 if (pref->order[x] == (idx + 1)) { 01157 pref->framing[x] = framems; 01158 break; 01159 } 01160 } 01161 01162 return x; 01163 }
| 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 991 of file frame.c.
References ast_codec_pref_index(), and ast_getformatname().
Referenced by dump_prefs(), and socket_process().
00992 { 00993 int x, codec; 00994 size_t total_len, slen; 00995 char *formatname; 00996 00997 memset(buf,0,size); 00998 total_len = size; 00999 buf[0] = '('; 01000 total_len--; 01001 for(x = 0; x < 32 ; x++) { 01002 if (total_len <= 0) 01003 break; 01004 if (!(codec = ast_codec_pref_index(pref,x))) 01005 break; 01006 if ((formatname = ast_getformatname(codec))) { 01007 slen = strlen(formatname); 01008 if (slen > total_len) 01009 break; 01010 strncat(buf, formatname, total_len - 1); /* safe */ 01011 total_len -= slen; 01012 } 01013 if (total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) { 01014 strncat(buf, "|", total_len - 1); /* safe */ 01015 total_len--; 01016 } 01017 } 01018 if (total_len) { 01019 strncat(buf, ")", total_len - 1); /* safe */ 01020 total_len--; 01021 } 01022 01023 return size - total_len; 01024 }
| static force_inline int ast_format_rate | ( | int | format | ) | [static] |
Get the sample rate for a given format.
Definition at line 682 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().
00683 { 00684 if (format == AST_FORMAT_G722 || format == AST_FORMAT_SLINEAR16) 00685 return 16000; 00686 00687 return 8000; 00688 }
| 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 1508 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().
01509 { 01510 int count; 01511 short *fdata = f->data.ptr; 01512 short adjust_value = abs(adjustment); 01513 01514 if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR)) 01515 return -1; 01516 01517 if (!adjustment) 01518 return 0; 01519 01520 for (count = 0; count < f->samples; count++) { 01521 if (adjustment > 0) { 01522 ast_slinear_saturated_multiply(&fdata[count], &adjust_value); 01523 } else if (adjustment < 0) { 01524 ast_slinear_saturated_divide(&fdata[count], &adjust_value); 01525 } 01526 } 01527 01528 return 0; 01529 }
| void ast_frame_dump | ( | const char * | name, | |
| struct ast_frame * | f, | |||
| char * | prefix | |||
| ) |
Dump a frame for debugging purposes
Definition at line 750 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().
00751 { 00752 const char noname[] = "unknown"; 00753 char ftype[40] = "Unknown Frametype"; 00754 char cft[80]; 00755 char subclass[40] = "Unknown Subclass"; 00756 char csub[80]; 00757 char moreinfo[40] = ""; 00758 char cn[60]; 00759 char cp[40]; 00760 char cmn[40]; 00761 const char *message = "Unknown"; 00762 00763 if (!name) 00764 name = noname; 00765 00766 00767 if (!f) { 00768 ast_verbose("%s [ %s (NULL) ] [%s]\n", 00769 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00770 term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00771 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00772 return; 00773 } 00774 /* XXX We should probably print one each of voice and video when the format changes XXX */ 00775 if (f->frametype == AST_FRAME_VOICE) 00776 return; 00777 if (f->frametype == AST_FRAME_VIDEO) 00778 return; 00779 switch(f->frametype) { 00780 case AST_FRAME_DTMF_BEGIN: 00781 strcpy(ftype, "DTMF Begin"); 00782 subclass[0] = f->subclass; 00783 subclass[1] = '\0'; 00784 break; 00785 case AST_FRAME_DTMF_END: 00786 strcpy(ftype, "DTMF End"); 00787 subclass[0] = f->subclass; 00788 subclass[1] = '\0'; 00789 break; 00790 case AST_FRAME_CONTROL: 00791 strcpy(ftype, "Control"); 00792 switch(f->subclass) { 00793 case AST_CONTROL_HANGUP: 00794 strcpy(subclass, "Hangup"); 00795 break; 00796 case AST_CONTROL_RING: 00797 strcpy(subclass, "Ring"); 00798 break; 00799 case AST_CONTROL_RINGING: 00800 strcpy(subclass, "Ringing"); 00801 break; 00802 case AST_CONTROL_ANSWER: 00803 strcpy(subclass, "Answer"); 00804 break; 00805 case AST_CONTROL_BUSY: 00806 strcpy(subclass, "Busy"); 00807 break; 00808 case AST_CONTROL_TAKEOFFHOOK: 00809 strcpy(subclass, "Take Off Hook"); 00810 break; 00811 case AST_CONTROL_OFFHOOK: 00812 strcpy(subclass, "Line Off Hook"); 00813 break; 00814 case AST_CONTROL_CONGESTION: 00815 strcpy(subclass, "Congestion"); 00816 break; 00817 case AST_CONTROL_FLASH: 00818 strcpy(subclass, "Flash"); 00819 break; 00820 case