#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) |
ast_smoother * | ast_smoother_new (int bytes) |
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_MAX_TEXT (1 << 28)) |
#define | AST_FORMAT_MP4_VIDEO (1 << 22) |
#define | AST_FORMAT_PNG (1 << 17) |
#define | AST_FORMAT_SLINEAR (1 << 6) |
#define | AST_FORMAT_SLINEAR16 (1 << 15) |
#define | AST_FORMAT_SPEEX (1 << 9) |
#define | AST_FORMAT_T140 (1 << 27) |
#define | AST_FORMAT_T140RED (1 << 26) |
#define | AST_FORMAT_TEXT_MASK (((1 << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK)) |
#define | AST_FORMAT_ULAW (1 << 2) |
#define | AST_FORMAT_VIDEO_MASK (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) |
#define | ast_frame_byteswap_be(fr) do { ; } while(0) |
#define | ast_frame_byteswap_le(fr) do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data.ptr, __f->data.ptr, __f->samples); } while(0) |
#define | AST_FRAME_DTMF AST_FRAME_DTMF_END |
#define | AST_FRAME_SET_BUFFER(fr, _base, _ofs, _datalen) |
#define | ast_frfree(fr) ast_frame_free(fr, 1) |
#define | AST_FRIENDLY_OFFSET 64 |
Offset into a frame's data buffer. | |
#define | AST_HTML_BEGIN 4 |
#define | AST_HTML_DATA 2 |
#define | AST_HTML_END 8 |
#define | AST_HTML_LDCOMPLETE 16 |
#define | AST_HTML_LINKREJECT 20 |
#define | AST_HTML_LINKURL 18 |
#define | AST_HTML_NOSUPPORT 17 |
#define | AST_HTML_UNLINK 19 |
#define | AST_HTML_URL 1 |
#define | AST_MALLOCD_DATA (1 << 1) |
#define | AST_MALLOCD_HDR (1 << 0) |
#define | AST_MALLOCD_SRC (1 << 2) |
#define | AST_MIN_OFFSET 32 |
#define | AST_MODEM_T38 1 |
#define | AST_MODEM_V150 2 |
#define | AST_OPTION_AUDIO_MODE 4 |
#define | AST_OPTION_ECHOCAN 8 |
#define | AST_OPTION_FLAG_ACCEPT 1 |
#define | AST_OPTION_FLAG_ANSWER 5 |
#define | AST_OPTION_FLAG_QUERY 4 |
#define | AST_OPTION_FLAG_REJECT 2 |
#define | AST_OPTION_FLAG_REQUEST 0 |
#define | AST_OPTION_FLAG_WTF 6 |
#define | AST_OPTION_OPRMODE 7 |
#define | AST_OPTION_RELAXDTMF 3 |
#define | AST_OPTION_RXGAIN 6 |
#define | AST_OPTION_T38_STATE 10 |
#define | AST_OPTION_TDD 2 |
#define | AST_OPTION_TONE_VERIFY 1 |
#define | AST_OPTION_TXGAIN 5 |
#define | AST_SMOOTHER_FLAG_BE (1 << 1) |
#define | AST_SMOOTHER_FLAG_G729 (1 << 0) |
Enumerations | |
enum | { AST_FRFLAG_HAS_TIMING_INFO = (1 << 0) } |
enum | ast_control_frame_type { AST_CONTROL_HANGUP = 1, AST_CONTROL_RING = 2, AST_CONTROL_RINGING = 3, AST_CONTROL_ANSWER = 4, AST_CONTROL_BUSY = 5, AST_CONTROL_TAKEOFFHOOK = 6, AST_CONTROL_OFFHOOK = 7, AST_CONTROL_CONGESTION = 8, AST_CONTROL_FLASH = 9, AST_CONTROL_WINK = 10, AST_CONTROL_OPTION = 11, AST_CONTROL_RADIO_KEY = 12, AST_CONTROL_RADIO_UNKEY = 13, AST_CONTROL_PROGRESS = 14, AST_CONTROL_PROCEEDING = 15, AST_CONTROL_HOLD = 16, AST_CONTROL_UNHOLD = 17, AST_CONTROL_VIDUPDATE = 18, _XXX_AST_CONTROL_T38 = 19, AST_CONTROL_SRCUPDATE = 20, AST_CONTROL_T38_PARAMETERS = 24, AST_CONTROL_SRCCHANGE = 25 } |
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. | |
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) |
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 Frees a frame or list of frames. | |
int | ast_frame_slinear_sum (struct ast_frame *f1, struct ast_frame *f2) |
Sums two frames of audio samples. | |
ast_frame * | ast_frdup (const struct ast_frame *fr) |
Copies a frame. | |
ast_frame * | ast_frisolate (struct ast_frame *fr) |
Makes a frame independent of any static storage. | |
ast_format_list * | ast_get_format_list (size_t *size) |
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 | |
ast_frame | ast_null_frame |
Definition in file frame.h.
#define AST_FORMAT_ADPCM (1 << 5) |
ADPCM (IMA)
Definition at line 244 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 240 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(), oh323_rtp_read(), pcm_seek(), pcm_write(), and start_rtp().
#define AST_FORMAT_AUDIO_MASK ((1 << 16)-1) |
Maximum audio mask
Definition at line 264 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 258 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 234 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 256 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 242 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 250 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 236 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 270 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 272 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 276 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 254 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 266 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 248 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 246 of file frame.h.
