frame.h File Reference
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 |
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) |
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_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 << 25) |
| #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, __f->data, __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 |
Detailed Description
Asterisk internal frame definitions.
- For an explanation of frames, see AST Multimedia and signalling frames
- Frames are send of Asterisk channels, see Asterisk Channels
Definition in file frame.h.
Define Documentation
| #define AST_FORMAT_ADPCM (1 << 5) |
ADPCM (IMA)
Definition at line 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(), 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_digitdetect(), 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(), dahdi_read(), dahdi_translate(), dahdi_write(), dictate_exec(), do_waiting(), eagi_exec(), extenspy_exec(), fax_generator_generate(), find_transcoders(), function_ilink(), 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 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 << 25) |
T.140 Text format - ITU T.140, RFC 4351
Definition at line 280 of file frame.h.
Referenced by ast_write().
| #define AST_FORMAT_TEXT_MASK (((1 << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK)) |
Definition at line 281 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(), 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_openvstream(), ast_request(), ast_rtp_read(), ast_translate_available_formats(), check_peer_ok(), check_user_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) |
| #define ast_frame_byteswap_le | ( | fr | ) | do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data, __f->data, __f->samples); } while(0) |
| #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_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(), process_ast_dsp(), receive_dtmf_digits(), record_exec(), rpt(), rpt_call(), rpt_exec(), send_waveform_to_channel(), sip_rtp_read(), speech_background(), ss_thread(), unistim_do_senddigit(), unistim_senddigit_end(), volume_callback(), wait_for_answer(), and wait_for_winner().
| #define AST_FRAME_SET_BUFFER | ( | fr, | |||
| _base, | |||||
| _ofs, | |||||
| _datalen | ) |
Value:
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 448 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_parking_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(), 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(), and ast_frisolate().
| #define AST_MIN_OFFSET 32 |
| #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 362 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 384 of file frame.h.
Referenced by dahdi_setoption().
| #define AST_OPTION_FLAG_REQUEST 0 |
| #define AST_OPTION_OPRMODE 7 |
| #define AST_OPTION_RELAXDTMF 3 |
Relax the parameters for DTMF reception (mainly for radio use)
Definition at line 359 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 378 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 390 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 356 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 353 of file frame.h.
Referenced by conf_run(), dahdi_hangup(), dahdi_setoption(), rpt(), 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 370 of file frame.h.
Referenced by common_exec(), dahdi_setoption(), func_channel_write(), iax2_setoption(), reset_volumes(), and set_listen_volume().
| #define AST_SMOOTHER_FLAG_BE (1 << 1) |
| #define AST_SMOOTHER_FLAG_G729 (1 << 0) |
Definition at line 340 of file frame.h.
Referenced by __ast_smoother_feed(), ast_smoother_read(), and smoother_frame_feed().
Enumeration Type Documentation
| 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 };
- Enumerator:
-
AST_CONTROL_HANGUP Other end has hungup AST_CONTROL_RING Local ring AST_CONTROL_RINGING Remote end is ringing AST_CONTROL_ANSWER Remote end has answered AST_CONTROL_BUSY Remote end is busy AST_CONTROL_TAKEOFFHOOK Make it go off hook AST_CONTROL_OFFHOOK Line is off hook AST_CONTROL_CONGESTION Congestion (circuits busy) AST_CONTROL_FLASH Flash hook AST_CONTROL_WINK Wink AST_CONTROL_OPTION Set a low-level option AST_CONTROL_RADIO_KEY Key Radio AST_CONTROL_RADIO_UNKEY Un-Key Radio AST_CONTROL_PROGRESS Indicate PROGRESS AST_CONTROL_PROCEEDING Indicate CALL PROCEEDING AST_CONTROL_HOLD Indicate call is placed on hold AST_CONTROL_UNHOLD Indicate call is left from hold AST_CONTROL_VIDUPDATE Indicate video frame update _XXX_AST_CONTROL_T38 T38 state change request/notification - Deprecated:
- This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead.