AST_CONTROL_WINK: 00821 strcpy(subclass, "Wink"); 00822 break; 00823 case AST_CONTROL_OPTION: 00824 strcpy(subclass, "Option"); 00825 break; 00826 case AST_CONTROL_RADIO_KEY: 00827 strcpy(subclass, "Key Radio"); 00828 break; 00829 case AST_CONTROL_RADIO_UNKEY: 00830 strcpy(subclass, "Unkey Radio"); 00831 break; 00832 case AST_CONTROL_HOLD: 00833 strcpy(subclass, "Hold"); 00834 break; 00835 case AST_CONTROL_UNHOLD: 00836 strcpy(subclass, "Unhold"); 00837 break; 00838 case AST_CONTROL_T38_PARAMETERS: 00839 if (f->datalen != sizeof(struct ast_control_t38_parameters *)) { 00840 message = "Invalid"; 00841 } else { 00842 struct ast_control_t38_parameters *parameters = f->data.ptr; 00843 enum ast_control_t38 state = parameters->request_response; 00844 if (state == AST_T38_REQUEST_NEGOTIATE) 00845 message = "Negotiation Requested"; 00846 else if (state == AST_T38_REQUEST_TERMINATE) 00847 message = "Negotiation Request Terminated"; 00848 else if (state == AST_T38_NEGOTIATED) 00849 message = "Negotiated"; 00850 else if (state == AST_T38_TERMINATED) 00851 message = "Terminated"; 00852 else if (state == AST_T38_REFUSED) 00853 message = "Refused"; 00854 } 00855 snprintf(subclass, sizeof(subclass), "T38_Parameters/%s", message); 00856 break; 00857 case -1: 00858 strcpy(subclass, "Stop generators"); 00859 break; 00860 default: 00861 snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass); 00862 } 00863 break; 00864 case AST_FRAME_NULL: 00865 strcpy(ftype, "Null Frame"); 00866 strcpy(subclass, "N/A"); 00867 break; 00868 case AST_FRAME_IAX: 00869 /* Should never happen */ 00870 strcpy(ftype, "IAX Specific"); 00871 snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass); 00872 break; 00873 case AST_FRAME_TEXT: 00874 strcpy(ftype, "Text"); 00875 strcpy(subclass, "N/A"); 00876 ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); 00877 break; 00878 case AST_FRAME_IMAGE: 00879 strcpy(ftype, "Image"); 00880 snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass)); 00881 break; 00882 case AST_FRAME_HTML: 00883 strcpy(ftype, "HTML"); 00884 switch(f->subclass) { 00885 case AST_HTML_URL: 00886 strcpy(subclass, "URL"); 00887 ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); 00888 break; 00889 case AST_HTML_DATA: 00890 strcpy(subclass, "Data"); 00891 break; 00892 case AST_HTML_BEGIN: 00893 strcpy(subclass, "Begin"); 00894 break; 00895 case AST_HTML_END: 00896 strcpy(subclass, "End"); 00897 break; 00898 case AST_HTML_LDCOMPLETE: 00899 strcpy(subclass, "Load Complete"); 00900 break; 00901 case AST_HTML_NOSUPPORT: 00902 strcpy(subclass, "No Support"); 00903 break; 00904 case AST_HTML_LINKURL: 00905 strcpy(subclass, "Link URL"); 00906 ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); 00907 break; 00908 case AST_HTML_UNLINK: 00909 strcpy(subclass, "Unlink"); 00910 break; 00911 case AST_HTML_LINKREJECT: 00912 strcpy(subclass, "Link Reject"); 00913 break; 00914 default: 00915 snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass); 00916 break; 00917 } 00918 break; 00919 case AST_FRAME_MODEM: 00920 strcpy(ftype, "Modem"); 00921 switch (f->subclass) { 00922 case AST_MODEM_T38: 00923 strcpy(subclass, "T.38"); 00924 break; 00925 case AST_MODEM_V150: 00926 strcpy(subclass, "V.150"); 00927 break; 00928 default: 00929 snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass); 00930 break; 00931 } 00932 break; 00933 default: 00934 snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype); 00935 } 00936 if (!ast_strlen_zero(moreinfo)) 00937 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n", 00938 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00939 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00940 f->frametype, 00941 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00942 f->subclass, 00943 term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)), 00944 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00945 else 00946 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n", 00947 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00948 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00949 f->frametype, 00950 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00951 f->subclass, 00952 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00953 }
| 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 373 of file frame.c.