Referenced by __ast_play_and_record(), __ast_register_translator(), _moh_class_malloc(), action_originate(), agent_new(), alsa_new(), alsa_read(), alsa_request(), ast_audiohook_read_frame(), ast_best_codec(), ast_channel_make_compatible_helper(), ast_channel_start_silence_generator(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_call_progress(), ast_dsp_noise(), ast_dsp_process(), ast_dsp_silence(), ast_frame_adjust_volume(), ast_frame_slinear_sum(), ast_rtp_read(), ast_slinfactory_feed(), ast_speech_new(), attempt_reconnect(), audio_audiohook_write_list(), audiohook_read_frame_both(), audiohook_read_frame_single(), background_detect_exec(), build_conf(), chanspy_exec(), conf_run(), connect_link(), dahdi_read(), dahdi_translate(), dahdi_write(), dictate_exec(), do_waiting(), eagi_exec(), extenspy_exec(), fax_generator_generate(), find_transcoders(), handle_jack_audio(), handle_recordfile(), handle_speechcreate(), handle_speechrecognize(), iax_frame_wrap(), ices_exec(), init_outgoing(), is_encoder(), isAnsweringMachine(), jack_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(), mixmonitor_thread(), mp3_exec(), nbs_request(), nbs_xwrite(), NBScat_exec(), ogg_vorbis_read(), ogg_vorbis_write(), oh323_rtp_read(), orig_app(), orig_exten(), oss_new(), oss_read(), oss_request(), parkandannounce_exec(), phone_new(), phone_read(), phone_request(), phone_setup(), phone_write(), playtones_alloc(), 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(), 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 262 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 252 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 283 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 281 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 286 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 238 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 279 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) |
#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 |
Definition at line 125 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_feature_request_and_dial(), 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(), jingle_handle_dtmf(), keypad_digit(), mgcp_rtp_read(), misdn_bridge(), mp3_exec(), NBScat_exec(), oh323_rtp_read(), phone_exception(), process_ast_dsp(), receive_dtmf_digits(), rpt(), rpt_call(), send_waveform_to_channel(), sip_rtp_read(), speech_background(), ss_thread(), unistim_do_senddigit(), unistim_senddigit_end(), volume_callback(), and wait_for_winner().
#define AST_FRAME_SET_BUFFER | ( | fr, | |||
_base, | |||||
_ofs, | |||||
_datalen | ) |
Value:
{ \ (fr)->data.ptr = (char *)_base + (_ofs); \ (fr)->offset = (_ofs); \ (fr)->datalen = (_datalen); \ }
Definition at line 175 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 454 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_feature_request_and_dial(), ast_jb_destroy(), ast_jb_put(), ast_readaudio_callback(), ast_readvideo_callback(), ast_recvtext(), ast_rtp_write(), ast_safe_sleep_conditional(), ast_send_image(), ast_slinfactory_destroy(), ast_slinfactory_feed(), ast_slinfactory_flush(), ast_slinfactory_read(), ast_tonepair(), ast_translate(), ast_udptl_bridge(), ast_waitfordigit_full(), ast_write(), ast_writestream(), async_wait(), audio_audiohook_write_list(), autoservice_run(), background_detect_exec(), bridge_native_loop(), bridge_p2p_loop(), builtin_atxfer(), calc_cost(), channel_spy(), check_goto_on_transfer(), conf_exec(), conf_flush(), conf_free(), conf_run(), create_jb(), dahdi_bridge(), dial_exec_full(), dictate_exec(), disa_exec(), do_idle_thread(), do_waiting(), echo_exec(), eivr_comm(), find_cache(), gen_generate(), handle_cli_file_convert(), handle_recordfile(), handle_speechrecognize(), iax_park_thread(), ices_exec(), isAnsweringMachine(), jb_empty_and_reset_adaptive(), jb_empty_and_reset_fixed(), jb_get_and_deliver(), launch_asyncagi(), manage_parkinglot(), masq_park_call(), measurenoise(), moh_files_generator(), monitor_dial(), mp3_exec(), NBScat_exec(), read_frame(), receive_dtmf_digits(), recordthread(), rpt(), 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 196 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(), process_rfc3389(), send_tone_burst(), send_waveform_to_channel(), slinear_read(), sms_generate(), 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 222 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 224 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 214 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 202 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 200 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 204 of file frame.h.
Referenced by __frame_free(), ast_frisolate(), and speex_callback().
#define AST_MIN_OFFSET 32 |
#define AST_MODEM_T38 1 |
T.38 Fax-over-IP
Definition at line 208 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 368 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 390 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 365 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 384 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 396 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 362 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 359 of file frame.h.
Referenced by conf_run(), dahdi_hangup(), dahdi_setoption(), rpt(), and rpt_exec().
#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 376 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 346 of file frame.h.
Referenced by __ast_smoother_feed(), ast_smoother_read(), and smoother_frame_feed().
anonymous enum |
Definition at line 127 of file frame.h.
00127 { 00128 /*! This frame contains valid timing information */ 00129 AST_FRFLAG_HAS_TIMING_INFO = (1 << 0), 00130 };
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 |
AST_CONTROL_SRCCHANGE | Media source has changed and requires a new RTP SSRC |
Definition at line 288 of file frame.h.
00288 { 00289 AST_CONTROL_HANGUP = 1, /*!< Other end has hungup */ 00290 AST_CONTROL_RING = 2, /*!< Local ring */ 00291 AST_CONTROL_RINGING = 3, /*!< Remote end is ringing */ 00292 AST_CONTROL_ANSWER = 4, /*!< Remote end has answered */ 00293 AST_CONTROL_BUSY = 5, /*!< Remote end is busy */ 00294 AST_CONTROL_TAKEOFFHOOK = 6, /*!< Make it go off hook */ 00295 AST_CONTROL_OFFHOOK = 7, /*!< Line is off hook */ 00296 AST_CONTROL_CONGESTION = 8, /*!< Congestion (circuits busy) */ 00297 AST_CONTROL_FLASH = 9, /*!< Flash hook */ 00298 AST_CONTROL_WINK = 10, /*!< Wink */ 00299 AST_CONTROL_OPTION = 11, /*!< Set a low-level option */ 00300 AST_CONTROL_RADIO_KEY = 12, /*!< Key Radio */ 00301 AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */ 00302 AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */ 00303 AST_CONTROL_PROCEEDING = 15, /*!< Indicate CALL PROCEEDING */ 00304 AST_CONTROL_HOLD = 16, /*!< Indicate call is placed on hold */ 00305 AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold */ 00306 AST_CONTROL_VIDUPDATE = 18, /*!< Indicate video frame update */ 00307 _XXX_AST_CONTROL_T38 = 19, /*!< T38 state change request/notification \deprecated This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead. */ 00308 AST_CONTROL_SRCUPDATE = 20, /*!< Indicate source of media has changed */ 00309 AST_CONTROL_T38_PARAMETERS = 24, /*!< T38 state change request/notification with parameters */ 00310 AST_CONTROL_SRCCHANGE = 25, /*!< Media source has changed and requires a new RTP SSRC */ 00311 };
enum ast_control_t38 |
Definition at line 313 of file frame.h.