AST_CONTROL_SRCUPDATE Indicate source of media has changed AST_CONTROL_T38_PARAMETERS T38 state change request/notification with parameters
Definition at line 283 of file frame.h.
00283 { 00284 AST_CONTROL_HANGUP = 1, /*!< Other end has hungup */ 00285 AST_CONTROL_RING = 2, /*!< Local ring */ 00286 AST_CONTROL_RINGING = 3, /*!< Remote end is ringing */ 00287 AST_CONTROL_ANSWER = 4, /*!< Remote end has answered */ 00288 AST_CONTROL_BUSY = 5, /*!< Remote end is busy */ 00289 AST_CONTROL_TAKEOFFHOOK = 6, /*!< Make it go off hook */ 00290 AST_CONTROL_OFFHOOK = 7, /*!< Line is off hook */ 00291 AST_CONTROL_CONGESTION = 8, /*!< Congestion (circuits busy) */ 00292 AST_CONTROL_FLASH = 9, /*!< Flash hook */ 00293 AST_CONTROL_WINK = 10, /*!< Wink */ 00294 AST_CONTROL_OPTION = 11, /*!< Set a low-level option */ 00295 AST_CONTROL_RADIO_KEY = 12, /*!< Key Radio */ 00296 AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */ 00297 AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */ 00298 AST_CONTROL_PROCEEDING = 15, /*!< Indicate CALL PROCEEDING */ 00299 AST_CONTROL_HOLD = 16, /*!< Indicate call is placed on hold */ 00300 AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold */ 00301 AST_CONTROL_VIDUPDATE = 18, /*!< Indicate video frame update */ 00302 _XXX_AST_CONTROL_T38 = 19, /*!< T38 state change request/notification \deprecated This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead. */ 00303 AST_CONTROL_SRCUPDATE = 20, /*!< Indicate source of media has changed */ 00304 AST_CONTROL_T38_PARAMETERS = 24, /*!< T38 state change request/notification with parameters */ 00305 };
| enum ast_control_t38 |
- Enumerator:
Definition at line 307 of file frame.h.
00307 { 00308 AST_T38_REQUEST_NEGOTIATE = 1, /*!< Request T38 on a channel (voice to fax) */ 00309 AST_T38_REQUEST_TERMINATE, /*!< Terminate T38 on a channel (fax to voice) */ 00310 AST_T38_NEGOTIATED, /*!< T38 negotiated (fax mode) */ 00311 AST_T38_TERMINATED, /*!< T38 terminated (back to voice) */ 00312 AST_T38_REFUSED /*!< T38 refused for some reason (usually rejected by remote end) */ 00313 };
| enum ast_control_t38_rate |
- Enumerator:
-
AST_T38_RATE_2400 AST_T38_RATE_4800 AST_T38_RATE_7200 AST_T38_RATE_9600 AST_T38_RATE_12000 AST_T38_RATE_14400
Definition at line 315 of file frame.h.
00315 { 00316 AST_T38_RATE_2400 = 0, 00317 AST_T38_RATE_4800, 00318 AST_T38_RATE_7200, 00319 AST_T38_RATE_9600, 00320 AST_T38_RATE_12000, 00321 AST_T38_RATE_14400, 00322 };
Definition at line 324 of file frame.h.
00324 { 00325 AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF = 0, 00326 AST_T38_RATE_MANAGEMENT_LOCAL_TCF, 00327 };
| enum ast_frame_type |
Frame types.
- Note:
- It is important that the values of each frame type are never changed, because it will break backwards compatability with older versions. This is because these constants are transmitted directly over IAX2.
- Enumerator:
Definition at line 97 of file frame.h.