References __frame_free(), and AST_LIST_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 }
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 1531 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.
01532 { 01533 int count; 01534 short *data1, *data2; 01535 01536 if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR)) 01537 return -1; 01538 01539 if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR)) 01540 return -1; 01541 01542 if (f1->samples != f2->samples) 01543 return -1; 01544 01545 for (count = 0, data1 = f1->data.ptr, data2 = f2->data.ptr; 01546 count < f1->samples; 01547 count++, data1++, data2++) 01548 ast_slinear_saturated_add(data1, data2); 01549 01550 return 0; 01551 }
Copies a frame.
| fr | frame to copy Duplicates a frame -- should only rarely be used, typically frisolate is good enough |
Definition at line 470 of file frame.c.
References ast_calloc_cache, ast_copy_flags, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, AST_LIST_REMOVE_CURRENT, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, AST_MALLOCD_HDR, ast_threadstorage_get(), buf, ast_frame::data, ast_frame::datalen, ast_frame::delivery, 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().
00471 { 00472 struct ast_frame *out = NULL; 00473 int len, srclen = 0; 00474 void *buf = NULL; 00475 00476 #if !defined(LOW_MEMORY) 00477 struct ast_frame_cache *frames; 00478 #endif 00479 00480 /* Start with standard stuff */ 00481 len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00482 /* If we have a source, add space for it */ 00483 /* 00484 * XXX Watch out here - if we receive a src which is not terminated 00485 * properly, we can be easily attacked. Should limit the size we deal with. 00486 */ 00487 if (f->src) 00488 srclen = strlen(f->src); 00489 if (srclen > 0) 00490 len += srclen + 1; 00491 00492 #if !defined(LOW_MEMORY) 00493 if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) { 00494 AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) { 00495 if (out->mallocd_hdr_len >= len) { 00496 size_t mallocd_len = out->mallocd_hdr_len; 00497 00498 AST_LIST_REMOVE_CURRENT(frame_list); 00499 memset(out, 0, sizeof(*out)); 00500 out->mallocd_hdr_len = mallocd_len; 00501 buf = out; 00502 frames->size--; 00503 break; 00504 } 00505 } 00506 AST_LIST_TRAVERSE_SAFE_END; 00507 } 00508 #endif 00509 00510 if (!buf) { 00511 if (!(buf = ast_calloc_cache(1, len))) 00512 return NULL; 00513 out = buf; 00514 out->mallocd_hdr_len = len; 00515 } 00516 00517 out->frametype = f->frametype; 00518 out->subclass = f->subclass; 00519 out->datalen = f->datalen; 00520 out->samples = f->samples; 00521 out->delivery = f->delivery; 00522 /* Set us as having malloc'd header only, so it will eventually 00523 get freed. */ 00524 out->mallocd = AST_MALLOCD_HDR; 00525 out->offset = AST_FRIENDLY_OFFSET; 00526 if (out->datalen) { 00527 out->data.ptr = buf + sizeof(*out) + AST_FRIENDLY_OFFSET; 00528 memcpy(out->data.ptr, f->data.ptr, out->datalen); 00529 } else { 00530 out->data.uint32 = f->data.uint32; 00531 } 00532 if (srclen > 0) { 00533 /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */ 00534 char *src; 00535 out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00536 src = (char *) out->src; 00537 /* Must have space since we allocated for it */ 00538 strcpy(src, f->src); 00539 } 00540 ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO); 00541 out->ts = f->ts; 00542 out->len = f->len; 00543 out->seqno = f->seqno; 00544 return out; 00545 }
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 389 of file frame.c.
