Asterisk internal frame definitions. More...
#include <sys/time.h>
#include "asterisk/frame_defs.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_read_action_payload |
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... | |
union | ast_frame_subclass |
struct | ast_option_header |
struct | oprmode |
Defines | |
#define | AST_FORMAT_ADPCM (1ULL << 5) |
#define | AST_FORMAT_ALAW (1ULL << 3) |
#define | AST_FORMAT_AUDIO_MASK 0xFFFF0000FFFFULL |
#define | AST_FORMAT_FIRST_VIDEO_BIT AST_FORMAT_H261 |
#define | AST_FORMAT_G719 (1ULL << 32) |
#define | AST_FORMAT_G722 (1ULL << 12) |
#define | AST_FORMAT_G723_1 (1ULL << 0) |
#define | AST_FORMAT_G726 (1ULL << 11) |
#define | AST_FORMAT_G726_AAL2 (1ULL << 4) |
#define | AST_FORMAT_G729A (1ULL << 8) |
#define | AST_FORMAT_GSM (1ULL << 1) |
#define | AST_FORMAT_H261 (1ULL << 18) |
#define | AST_FORMAT_H263 (1ULL << 19) |
#define | AST_FORMAT_H263_PLUS (1ULL << 20) |
#define | AST_FORMAT_H264 (1ULL << 21) |
#define | AST_FORMAT_ILBC (1ULL << 10) |
#define | AST_FORMAT_JPEG (1ULL << 16) |
#define | AST_FORMAT_LPC10 (1ULL << 7) |
#define | AST_FORMAT_MAX_TEXT (1ULL << 28) |
#define | AST_FORMAT_MP4_VIDEO (1ULL << 22) |
#define | AST_FORMAT_PNG (1ULL << 17) |
#define | AST_FORMAT_RESERVED (1ULL << 63) |
#define | AST_FORMAT_SIREN14 (1ULL << 14) |
#define | AST_FORMAT_SIREN7 (1ULL << 13) |
#define | AST_FORMAT_SLINEAR (1ULL << 6) |
#define | AST_FORMAT_SLINEAR16 (1ULL << 15) |
#define | AST_FORMAT_SPEEX (1ULL << 9) |
#define | AST_FORMAT_SPEEX16 (1ULL << 33) |
#define | AST_FORMAT_T140 (1ULL << 27) |
#define | AST_FORMAT_T140RED (1ULL << 26) |
#define | AST_FORMAT_TESTLAW (1ULL << 47) |
#define | AST_FORMAT_TEXT_MASK (((1ULL << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK)) |
#define | AST_FORMAT_ULAW (1ULL << 2) |
#define | AST_FORMAT_VIDEO_MASK ((((1ULL << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) | 0x7FFF000000000000ULL) |
#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_CC_AGENT_TYPE 17 |
#define | AST_OPTION_CHANNEL_WRITE 9 |
Handle channel write data If a channel needs to process the data from a func_channel write operation after func_channel_write executes, it can define the setoption callback and process this option. A pointer to an ast_chan_write_info_t will be passed. | |
#define | AST_OPTION_DEVICE_NAME 16 |
#define | AST_OPTION_DIGIT_DETECT 14 |
#define | AST_OPTION_ECHOCAN 8 |
#define | AST_OPTION_FAX_DETECT 15 |
#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_FORMAT_READ 11 |
#define | AST_OPTION_FORMAT_WRITE 12 |
#define | AST_OPTION_MAKE_COMPATIBLE 13 |
#define | AST_OPTION_OPRMODE 7 |
#define | AST_OPTION_RELAXDTMF 3 |
#define | AST_OPTION_RXGAIN 6 |
#define | AST_OPTION_SECURE_MEDIA 19 |
#define | AST_OPTION_SECURE_SIGNALING 18 |
#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_TRANSFER = 21, AST_CONTROL_CONNECTED_LINE = 22, AST_CONTROL_REDIRECTING = 23, AST_CONTROL_T38_PARAMETERS = 24, AST_CONTROL_CC = 25, AST_CONTROL_SRCCHANGE = 26, AST_CONTROL_READ_ACTION = 27, AST_CONTROL_AOC = 28, AST_CONTROL_END_OF_Q = 29, AST_CONTROL_INCOMPLETE = 30, AST_CONTROL_UPDATE_RTP_PEER = 31 } |
enum | ast_control_t38 { AST_T38_REQUEST_NEGOTIATE = 1, AST_T38_REQUEST_TERMINATE, AST_T38_NEGOTIATED, AST_T38_TERMINATED, AST_T38_REFUSED, AST_T38_REQUEST_PARMS } |
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_control_transfer { AST_TRANSFER_SUCCESS = 0, AST_TRANSFER_FAILED } |
enum | ast_frame_read_action { AST_FRAME_READ_ACTION_CONNECTED_LINE_MACRO } |
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 (format_t codec) |
Get a name from a format Gets a name from a format. | |
format_t | ast_codec_choose (struct ast_codec_pref *pref, format_t 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 (format_t 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 (format_t format) |
Gets duration in ms of interpolation frame for a format. | |