00097 { 00098 /*! DTMF end event, subclass is the digit */ 00099 AST_FRAME_DTMF_END = 1, 00100 /*! Voice data, subclass is AST_FORMAT_* */ 00101 AST_FRAME_VOICE, 00102 /*! Video frame, maybe?? :) */ 00103 AST_FRAME_VIDEO, 00104 /*! A control frame, subclass is AST_CONTROL_* */ 00105 AST_FRAME_CONTROL, 00106 /*! An empty, useless frame */ 00107 AST_FRAME_NULL, 00108 /*! Inter Asterisk Exchange private frame type */ 00109 AST_FRAME_IAX, 00110 /*! Text messages */ 00111 AST_FRAME_TEXT, 00112 /*! Image Frames */ 00113 AST_FRAME_IMAGE, 00114 /*! HTML Frame */ 00115 AST_FRAME_HTML, 00116 /*! Comfort Noise frame (subclass is level of CNG in -dBov), 00117 body may include zero or more 8-bit quantization coefficients */ 00118 AST_FRAME_CNG, 00119 /*! Modem-over-IP data streams */ 00120 AST_FRAME_MODEM, 00121 /*! DTMF begin event, subclass is the digit */ 00122 AST_FRAME_DTMF_BEGIN, 00123 };
Function Documentation
| int __ast_smoother_feed | ( | struct ast_smoother * | s, | |
| struct ast_frame * | f, | |||
| int | swap | |||
| ) |
Definition at line 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::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, f->data, 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.
- Parameters:
-
codec codec number (1,2,4,8,16,etc.)
- Returns:
- This returns a static string identifying the format on success, 0 on error.
Definition at line 635 of file frame.c.
References ARRAY_LEN, and ast_format_list::desc.
Referenced by moh_alloc(), show_codec_n(), and show_codecs().
00636 { 00637 int x; 00638 char *ret = "unknown"; 00639 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00640 if (AST_FORMAT_LIST[x].bits == codec) { 00641 ret = AST_FORMAT_LIST[x].desc; 00642 break; 00643 } 00644 } 00645 return ret; 00646 }
| 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, 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, f->datalen); 01420 break; 01421 case AST_FORMAT_G723_1: 01422 samples = g723_samples(f->data, 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))[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 639 of file frame.h.
References AST_FORMAT_ILBC.
Referenced by __get_from_jb(), and jb_get_and_deliver().
00640 { 00641 return (format == AST_FORMAT_ILBC) ? 30 : 20; 00642 }
| 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 1053 of file frame.c.
References ARRAY_LEN, ast_codec_pref_remove(), and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow().
01054 { 01055 int x, newindex = 0; 01056 01057 ast_codec_pref_remove(pref, format); 01058 01059 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01060 if (AST_FORMAT_LIST[x].bits == format) { 01061 newindex = x + 1; 01062 break; 01063 } 01064 } 01065 01066 if (newindex) { 01067 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01068 if (!pref->order[x]) { 01069 pref->order[x] = newindex; 01070 break; 01071 } 01072 } 01073 } 01074 01075 return x; 01076 }
| 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 955 of file frame.c.
References ast_codec_pref::order.
Referenced by check_access(), create_addr(), dump_prefs(), and socket_process().
00956 { 00957 int x, differential = (int) 'A', mem; 00958 char *from, *to; 00959 00960 if (right) { 00961 from = pref->order; 00962 to = buf; 00963 mem = size; 00964 } else { 00965 to = pref->order; 00966 from = buf; 00967 mem = 32; 00968 } 00969 00970 memset(to, 0, mem); 00971 for (x = 0; x < 32 ; x++) { 00972 if (!from[x]) 00973 break; 00974 to[x] = right ? (from[x] + differential) : (from[x] - differential); 00975 } 00976 }
| 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, index = -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 index = x; 01164 break; 01165 } 01166 } 01167 01168 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01169 if (pref->order[x] == (index + 1)) { 01170 framems = pref->framing[x]; 01171 break; 01172 } 01173 } 01174 01175 /* size validation */ 01176 if (!framems) 01177 framems = AST_FORMAT_LIST[index].def_ms; 01178 01179 if (AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */ 01180 framems -= framems % AST_FORMAT_LIST[index].inc_ms; 01181 01182 if (framems < AST_FORMAT_LIST[index].min_ms) 01183 framems = AST_FORMAT_LIST[index].min_ms; 01184 01185 if (framems > AST_FORMAT_LIST[index].max_ms) 01186 framems = AST_FORMAT_LIST[index].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 1013 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().