00313 { 00314 AST_T38_REQUEST_NEGOTIATE = 1, /*!< Request T38 on a channel (voice to fax) */ 00315 AST_T38_REQUEST_TERMINATE, /*!< Terminate T38 on a channel (fax to voice) */ 00316 AST_T38_NEGOTIATED, /*!< T38 negotiated (fax mode) */ 00317 AST_T38_TERMINATED, /*!< T38 terminated (back to voice) */ 00318 AST_T38_REFUSED /*!< T38 refused for some reason (usually rejected by remote end) */ 00319 };
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 321 of file frame.h.
00321 { 00322 AST_T38_RATE_2400 = 0, 00323 AST_T38_RATE_4800, 00324 AST_T38_RATE_7200, 00325 AST_T38_RATE_9600, 00326 AST_T38_RATE_12000, 00327 AST_T38_RATE_14400, 00328 };
Definition at line 330 of file frame.h.
00330 { 00331 AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF = 0, 00332 AST_T38_RATE_MANAGEMENT_LOCAL_TCF, 00333 };
enum ast_frame_type |
Frame types.
Definition at line 98 of file frame.h.
00098 { 00099 /*! DTMF end event, subclass is the digit */ 00100 AST_FRAME_DTMF_END = 1, 00101 /*! Voice data, subclass is AST_FORMAT_* */ 00102 AST_FRAME_VOICE, 00103 /*! Video frame, maybe?? :) */ 00104 AST_FRAME_VIDEO, 00105 /*! A control frame, subclass is AST_CONTROL_* */ 00106 AST_FRAME_CONTROL, 00107 /*! An empty, useless frame */ 00108 AST_FRAME_NULL, 00109 /*! Inter Asterisk Exchange private frame type */ 00110 AST_FRAME_IAX, 00111 /*! Text messages */ 00112 AST_FRAME_TEXT, 00113 /*! Image Frames */ 00114 AST_FRAME_IMAGE, 00115 /*! HTML Frame */ 00116 AST_FRAME_HTML, 00117 /*! Comfort Noise frame (subclass is level of CNG in -dBov), 00118 body may include zero or more 8-bit quantization coefficients */ 00119 AST_FRAME_CNG, 00120 /*! Modem-over-IP data streams */ 00121 AST_FRAME_MODEM, 00122 /*! DTMF begin event, subclass is the digit */ 00123 AST_FRAME_DTMF_BEGIN, 00124 };
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(), f, LOG_WARNING, s, smoother_frame_feed(), and SMOOTHER_SIZE.
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 642 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, and ast_format_list::desc.
Referenced by moh_alloc(), show_codec_n(), and show_codecs().
00643 { 00644 int x; 00645 char *ret = "unknown"; 00646 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00647 if (AST_FORMAT_LIST[x].bits == codec) { 00648 ret = AST_FORMAT_LIST[x].desc; 00649 break; 00650 } 00651 } 00652 return ret; 00653 }
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 1199 of file frame.c.
References ARRAY_LEN, ast_best_codec(), ast_debug, AST_FORMAT_AUDIO_MASK, AST_FORMAT_LIST, ast_format_list::bits, and ast_codec_pref::order.
Referenced by __oh323_new(), gtalk_new(), jingle_new(), process_sdp(), sip_new(), and socket_process().
01200 { 01201 int x, ret = 0, slot; 01202 01203 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01204 slot = pref->order[x]; 01205 01206 if (!slot) 01207 break; 01208 if (formats & AST_FORMAT_LIST[slot-1].bits) { 01209 ret = AST_FORMAT_LIST[slot-1].bits; 01210 break; 01211 } 01212 } 01213 if (ret & AST_FORMAT_AUDIO_MASK) 01214 return ret; 01215 01216 ast_debug(4, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec"); 01217 01218 return find_best ? ast_best_codec(formats) : 0; 01219 }
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 1463 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().
01464 { 01465 int len = 0; 01466 01467 /* XXX Still need speex, g723, and lpc10 XXX */ 01468 switch(format) { 01469 case AST_FORMAT_G723_1: 01470 len = (samples / 240) * 20; 01471 break; 01472 case AST_FORMAT_ILBC: 01473 len = (samples / 240) * 50; 01474 break; 01475 case AST_FORMAT_GSM: 01476 len = (samples / 160) * 33; 01477 break; 01478 case AST_FORMAT_G729A: 01479 len = samples / 8; 01480 break; 01481 case AST_FORMAT_SLINEAR: 01482 case AST_FORMAT_SLINEAR16: 01483 len = samples * 2; 01484 break; 01485 case AST_FORMAT_ULAW: 01486 case AST_FORMAT_ALAW: 01487 len = samples; 01488 break; 01489 case AST_FORMAT_G722: 01490 case AST_FORMAT_ADPCM: 01491 case AST_FORMAT_G726: 01492 case AST_FORMAT_G726_AAL2: 01493 len = samples / 2; 01494 break; 01495 default: 01496 ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format)); 01497 } 01498 01499 return len; 01500 }
int ast_codec_get_samples | ( | struct ast_frame * | f | ) |
Returns the number of samples contained in the frame.