References ast_clear_flag, ast_copy_flags, ast_frame_header_new(), ast_frdup(), ast_free, AST_FRFLAG_FROM_DSP, AST_FRFLAG_FROM_FILESTREAM, AST_FRFLAG_FROM_TRANSLATOR, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, ast_malloc, AST_MALLOCD_DATA, AST_MALLOCD_HDR, AST_MALLOCD_SRC, ast_strdup, ast_test_flag, ast_frame::data, ast_frame::datalen, ast_frame::frametype, ast_frame::len, ast_frame::mallocd, ast_frame::offset, ast_frame::ptr, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, and ast_frame::ts.
Referenced by __ast_answer(), ast_slinfactory_feed(), autoservice_run(), and jpeg_read_image().
00390 { 00391 struct ast_frame *out; 00392 void *newdata; 00393 00394 /* if none of the existing frame is malloc'd, let ast_frdup() do it 00395 since it is more efficient 00396 */ 00397 if (fr->mallocd == 0) { 00398 return ast_frdup(fr); 00399 } 00400 00401 /* if everything is already malloc'd, we are done */ 00402 if ((fr->mallocd & (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) == 00403 (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) { 00404 return fr; 00405 } 00406 00407 if (!(fr->mallocd & AST_MALLOCD_HDR)) { 00408 /* Allocate a new header if needed */ 00409 if (!(out = ast_frame_header_new())) { 00410 return NULL; 00411 } 00412 out->frametype = fr->frametype; 00413 out->subclass = fr->subclass; 00414 out->datalen = fr->datalen; 00415 out->samples = fr->samples; 00416 out->offset = fr->offset; 00417 /* Copy the timing data */ 00418 ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO); 00419 if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) { 00420 out->ts = fr->ts; 00421 out->len = fr->len; 00422 out->seqno = fr->seqno; 00423 } 00424 } else { 00425 ast_clear_flag(fr, AST_FRFLAG_FROM_TRANSLATOR); 00426 ast_clear_flag(fr, AST_FRFLAG_FROM_DSP); 00427 ast_clear_flag(fr, AST_FRFLAG_FROM_FILESTREAM); 00428 out = fr; 00429 } 00430 00431 if (!(fr->mallocd & AST_MALLOCD_SRC) && fr->src) { 00432 if (!(out->src = ast_strdup(fr->src))) { 00433 if (out != fr) { 00434 ast_free(out); 00435 } 00436 return NULL; 00437 } 00438 } else { 00439 out->src = fr->src; 00440 fr->src = NULL; 00441 fr->mallocd &= ~AST_MALLOCD_SRC; 00442 } 00443 00444 if (!(fr->mallocd & AST_MALLOCD_DATA)) { 00445 if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) { 00446 if (out->src != fr->src) { 00447 ast_free((void *) out->src); 00448 } 00449 if (out != fr) { 00450 ast_free(out); 00451 } 00452 return NULL; 00453 } 00454 newdata += AST_FRIENDLY_OFFSET; 00455 out->offset = AST_FRIENDLY_OFFSET; 00456 out->datalen = fr->datalen; 00457 memcpy(newdata, fr->data.ptr, fr->datalen); 00458 out->data.ptr = newdata; 00459 } else { 00460 out->data = fr->data; 00461 memset(&fr->data, 0, sizeof(fr->data)); 00462 fr->mallocd &= ~AST_MALLOCD_DATA; 00463 } 00464 00465 out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA; 00466 00467 return out; 00468 }
| struct ast_format_list* ast_get_format_list | ( | size_t * | size | ) | [read] |
Definition at line 563 of file frame.c.