01014 { 01015 int slot = 0; 01016 01017 01018 if ((index >= 0) && (index < sizeof(pref->order))) { 01019 slot = pref->order[index]; 01020 } 01021 01022 return slot ? AST_FORMAT_LIST[slot-1].bits : 0; 01023 }
| void ast_codec_pref_init | ( | struct ast_codec_pref * | pref | ) |
| 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 1079 of file frame.c.
References ARRAY_LEN, ast_codec_pref::framing, and ast_codec_pref::order.
Referenced by create_addr().
01080 { 01081 int x, newindex = 0; 01082 01083 /* First step is to get the codecs "index number" */ 01084 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01085 if (AST_FORMAT_LIST[x].bits == format) { 01086 newindex = x + 1; 01087 break; 01088 } 01089 } 01090 /* Done if its unknown */ 01091 if (!newindex) 01092 return; 01093 01094 /* Now find any existing occurrence, or the end */ 01095 for (x = 0; x < 32; x++) { 01096 if (!pref->order[x] || pref->order[x] == newindex) 01097 break; 01098 } 01099 01100 if (only_if_existing && !pref->order[x]) 01101 return; 01102 01103 /* Move down to make space to insert - either all the way to the end, 01104 or as far as the existing location (which will be overwritten) */ 01105 for (; x > 0; x--) { 01106 pref->order[x] = pref->order[x - 1]; 01107 pref->framing[x] = pref->framing[x - 1]; 01108 } 01109 01110 /* And insert the new entry */ 01111 pref->order[0] = newindex; 01112 pref->framing[0] = 0; /* ? */ 01113 }
| void ast_codec_pref_remove | ( | struct ast_codec_pref * | pref, | |
| int | format | |||
| ) |
Remove audio a codec from a preference list.
Definition at line 1026 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().
01027 { 01028 struct ast_codec_pref oldorder; 01029 int x, y = 0; 01030 int slot; 01031 int size; 01032 01033 if (!pref->order[0]) 01034 return; 01035 01036 memcpy(&oldorder, pref, sizeof(oldorder)); 01037 memset(pref, 0, sizeof(*pref)); 01038 01039 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01040 slot = oldorder.order[x]; 01041 size = oldorder.framing[x]; 01042 if (! slot) 01043 break; 01044 if (AST_FORMAT_LIST[slot-1].bits != format) { 01045 pref->order[y] = slot; 01046 pref->framing[y++] = size; 01047 } 01048 } 01049 01050 }
| int ast_codec_pref_setsize | ( | struct ast_codec_pref * | pref, | |
| int | format, | |||
| int | framems | |||
| ) |
Set packet size for codec.
Definition at line 1116 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().
01117 { 01118 int x, index = -1; 01119 01120 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01121 if (AST_FORMAT_LIST[x].bits == format) { 01122 index = x; 01123 break; 01124 } 01125 } 01126 01127 if (index < 0) 01128 return -1; 01129 01130 /* size validation */ 01131 if (!framems) 01132 framems = AST_FORMAT_LIST[index].def_ms; 01133 01134 if (AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */ 01135 framems -= framems % AST_FORMAT_LIST[index].inc_ms; 01136 01137 if (framems < AST_FORMAT_LIST[index].min_ms) 01138 framems = AST_FORMAT_LIST[index].min_ms; 01139 01140 if (framems > AST_FORMAT_LIST[index].max_ms) 01141 framems = AST_FORMAT_LIST[index].max_ms; 01142 01143 01144 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01145 if (pref->order[x] == (index + 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 978 of file frame.c.