Definition at line 1419 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(), f, g723_samples(), LOG_WARNING, and speex_samples().
Referenced by ast_rtp_read(), isAnsweringMachine(), moh_generate(), schedule_delivery(), socket_process(), and socket_process_meta().
01420 { 01421 int samples=0; 01422 switch(f->subclass) { 01423 case AST_FORMAT_SPEEX: 01424 samples = speex_samples(f->data.ptr, f->datalen); 01425 break; 01426 case AST_FORMAT_G723_1: 01427 samples = g723_samples(f->data.ptr, f->datalen); 01428 break; 01429 case AST_FORMAT_ILBC: 01430 samples = 240 * (f->datalen / 50); 01431 break; 01432 case AST_FORMAT_GSM: 01433 samples = 160 * (f->datalen / 33); 01434 break; 01435 case AST_FORMAT_G729A: 01436 samples = f->datalen * 8; 01437 break; 01438 case AST_FORMAT_SLINEAR: 01439 case AST_FORMAT_SLINEAR16: 01440 samples = f->datalen / 2; 01441 break; 01442 case AST_FORMAT_LPC10: 01443 /* assumes that the RTP packet contains one LPC10 frame */ 01444 samples = 22 * 8; 01445 samples += (((char *)(f->data.ptr))[7] & 0x1) * 8; 01446 break; 01447 case AST_FORMAT_ULAW: 01448 case AST_FORMAT_ALAW: 01449 samples = f->datalen; 01450 break; 01451 case AST_FORMAT_G722: 01452 case AST_FORMAT_ADPCM: 01453 case AST_FORMAT_G726: 01454 case AST_FORMAT_G726_AAL2: 01455 samples = f->datalen * 2; 01456 break; 01457 default: 01458 ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass)); 01459 } 01460 return samples; 01461 }
static int ast_codec_interp_len | ( | int | format | ) | [inline, static] |
Gets duration in ms of interpolation frame for a format.
Definition at line 645 of file frame.h.
References AST_FORMAT_ILBC.
Referenced by __get_from_jb(), and jb_get_and_deliver().
00646 { 00647 return (format == AST_FORMAT_ILBC) ? 30 : 20; 00648 }
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 1059 of file frame.c.
References ARRAY_LEN, ast_codec_pref_remove(), AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow().
01060 { 01061 int x, newindex = 0; 01062 01063 ast_codec_pref_remove(pref, format); 01064 01065 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01066 if (AST_FORMAT_LIST[x].bits == format) { 01067 newindex = x + 1; 01068 break; 01069 } 01070 } 01071 01072 if (newindex) { 01073 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01074 if (!pref->order[x]) { 01075 pref->order[x] = newindex; 01076 break; 01077 } 01078 } 01079 } 01080 01081 return x; 01082 }
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 962 of file frame.c.
References ast_codec_pref::order.
Referenced by check_access(), create_addr(), dump_prefs(), and socket_process().
00963 { 00964 int x, differential = (int) 'A', mem; 00965 char *from, *to; 00966 00967 if (right) { 00968 from = pref->order; 00969 to = buf; 00970 mem = size; 00971 } else { 00972 to = pref->order; 00973 from = buf; 00974 mem = 32; 00975 } 00976 00977 memset(to, 0, mem); 00978 for (x = 0; x < 32 ; x++) { 00979 if (!from[x]) 00980 break; 00981 to[x] = right ? (from[x] + differential) : (from[x] - differential); 00982 } 00983 }
struct ast_format_list ast_codec_pref_getsize | ( | struct ast_codec_pref * | pref, | |
int | format | |||
) |
Get packet size for codec.
Definition at line 1160 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, 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().
01161 { 01162 int x, idx = -1, framems = 0; 01163 struct ast_format_list fmt = { 0, }; 01164 01165 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01166 if (AST_FORMAT_LIST[x].bits == format) { 01167 fmt = AST_FORMAT_LIST[x]; 01168 idx = x; 01169 break; 01170 } 01171 } 01172 01173 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01174 if (pref->order[x] == (idx + 1)) { 01175 framems = pref->framing[x]; 01176 break; 01177 } 01178 } 01179 01180 /* size validation */ 01181 if (!framems) 01182 framems = AST_FORMAT_LIST[idx].def_ms; 01183 01184 if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */ 01185 framems -= framems % AST_FORMAT_LIST[idx].inc_ms; 01186 01187 if (framems < AST_FORMAT_LIST[idx].min_ms) 01188 framems = AST_FORMAT_LIST[idx].min_ms; 01189 01190 if (framems > AST_FORMAT_LIST[idx].max_ms) 01191 framems = AST_FORMAT_LIST[idx].max_ms; 01192 01193 fmt.cur_ms = framems; 01194 01195 return fmt; 01196 }
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 1020 of file frame.c.
References AST_FORMAT_LIST, 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().
01021 { 01022 int slot = 0; 01023 01024 if ((idx >= 0) && (idx < sizeof(pref->order))) { 01025 slot = pref->order[idx]; 01026 } 01027 01028 return slot ? AST_FORMAT_LIST[slot - 1].bits : 0; 01029 }
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 1085 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_codec_pref::framing, and ast_codec_pref::order.
Referenced by create_addr().