References ARRAY_LEN.
00564 { 00565 *size = ARRAY_LEN(AST_FORMAT_LIST); 00566 return AST_FORMAT_LIST; 00567 }
| struct ast_format_list* ast_get_format_list_index | ( | int | index | ) | [read] |
Definition at line 558 of file frame.c.
00559 { 00560 return &AST_FORMAT_LIST[idx]; 00561 }
| int ast_getformatbyname | ( | const char * | name | ) |
Gets a format from a name.
| name | string of format |
Definition at line 630 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().
00631 { 00632 int x, all, format = 0; 00633 00634 all = strcasecmp(name, "all") ? 0 : 1; 00635 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00636 if (all || 00637 !strcasecmp(AST_FORMAT_LIST[x].name,name) || 00638 !strcasecmp(AST_FORMAT_LIST[x].name, ast_expand_codec_alias(name))) { 00639 format |= AST_FORMAT_LIST[x].bits; 00640 if (!all) 00641 break; 00642 } 00643 } 00644 00645 return format; 00646 }
| char* ast_getformatname | ( | int | format | ) |
Get the name of a format.
| format | id of format |
Definition at line 569 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().
00570 { 00571 int x; 00572 char *ret = "unknown"; 00573 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00574 if (AST_FORMAT_LIST[x].bits == format) { 00575 ret = AST_FORMAT_LIST[x].name; 00576 break; 00577 } 00578 } 00579 return ret; 00580 }
| char* ast_getformatname_multiple | ( | char * | buf, | |
| size_t | size, | |||
| 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 582 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().
00583 { 00584 int x; 00585 unsigned len; 00586 char *start, *end = buf; 00587 00588 if (!size) 00589 return buf; 00590 snprintf(end, size, "0x%x (", format); 00591 len = strlen(end); 00592 end += len; 00593 size -= len; 00594 start = end; 00595 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00596 if (AST_FORMAT_LIST[x].bits & format) { 00597 snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name); 00598 len = strlen(end); 00599 end += len; 00600 size -= len; 00601 } 00602 } 00603 if (start == end) 00604 ast_copy_string(start, "nothing)", size); 00605 else if (size > 1) 00606 *(end -1) = ')'; 00607 return buf; 00608 }
| int ast_parse_allow_disallow | ( | struct ast_codec_pref * | pref, | |
| 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 1227 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().
01228 { 01229 int errors = 0; 01230 char *parse = NULL, *this = NULL, *psize = NULL; 01231 int format = 0, framems = 0; 01232 01233 parse = ast_strdupa(list); 01234 while ((this = strsep(&parse, ","))) { 01235 framems = 0; 01236 if ((psize = strrchr(this, ':'))) { 01237 *psize++ = '\0'; 01238 ast_debug(1, "Packetization for codec: %s is %s\n", this, psize); 01239 framems = atoi(psize); 01240 if (framems < 0) { 01241 framems = 0; 01242 errors++; 01243 ast_log(LOG_WARNING, "Bad packetization value for codec %s\n", this); 01244 } 01245 } 01246 if (!(format = ast_getformatbyname(this))) { 01247 ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this); 01248 errors++; 01249 continue; 01250 } 01251 01252 if (mask) { 01253 if (allowing) 01254 *mask |= format; 01255 else 01256 *mask &= ~format; 01257 } 01258 01259 /* Set up a preference list for audio. Do not include video in preferences 01260 since we can not transcode video and have to use whatever is offered 01261 */ 01262 if (pref && (format & AST_FORMAT_AUDIO_MASK)) { 01263 if (strcasecmp(this, "all")) { 01264 if (allowing) { 01265 ast_codec_pref_append(pref, format); 01266 ast_codec_pref_setsize(pref, format, framems); 01267 } 01268 else 01269 ast_codec_pref_remove(pref, format); 01270 } else if (!allowing) { 01271 memset(pref, 0, sizeof(*pref)); 01272 } 01273 } 01274 } 01275 return errors; 01276 }
| 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 547 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