References ast_codec_pref_index(), and ast_getformatname().
Referenced by dump_prefs(), and socket_process().
00979 { 00980 int x, codec; 00981 size_t total_len, slen; 00982 char *formatname; 00983 00984 memset(buf,0,size); 00985 total_len = size; 00986 buf[0] = '('; 00987 total_len--; 00988 for(x = 0; x < 32 ; x++) { 00989 if (total_len <= 0) 00990 break; 00991 if (!(codec = ast_codec_pref_index(pref,x))) 00992 break; 00993 if ((formatname = ast_getformatname(codec))) { 00994 slen = strlen(formatname); 00995 if (slen > total_len) 00996 break; 00997 strncat(buf, formatname, total_len - 1); /* safe */ 00998 total_len -= slen; 00999 } 01000 if (total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) { 01001 strncat(buf, "|", total_len - 1); /* safe */ 01002 total_len--; 01003 } 01004 } 01005 if (total_len) { 01006 strncat(buf, ")", total_len - 1); /* safe */ 01007 total_len--; 01008 } 01009 01010 return size - total_len; 01011 }
| static force_inline int ast_format_rate | ( | int | format | ) | [static] |
Get the sample rate for a given format.
Definition at line 666 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().
00667 { 00668 if (format == AST_FORMAT_G722 || format == AST_FORMAT_SLINEAR16) 00669 return 16000; 00670 00671 return 8000; 00672 }
| int ast_frame_adjust_volume | ( | struct ast_frame * | f, | |
| int | adjustment | |||
| ) |
Adjusts the volume of the audio samples contained in a frame.
- Parameters:
-
f The frame containing the samples (must be AST_FRAME_VOICE and AST_FORMAT_SLINEAR) adjustment The number of dB to adjust up or down.
- Returns:
- 0 for success, non-zero for an error
Definition at line 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::samples, and ast_frame::subclass.
Referenced by audiohook_read_frame_single(), conf_run(), and volume_callback().
01498 { 01499 int count; 01500 short *fdata = f->data; 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 737 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_control_t38_parameters::request_response, ast_frame::subclass, and term_color().
Referenced by __ast_read(), and ast_write().
00738 { 00739 const char noname[] = "unknown"; 00740 char ftype[40] = "Unknown Frametype"; 00741 char cft[80]; 00742 char subclass[40] = "Unknown Subclass"; 00743 char csub[80]; 00744 char moreinfo[40] = ""; 00745 char cn[60]; 00746 char cp[40]; 00747 char cmn[40]; 00748 const char *message = "Unknown"; 00749 00750 if (!name) 00751 name = noname; 00752 00753 00754 if (!f) { 00755 ast_verbose("%s [ %s (NULL) ] [%s]\n", 00756 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00757 term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00758 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00759 return; 00760 } 00761 /* XXX We should probably print one each of voice and video when the format changes XXX */ 00762 if (f->frametype == AST_FRAME_VOICE) 00763 return; 00764 if (f->frametype == AST_FRAME_VIDEO) 00765 return; 00766 switch(f->frametype) { 00767 case AST_FRAME_DTMF_BEGIN: 00768 strcpy(ftype, "DTMF Begin"); 00769 subclass[0] = f->subclass; 00770 subclass[1] = '\0'; 00771 break; 00772 case AST_FRAME_DTMF_END: 00773 strcpy(ftype, "DTMF End"); 00774 subclass[0] = f->subclass; 00775 subclass[1] = '\0'; 00776 break; 00777 case AST_FRAME_CONTROL: 00778 strcpy(ftype, "Control"); 00779 switch(f->subclass) { 00780 case AST_CONTROL_HANGUP: 00781 strcpy(subclass, "Hangup"); 00782 break; 00783 case AST_CONTROL_RING: 00784 strcpy(subclass, "Ring"); 00785 break; 00786 case AST_CONTROL_RINGING: 00787 strcpy(subclass, "Ringing"); 00788 break; 00789 case AST_CONTROL_ANSWER: 00790 strcpy(subclass, "Answer"); 00791 break; 00792 case AST_CONTROL_BUSY: 00793 strcpy(subclass, "Busy"); 00794 break; 00795 case AST_CONTROL_TAKEOFFHOOK: 00796 strcpy(subclass, "Take Off Hook"); 00797 