01086 { 01087 int x, newindex = 0; 01088 01089 /* First step is to get the codecs "index number" */ 01090 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01091 if (AST_FORMAT_LIST[x].bits == format) { 01092 newindex = x + 1; 01093 break; 01094 } 01095 } 01096 /* Done if its unknown */ 01097 if (!newindex) 01098 return; 01099 01100 /* Now find any existing occurrence, or the end */ 01101 for (x = 0; x < 32; x++) { 01102 if (!pref->order[x] || pref->order[x] == newindex) 01103 break; 01104 } 01105 01106 if (only_if_existing && !pref->order[x]) 01107 return; 01108 01109 /* Move down to make space to insert - either all the way to the end, 01110 or as far as the existing location (which will be overwritten) */ 01111 for (; x > 0; x--) { 01112 pref->order[x] = pref->order[x - 1]; 01113 pref->framing[x] = pref->framing[x - 1]; 01114 } 01115 01116 /* And insert the new entry */ 01117 pref->order[0] = newindex; 01118 pref->framing[0] = 0; /* ? */ 01119 }
void ast_codec_pref_remove | ( | struct ast_codec_pref * | pref, | |
int | format | |||
) |
Remove audio a codec from a preference list.
Definition at line 1032 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_codec_pref_append(), and ast_parse_allow_disallow().
01033 { 01034 struct ast_codec_pref oldorder; 01035 int x, y = 0; 01036 int slot; 01037 int size; 01038 01039 if (!pref->order[0]) 01040 return; 01041 01042 memcpy(&oldorder, pref, sizeof(oldorder)); 01043 memset(pref, 0, sizeof(*pref)); 01044 01045 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01046 slot = oldorder.order[x]; 01047 size = oldorder.framing[x]; 01048 if (! slot) 01049 break; 01050 if (AST_FORMAT_LIST[slot-1].bits != format) { 01051 pref->order[y] = slot; 01052 pref->framing[y++] = size; 01053 } 01054 } 01055 01056 }
int ast_codec_pref_setsize | ( | struct ast_codec_pref * | pref, | |
int | format, | |||
int | framems | |||
) |
Set packet size for codec.
Definition at line 1122 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_format_list::def_ms, ast_codec_pref::framing, ast_format_list::inc_ms, ast_format_list::max_ms, ast_format_list::min_ms, and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow(), and process_sdp_a_audio().
01123 { 01124 int x, idx = -1; 01125 01126 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01127 if (AST_FORMAT_LIST[x].bits == format) { 01128 idx = x; 01129 break; 01130 } 01131 } 01132 01133 if (idx < 0) 01134 return -1; 01135 01136 /* size validation */ 01137 if (!framems) 01138 framems = AST_FORMAT_LIST[idx].def_ms; 01139 01140 if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */ 01141 framems -= framems % AST_FORMAT_LIST[idx].inc_ms; 01142 01143 if (framems < AST_FORMAT_LIST[idx].min_ms) 01144 framems = AST_FORMAT_LIST[idx].min_ms; 01145 01146 if (framems > AST_FORMAT_LIST[idx].max_ms) 01147 framems = AST_FORMAT_LIST[idx].max_ms; 01148 01149 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01150 if (pref->order[x] == (idx + 1)) { 01151 pref->framing[x] = framems; 01152 break; 01153 } 01154 } 01155 01156 return x; 01157 }
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 985 of file frame.c.
References ast_codec_pref_index(), and ast_getformatname().
Referenced by dump_prefs(), and socket_process().
00986 { 00987 int x, codec; 00988 size_t total_len, slen; 00989 char *formatname; 00990 00991 memset(buf,0,size); 00992 total_len = size; 00993 buf[0] = '('; 00994 total_len--; 00995 for(x = 0; x < 32 ; x++) { 00996 if (total_len <= 0) 00997 break; 00998 if (!(codec = ast_codec_pref_index(pref,x))) 00999 break; 01000 if ((formatname = ast_getformatname(codec))) { 01001 slen = strlen(formatname); 01002 if (slen > total_len) 01003 break; 01004 strncat(buf, formatname, total_len - 1); /* safe */ 01005 total_len -= slen; 01006 } 01007 if (total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) { 01008 strncat(buf, "|", total_len - 1); /* safe */ 01009 total_len--; 01010 } 01011 } 01012 if (total_len) { 01013 strncat(buf, ")", total_len - 1); /* safe */ 01014 total_len--; 01015 } 01016 01017 return size - total_len; 01018 }
static force_inline int ast_format_rate | ( | int | format | ) | [static] |
Get the sample rate for a given format.
Definition at line 672 of file frame.h.
References AST_FORMAT_G722, and AST_FORMAT_SLINEAR16.
Referenced by __ast_read(), __get_from_jb(), ast_read_generator_actions(), ast_readaudio_callback(), ast_readvideo_callback(), ast_rtp_read(), ast_smoother_read(), ast_translate(), ast_write(), calc_cost(), calc_timestamp(), generator_force(), rtp_get_rate(), and schedule_delivery().
00673 { 00674 if (format == AST_FORMAT_G722 || format == AST_FORMAT_SLINEAR16) 00675 return 16000; 00676 00677 return 8000; 00678 }
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 1502 of file frame.c.
References AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_slinear_saturated_divide(), ast_slinear_saturated_multiply(), and f.
Referenced by audiohook_read_frame_single(), audiohook_volume_callback(), conf_run(), and volume_callback().
01503 { 01504 int count; 01505 short *fdata = f->data.ptr; 01506 short adjust_value = abs(adjustment); 01507 01508 if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR)) 01509 return -1; 01510 01511 if (!adjustment) 01512 return 0; 01513 01514 for (count = 0; count < f->samples; count++) { 01515 if (adjustment > 0) { 01516 ast_slinear_saturated_multiply(&fdata[count], &adjust_value); 01517 } else if (adjustment < 0) { 01518 ast_slinear_saturated_divide(&fdata[count], &adjust_value); 01519 } 01520 } 01521 01522 return 0; 01523 }
void ast_frame_dump | ( | const char * | name, | |
struct ast_frame * | f, | |||
char * | prefix | |||
) |
Dump a frame for debugging purposes
Definition at line 744 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, f, ast_control_t38_parameters::request_response, and term_color().
Referenced by __ast_read(), and ast_write().
00745 { 00746 const char noname[] = "unknown"; 00747 char ftype[40] = "Unknown Frametype"; 00748 char cft[80]; 00749 char subclass[40] = "Unknown Subclass"; 00750 char csub[80]; 00751 char moreinfo[40] = ""; 00752 char cn[60]; 00753 char cp[40]; 00754 char cmn[40]; 00755 const char *message = "Unknown"; 00756 00757 if (!name) 00758 name = noname; 00759 00760 00761 if (!f) { 00762 ast_verbose("%s [ %s (NULL) ] [%s]\n", 00763 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00764 term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00765 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00766 return; 00767 } 00768 /* XXX We should probably print one each of voice and video when the format changes XXX */ 00769 if (f->frametype == AST_FRAME_VOICE) 00770 return; 00771 if (f->frametype == AST_FRAME_VIDEO) 00772 return; 00773 switch(f->frametype) { 00774 case AST_FRAME_DTMF_BEGIN: 00775 strcpy(ftype, "DTMF Begin"); 00776 subclass[0] = f->subclass; 00777 subclass[1] = '\0'; 00778 break; 00779 case AST_FRAME_DTMF_END: 00780 strcpy(ftype, "DTMF End"); 00781 subclass[0] = f->subclass; 00782 subclass[1] = '\0'; 00783 break; 00784 case AST_FRAME_CONTROL: 00785 strcpy(ftype, "Control"); 00786 switch(f->subclass) { 00787 case AST_CONTROL_HANGUP: 00788 strcpy(subclass, "Hangup"); 00789 break; 00790 case AST_CONTROL_RING: 00791 strcpy(subclass, "Ring"); 00792 break; 00793 case AST_CONTROL_RINGING: 00794 strcpy(subclass, "Ringing"); 00795 break; 00796 case AST_CONTROL_ANSWER: 00797 strcpy(subclass, "Answer"); 00798 break; 00799 case AST_CONTROL_BUSY: 00800 strcpy(subclass, "Busy"); 00801 break; 00802 case AST_CONTROL_TAKEOFFHOOK: 00803 strcpy(subclass, "Take Off Hook"); 00804 break; 00805 case AST_CONTROL_OFFHOOK: 00806 strcpy(subclass, "Line Off Hook"); 00807 break; 00808 case AST_CONTROL_CONGESTION: 00809 strcpy(subclass, "Congestion"); 00810 break; 00811 case AST_CONTROL_FLASH: 00812 strcpy(subclass, "Flash"); 00813 break; 00814 case AST_CONTROL_WINK: 00815 strcpy(subclass, "Wink"); 00816 break; 00817 case AST_CONTROL_OPTION: 00818 strcpy(subclass, "Option"); 00819 break; 00820 case AST_CONTROL_RADIO_KEY: 00821 strcpy(subclass, "Key Radio"); 00822 break; 00823 case AST_CONTROL_RADIO_UNKEY: 00824 strcpy(subclass, "Unkey Radio"); 00825 break; 00826 case AST_CONTROL_HOLD: 00827 strcpy(subclass, "Hold"); 00828 break; 00829 case AST_CONTROL_UNHOLD: 00830 strcpy(subclass, "Unhold"); 00831 break; 00832 case AST_CONTROL_T38_PARAMETERS: 00833 if (f->datalen != sizeof(struct ast_control_t38_parameters)) { 00834 message = "Invalid"; 00835 } else { 00836 struct ast_control_t38_parameters *parameters = f->data.ptr; 00837 enum ast_control_t38 state = parameters->request_response; 00838 if (state == AST_T38_REQUEST_NEGOTIATE) 00839 message = "Negotiation Requested"; 00840 else if (state == AST_T38_REQUEST_TERMINATE) 00841 message = "Negotiation Request Terminated"; 00842 else if (state == AST_T38_NEGOTIATED) 00843 message = "Negotiated"; 00844 else if (state == AST_T38_TERMINATED) 00845 message = "Terminated"; 00846 else if (state == AST_T38_REFUSED) 00847 message = "Refused"; 00848 } 00849 snprintf(subclass, sizeof(subclass), "T38_Parameters/%s", message); 00850 break; 00851 case -1: 00852 strcpy(subclass, "Stop generators"); 00853 break; 00854 default: 00855 snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass); 00856 } 00857 break; 00858 case AST_FRAME_NULL: 00859 strcpy(ftype, "Null Frame"); 00860 strcpy(subclass, "N/A"); 00861 break; 00862 case AST_FRAME_IAX: 00863 /* Should never happen */ 00864 strcpy(ftype, "IAX Specific"); 00865 snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass); 00866 break; 00867 case AST_FRAME_TEXT: 00868 strcpy(ftype, "Text"); 00869 strcpy(subclass, "N/A"); 00870 ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); 00871 break; 00872 case AST_FRAME_IMAGE: 00873 strcpy(ftype, "Image"); 00874 snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass)); 00875 break; 00876 case AST_FRAME_HTML: 00877 strcpy(ftype, "HTML"); 00878 switch(f->subclass) { 00879 case AST_HTML_URL: 00880 strcpy(subclass, "URL"); 00881 ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); 00882 break; 00883 case AST_HTML_DATA: 00884 strcpy(subclass, "Data"); 00885 break; 00886 case AST_HTML_BEGIN: 00887 strcpy(subclass, "Begin"); 00888 break; 00889 case AST_HTML_END: 00890 strcpy(subclass, "End"); 00891 break; 00892 case AST_HTML_LDCOMPLETE: 00893 strcpy(subclass, "Load Complete"); 00894 break; 00895 case AST_HTML_NOSUPPORT: 00896 strcpy(subclass, "No Support"); 00897 break; 00898 case AST_HTML_LINKURL: 00899 strcpy(subclass, "Link URL"); 00900 ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); 00901 break; 00902 case AST_HTML_UNLINK: 00903 strcpy(subclass, "Unlink"); 00904 break; 00905 case AST_HTML_LINKREJECT: 00906 strcpy(subclass, "Link Reject"); 00907 break; 00908 default: 00909 snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass); 00910 break; 00911 } 00912 break; 00913 case AST_FRAME_MODEM: 00914 strcpy(ftype, "Modem"); 00915 switch (f->subclass) { 00916 case AST_MODEM_T38: 00917 strcpy(subclass, "T.38"); 00918 break; 00919 case AST_MODEM_V150: 00920 strcpy(subclass, "V.150"); 00921 break; 00922 default: 00923 snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass); 00924 break; 00925 } 00926 break; 00927 default: 00928 snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype); 00929 } 00930 if (!ast_strlen_zero(moreinfo)) 00931 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n", 00932 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00933 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00934 f->frametype, 00935 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00936 f->subclass, 00937 term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)), 00938 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00939 else 00940 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n", 00941 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00942 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00943 f->frametype, 00944 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00945 f->subclass, 00946 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00947 }
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 Frees a frame or list of frames.