break; 00798 case AST_CONTROL_OFFHOOK: 00799 strcpy(subclass, "Line Off Hook"); 00800 break; 00801 case AST_CONTROL_CONGESTION: 00802 strcpy(subclass, "Congestion"); 00803 break; 00804 case AST_CONTROL_FLASH: 00805 strcpy(subclass, "Flash"); 00806 break; 00807 case AST_CONTROL_WINK: 00808 strcpy(subclass, "Wink"); 00809 break; 00810 case AST_CONTROL_OPTION: 00811 strcpy(subclass, "Option"); 00812 break; 00813 case AST_CONTROL_RADIO_KEY: 00814 strcpy(subclass, "Key Radio"); 00815 break; 00816 case AST_CONTROL_RADIO_UNKEY: 00817 strcpy(subclass, "Unkey Radio"); 00818 break; 00819 case AST_CONTROL_HOLD: 00820 strcpy(subclass, "Hold"); 00821 break; 00822 case AST_CONTROL_UNHOLD: 00823 strcpy(subclass, "Unhold"); 00824 break; 00825 case AST_CONTROL_T38_PARAMETERS: 00826 if (f->datalen != sizeof(struct ast_control_t38_parameters)) { 00827 message = "Invalid"; 00828 } else { 00829 struct ast_control_t38_parameters *parameters = f->data; 00830 enum ast_control_t38 state = parameters->request_response; 00831 if (state == AST_T38_REQUEST_NEGOTIATE) 00832 message = "Negotiation Requested"; 00833 else if (state == AST_T38_REQUEST_TERMINATE) 00834 message = "Negotiation Request Terminated"; 00835 else if (state == AST_T38_NEGOTIATED) 00836 message = "Negotiated"; 00837 else if (state == AST_T38_TERMINATED) 00838 message = "Terminated"; 00839 else if (state == AST_T38_REFUSED) 00840 message = "Refused"; 00841 } 00842 snprintf(subclass, sizeof(subclass), "T38_Parameters/%s", message); 00843 break; 00844 case -1: 00845 strcpy(subclass, "Stop generators"); 00846 break; 00847 default: 00848 snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass); 00849 } 00850 break; 00851 case AST_FRAME_NULL: 00852 strcpy(ftype, "Null Frame"); 00853 strcpy(subclass, "N/A"); 00854 break; 00855 case AST_FRAME_IAX: 00856 /* Should never happen */ 00857 strcpy(ftype, "IAX Specific"); 00858 snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass); 00859 break; 00860 case AST_FRAME_TEXT: 00861 strcpy(ftype, "Text"); 00862 strcpy(subclass, "N/A"); 00863 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00864 break; 00865 case AST_FRAME_IMAGE: 00866 strcpy(ftype, "Image"); 00867 snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass)); 00868 break; 00869 case AST_FRAME_HTML: 00870 strcpy(ftype, "HTML"); 00871 switch(f->subclass) { 00872 case AST_HTML_URL: 00873 strcpy(subclass, "URL"); 00874 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00875 break; 00876 case AST_HTML_DATA: 00877 strcpy(subclass, "Data"); 00878 break; 00879 case AST_HTML_BEGIN: 00880 strcpy(subclass, "Begin"); 00881 break; 00882 case AST_HTML_END: 00883 strcpy(subclass, "End"); 00884 break; 00885 case AST_HTML_LDCOMPLETE: 00886 strcpy(subclass, "Load Complete"); 00887 break; 00888 case AST_HTML_NOSUPPORT: 00889 strcpy(subclass, "No Support"); 00890 break; 00891 case AST_HTML_LINKURL: 00892 strcpy(subclass, "Link URL"); 00893 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00894 break; 00895 case AST_HTML_UNLINK: 00896 strcpy(subclass, "Unlink"); 00897 break; 00898 case AST_HTML_LINKREJECT: 00899 strcpy(subclass, "Link Reject"); 00900 break; 00901 default: 00902 snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass); 00903 break; 00904 } 00905 break; 00906 case AST_FRAME_MODEM: 00907 strcpy(ftype, "Modem"); 00908 switch (f->subclass) { 00909 case AST_MODEM_T38: 00910 strcpy(subclass, "T.38"); 00911 break; 00912 case AST_MODEM_V150: 00913 strcpy(subclass, "V.150"); 00914 break; 00915 default: 00916 snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass); 00917 break; 00918 } 00919 break; 00920 default: 00921 snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype); 00922 } 00923 if (!ast_strlen_zero(moreinfo)) 00924 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n", 