fr | Frame to free, or head of list to free | |
cache | Whether to consider this frame for frame caching |
Definition at line 365 of file frame.c.
References __frame_free(), AST_LIST_NEXT, 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.
f1 | The first frame (which will contain the result) | |
f2 | The second frame |
Definition at line 1525 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.
01526 { 01527 int count; 01528 short *data1, *data2; 01529 01530 if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR)) 01531 return -1; 01532 01533 if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR)) 01534 return -1; 01535 01536 if (f1->samples != f2->samples) 01537 return -1; 01538 01539 for (count = 0, data1 = f1->data.ptr, data2 = f2->data.ptr; 01540 count < f1->samples; 01541 count++, data1++, data2++) 01542 ast_slinear_saturated_add(data1, data2); 01543 01544 return 0; 01545 }
Copies a frame.
fr | frame to copy Duplicates a frame -- should only rarely be used, typically frisolate is good enough |
Definition at line 464 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, f, frame_cache, frames, ast_frame::frametype, ast_frame::len, len(), ast_frame::mallocd, ast_frame::mallocd_hdr_len, ast_frame::offset, ast_frame::ptr, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, ast_frame::ts, and ast_frame::uint32.
Referenced by __ast_queue_frame(), ast_frisolate(), ast_jb_put(), ast_rtp_write(), ast_slinfactory_feed(), audiohook_read_frame_single(), autoservice_run(), process_rfc2833(), recordthread(), and rpt().
00465 { 00466 struct ast_frame *out = NULL; 00467 int len, srclen = 0; 00468 void *buf = NULL; 00469 00470 #if !defined(LOW_MEMORY) 00471 struct ast_frame_cache *frames; 00472 #endif 00473 00474 /* Start with standard stuff */ 00475 len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00476 /* If we have a source, add space for it */ 00477 /* 00478 * XXX Watch out here - if we receive a src which is not terminated 00479 * properly, we can be easily attacked. Should limit the size we deal with. 00480 */ 00481 if (f->src) 00482 srclen = strlen(f->src); 00483 if (srclen > 0) 00484 len += srclen + 1; 00485 00486 #if !defined(LOW_MEMORY) 00487 if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) { 00488 AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) { 00489 if (out->mallocd_hdr_len >= len) { 00490 size_t mallocd_len = out->mallocd_hdr_len; 00491 00492 AST_LIST_REMOVE_CURRENT(frame_list); 00493 memset(out, 0, sizeof(*out)); 00494 out->mallocd_hdr_len = mallocd_len; 00495 buf = out; 00496 frames->size--; 00497 break; 00498 } 00499 } 00500 AST_LIST_TRAVERSE_SAFE_END; 00501 } 00502 #endif 00503 00504 if (!buf) { 00505 if (!(buf = ast_calloc_cache(1, len))) 00506 return NULL; 00507 out = buf; 00508 out->mallocd_hdr_len = len; 00509 } 00510 00511 out->frametype = f->frametype; 00512 out->subclass = f->subclass; 00513 out->datalen = f->datalen; 00514 out->samples = f->samples; 00515 out->delivery = f->delivery; 00516 /* Set us as having malloc'd header only, so it will eventually 00517 get freed. */ 00518 out->mallocd = AST_MALLOCD_HDR; 00519 out->offset = AST_FRIENDLY_OFFSET; 00520 if (out->datalen) { 00521 out->data.ptr = buf + sizeof(*out) + AST_FRIENDLY_OFFSET; 00522 memcpy(out->data.ptr, f->data.ptr, out->datalen); 00523 } else { 00524 out->data.uint32 = f->data.uint32; 00525 } 00526 if (srclen > 0) { 00527 /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */ 00528 char *src; 00529 out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00530 src = (char *) out->src; 00531 /* Must have space since we allocated for it */ 00532 strcpy(src, f->src); 00533 } 00534 ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO); 00535 out->ts = f->ts; 00536 out->len = f->len; 00537 out->seqno = f->seqno; 00538 return out; 00539 }
Makes a frame independent of any static storage.
fr | frame to act upon Take a frame, and if it's not been malloc'd, make a malloc'd copy and if the data hasn't been malloced then make the data malloc'd. If you need to store frames, say for queueing, then you should call this function. |
Definition at line 381 of file frame.c.
References ast_copy_flags, ast_frame_header_new(), ast_frdup(), ast_free, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, ast_malloc, AST_MALLOCD_DATA, AST_MALLOCD_HDR, AST_MALLOCD_SRC, ast_strdup, ast_test_flag, ast_frame::data, ast_frame::datalen, ast_frame::frametype, ast_frame::len, ast_frame::mallocd, ast_frame::offset, ast_frame::ptr, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, ast_frame::ts, and ast_frame::uint32.
Referenced by __ast_answer(), ast_rtp_read(), 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 (!fr->datalen) { 00435 out->data.uint32 = fr->data.uint32; 00436 out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC; 00437 return out; 00438 } 00439 if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) { 00440 if (out->src != fr->src) { 00441 ast_free((void *) out->src); 00442 } 00443 if (out != fr) { 00444 ast_free(out); 00445 } 00446 return NULL; 00447 } 00448 newdata += AST_FRIENDLY_OFFSET; 00449 out->offset = AST_FRIENDLY_OFFSET; 00450 out->datalen = fr->datalen; 00451 memcpy(newdata, fr->data.ptr, fr->datalen); 00452 out->data.ptr = newdata; 00453 } else { 00454 out->data = fr->data; 00455 memset(&fr->data, 0, sizeof(fr->data)); 00456 fr->mallocd &= ~AST_MALLOCD_DATA; 00457 } 00458 00459 out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA; 00460 00461 return out; 00462 }
struct ast_format_list* ast_get_format_list | ( | size_t * | size | ) |
Definition at line 557 of file frame.c.
References ARRAY_LEN, and AST_FORMAT_LIST.
00558 { 00559 *size = ARRAY_LEN(AST_FORMAT_LIST); 00560 return AST_FORMAT_LIST; 00561 }
struct ast_format_list* ast_get_format_list_index | ( | int | index | ) |
Definition at line 552 of file frame.c.
References AST_FORMAT_LIST.
00553 { 00554 return &AST_FORMAT_LIST[idx]; 00555 }
int ast_getformatbyname | ( | const char * | name | ) |
Gets a format from a name.
name | string of format |
Definition at line 624 of file frame.c.
References ARRAY_LEN, ast_expand_codec_alias(), AST_FORMAT_LIST, 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().
00625 { 00626 int x, all, format = 0; 00627 00628 all = strcasecmp(name, "all") ? 0 : 1; 00629 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00630 if (all || 00631 !strcasecmp(AST_FORMAT_LIST[x].name,name) || 00632 !strcasecmp(AST_FORMAT_LIST[x].name, ast_expand_codec_alias(name))) { 00633 format |= AST_FORMAT_LIST[x].bits; 00634 if (!all) 00635 break; 00636 } 00637 } 00638 00639 return format; 00640 }
char* ast_getformatname | ( | int | format | ) |
Get the name of a format.
format | id of format |
Definition at line 563 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, 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().
00564 { 00565 int x; 00566 char *ret = "unknown"; 00567 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00568 if (AST_FORMAT_LIST[x].bits == format) { 00569 ret = AST_FORMAT_LIST[x].name; 00570 break; 00571 } 00572 } 00573 return ret; 00574 }
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 576 of file frame.c.
References ARRAY_LEN, ast_copy_string(), AST_FORMAT_LIST, 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(), process_sdp(), serialize_showchan(), set_format(), show_channels_cb(), sip_new(), sip_request_call(), sip_show_channel(), sip_show_settings(), and sip_write().
00577 { 00578 int x; 00579 unsigned len; 00580 char *start, *end = buf; 00581 00582 if (!size) 00583 return buf; 00584 snprintf(end, size, "0x%x (", format); 00585 len = strlen(end); 00586 end += len; 00587 size -= len; 00588 start = end; 00589 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00590 if (AST_FORMAT_LIST[x].bits & format) { 00591 snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name); 00592 len = strlen(end); 00593 end += len; 00594 size -= len; 00595 } 00596 } 00597 if (start == end) 00598 ast_copy_string(start, "nothing)", size); 00599 else if (size > 1) 00600 *(end -1) = ')'; 00601 return buf; 00602 }
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 1221 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().
01222 { 01223 int errors = 0; 01224 char *parse = NULL, *this = NULL, *psize = NULL; 01225 int format = 0, framems = 0; 01226 01227 parse = ast_strdupa(list); 01228 while ((this = strsep(&parse, ","))) { 01229 framems = 0; 01230 if ((psize = strrchr(this, ':'))) { 01231 *psize++ = '\0'; 01232 ast_debug(1, "Packetization for codec: %s is %s\n", this, psize); 01233 framems = atoi(psize); 01234 if (framems < 0) { 01235 framems = 0; 01236 errors++; 01237 ast_log(LOG_WARNING, "Bad packetization value for codec %s\n", this); 01238 } 01239 } 01240 if (!(format = ast_getformatbyname(this))) { 01241 ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this); 01242 errors++; 01243 continue; 01244 } 01245 01246 if (mask) { 01247 if (allowing) 01248 *mask |= format; 01249 else 01250 *mask &= ~format; 01251 } 01252 01253 /* Set up a preference list for audio. Do not include video in preferences 01254 since we can not transcode video and have to use whatever is offered 01255 */ 01256 if (pref && (format & AST_FORMAT_AUDIO_MASK)) { 01257 if (strcasecmp(this, "all")) { 01258 if (allowing) { 01259 ast_codec_pref_append(pref, format); 01260 ast_codec_pref_setsize(pref, format, framems); 01261 } 01262 else 01263 ast_codec_pref_remove(pref, format); 01264 } else if (!allowing) { 01265 memset(pref, 0, sizeof(*pref)); 01266 } 01267 } 01268 } 01269 return errors; 01270 }
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 | ) |
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 | ) |
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(), len(), LOG_WARNING, and s.
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 s, 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 s.
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 s.
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 s.
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 541 of file frame.c.
Referenced by __ast_smoother_feed(), iax_frame_wrap(), phone_write_buf(), and smoother_frame_feed().
00542 { 00543 int i; 00544 unsigned short *dst_s = dst; 00545 const unsigned short *src_s = src; 00546 00547 for (i = 0; i < samples; i++) 00548 dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8); 00549 }
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(), create_dtmf_frame(), 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_sdp(), sip_read(), sip_rtp_read(), skinny_rtp_read(), unistim_rtp_read(), and wakeup_sub().