00925 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00926 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00927 f->frametype, 00928 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00929 f->subclass, 00930 term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)), 00931 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00932 else 00933 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n", 00934 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00935 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00936 f->frametype, 00937 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00938 f->subclass, 00939 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00940 }
| struct ast_frame* ast_frame_enqueue | ( | struct ast_frame * | head, | |
| struct ast_frame * | f, | |||
| int | maxlen, | |||
| int | dupe | |||
| ) | [read] |
Appends a frame to the end of a list of frames, truncating the maximum length of the list.
| void ast_frame_free | ( | struct ast_frame * | fr, | |
| int | cache | |||
| ) |
Requests a frame to be allocated.
- Parameters:
-
source Request a frame be allocated. source is an optional source of the frame, len is the requested length, or "0" if the caller will supply the buffer
- Parameters:
-
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(), AST_LIST_NEXT, ast_frame::frame_list, and ast_frame::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.
- Parameters:
-
f1 The first frame (which will contain the result) f2 The second frame
- Returns:
- 0 for success, non-zero for an error
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::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, data2 = f2->data; 01535 count < f1->samples; 01536 count++, data1++, data2++) 01537 ast_slinear_saturated_add(data1, data2); 01538 01539 return 0; 01540 }
Copies a frame.
- Parameters:
-
fr frame to copy Duplicates a frame -- should only rarely be used, typically frisolate is good enough
- Returns:
- Returns a frame on success, NULL on error
Definition at line 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::samples, ast_frame::seqno, ast_frame_cache::size, ast_frame::src, ast_frame::subclass, and ast_frame::ts.
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(), and recordthread().
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 = buf + sizeof(*out) + AST_FRIENDLY_OFFSET; 00517 memcpy(out->data, f->data, out->datalen); 00518 } 00519 if (srclen > 0) { 00520 /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */ 00521 char *src; 00522 out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00523 src = (char *) out->src; 00524 /* Must have space since we allocated for it */ 00525 strcpy(src, f->src); 00526 } 00527 ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO); 00528 out->ts = f->ts; 00529 out->len = f->len; 00530 out->seqno = f->seqno; 00531 return out; 00532 }
Makes a frame independent of any static storage.
- Parameters:
-
fr frame to act upon Take a frame, and if it's not been malloc'd, make a malloc'd copy and if the data hasn't been malloced then make the data malloc'd. If you need to store frames, say for queueing, then you should call this function.
- Returns:
- Returns a frame on success, NULL on error
- Note:
- This function may modify the frame passed to it, so you must not assume the frame will be intact after the isolated frame has been produced. In other words, calling this function on a frame should be the last operation you do with that frame before freeing it (or exiting the block, if the frame is on the stack.)
Definition at line 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::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, fr->datalen); 00447 out->data = 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 550 of file frame.c.
References ARRAY_LEN.
00551 { 00552 *size = ARRAY_LEN(AST_FORMAT_LIST); 00553 return AST_FORMAT_LIST; 00554 }
| struct ast_format_list* ast_get_format_list_index | ( | int | index | ) | [read] |
| int ast_getformatbyname | ( | const char * | name | ) |
Gets a format from a name.
- Parameters:
-
name string of format
- Returns:
- This returns the form of the format in binary on success, 0 on error.
Definition at line 617 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().
00618 { 00619 int x, all, format = 0; 00620 00621 all = strcasecmp(name, "all") ? 0 : 1; 00622 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00623 if (all || 00624 !strcasecmp(AST_FORMAT_LIST[x].name,name) || 00625 !strcasecmp(AST_FORMAT_LIST[x].name,ast_expand_codec_alias(name))) { 00626 format |= AST_FORMAT_LIST[x].bits; 00627 if (!all) 00628 break; 00629 } 00630 } 00631 00632 return format; 00633 }
| char* ast_getformatname | ( | int | format | ) |
Get the name of a format.
- Parameters:
-
format id of format
- Returns:
- A static string containing the name of the format or "unknown" if unknown.
Definition at line 556 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().
00557 { 00558 int x; 00559 char *ret = "unknown"; 00560 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00561 if (AST_FORMAT_LIST[x].bits == format) { 00562 ret = AST_FORMAT_LIST[x].name; 00563 break; 00564 } 00565 } 00566 return ret; 00567 }
| char* ast_getformatname_multiple | ( | char * | buf, | |
| size_t | size, | |||
| int | format | |||
| ) |
Get the names of a set of formats.
- Parameters:
-
buf a buffer for the output string size size of buf (bytes) format the format (combined IDs of codecs) Prints a list of readable codec names corresponding to "format". ex: for format=AST_FORMAT_GSM|AST_FORMAT_SPEEX|AST_FORMAT_ILBC it will return "0x602 (GSM|SPEEX|ILBC)"
- Returns:
- The return value is buf.
Definition at line 569 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().
00570 { 00571 int x; 00572 unsigned len; 00573 char *start, *end = buf; 00574 00575 if (!size) 00576 return buf; 00577 snprintf(end, size, "0x%x (", format); 00578 len = strlen(end); 00579 end += len; 00580 size -= len; 00581 start = end; 00582 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00583 if (AST_FORMAT_LIST[x].bits & format) { 00584 snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name); 00585 len = strlen(end); 00586 end += len; 00587 size -= len; 00588 } 00589 } 00590 if (start == end) 00591 ast_copy_string(start, "nothing)", size); 00592 else if (size > 1) 00593 *(end -1) = ')'; 00594 return buf; 00595 }
| int ast_parse_allow_disallow | ( | struct ast_codec_pref * | pref, | |
| int * | mask, | |||
| const char * | list, | |||
| int | allowing | |||
| ) |
Parse an "allow" or "deny" line in a channel or device configuration and update the capabilities mask and pref if provided. Video codecs are not added to codec preference lists, since we can not transcode.
- Returns:
- Returns number of errors encountered during parsing
Definition at line 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::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 = 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, 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.
- Parameters:
-
s the smoother to reconfigure bytes the desired number of bytes per output frame
- Returns:
- nothing
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 534 of file frame.c.
Referenced by __ast_smoother_feed(), iax_frame_wrap(), phone_write_buf(), and smoother_frame_feed().
00535 { 00536 int i; 00537 unsigned short *dst_s = dst; 00538 const unsigned short *src_s = src; 00539 00540 for (i = 0; i < samples; i++) 00541 dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8); 00542 }
Variable Documentation
| struct ast_frame ast_null_frame |
Queueing a null frame is fairly common, so we declare a global null frame object for this purpose instead of having to declare one on the stack
Definition at line 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(), ast_do_masquerade(), ast_rtcp_read(), ast_rtp_read(), ast_softhangup_nolock(), ast_udptl_read(), conf_run(), console_read(), features_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().