int | ast_codec_pref_append (struct ast_codec_pref *pref, format_t 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, format_t format) |
Get packet size for codec. | |
format_t | 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, format_t 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, format_t format) |
Remove audio a codec from a preference list. | |
int | ast_codec_pref_setsize (struct ast_codec_pref *pref, format_t 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 (format_t 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. | |
int | ast_frame_clear (struct ast_frame *frame) |
Clear all audio samples from an ast_frame. The frame must be AST_FRAME_VOICE and AST_FORMAT_SLINEAR. | |
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) |
format_t | ast_getformatbyname (const char *name) |
Gets a format from a name. | |
char * | ast_getformatname (format_t format) |
Get the name of a format. | |
char * | ast_getformatname_multiple (char *buf, size_t size, format_t format) |
Get the names of a set of formats. | |
int | ast_parse_allow_disallow (struct ast_codec_pref *pref, format_t *mask, const char *list, int allowing) |
Parse an "allow" or "deny" line in a channel or device configuration and update the capabilities mask and pref if provided. Video codecs are not added to codec preference lists, since we can not transcode. | |
void | ast_swapcopy_samples (void *dst, const void *src, int samples) |
Variables | |
struct ast_frame | ast_null_frame |
AST_Smoother | |
| |
#define | ast_smoother_feed(s, f) __ast_smoother_feed(s, f, 0) |
#define | ast_smoother_feed_be(s, f) __ast_smoother_feed(s, f, 0) |
#define | ast_smoother_feed_le(s, f) __ast_smoother_feed(s, f, 1) |
int | __ast_smoother_feed (struct ast_smoother *s, struct ast_frame *f, int swap) |
void | ast_smoother_free (struct ast_smoother *s) |
int | ast_smoother_get_flags (struct ast_smoother *smoother) |
struct ast_smoother * | ast_smoother_new (int bytes) |
struct ast_frame * | ast_smoother_read (struct ast_smoother *s) |
void | ast_smoother_reconfigure (struct ast_smoother *s, int bytes) |
Reconfigure an existing smoother to output a different number of bytes per frame. | |
void | ast_smoother_reset (struct ast_smoother *s, int bytes) |
void | ast_smoother_set_flags (struct ast_smoother *smoother, int flags) |
int | ast_smoother_test_flag (struct ast_smoother *s, int flag) |
Asterisk internal frame definitions.
Definition in file frame.h.
#define AST_FORMAT_ADPCM (1ULL << 5) |
ADPCM (IMA)
Definition at line 252 of file frame.h.
Referenced by adpcm_sample(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), vox_read(), and vox_write().
#define AST_FORMAT_ALAW (1ULL << 3) |
Raw A-law data (G.711)
Definition at line 248 of file frame.h.
Referenced by alaw_sample(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), cb_events(), codec_ast2skinny(), codec_skinny2ast(), 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 0xFFFF0000FFFFULL |
Maximum audio mask
Definition at line 274 of file frame.h.
Referenced by add_sdp(), ast_best_codec(), ast_channel_make_compatible_helper(), ast_codec_choose(), ast_filehelper(), ast_openstream_full(), ast_parse_allow_disallow(), ast_playstream(), ast_request(), ast_rtp_read(), ast_translate_available_formats(), ast_translator_best_choice(), ast_write(), ast_writestream(), begin_dial_channel(), complete_trans_path_choice(), filestream_close(), func_channel_read(), generator_force(), gtalk_rtp_read(), handle_cli_core_show_translation(), jingle_rtp_read(), oh323_request(), phone_read(), process_sdp(), set_format(), show_codecs(), sip_call(), sip_request_call(), sip_rtp_read(), sip_write(), skinny_request(), transmit_connect(), transmit_connect_with_sdp(), transmit_modify_request(), and transmit_modify_with_sdp().
#define AST_FORMAT_FIRST_VIDEO_BIT AST_FORMAT_H261 |
Definition at line 281 of file frame.h.
Referenced by ast_openvstream().
#define AST_FORMAT_G719 (1ULL << 32) |
G.719 (64 kbps assumed)
Definition at line 299 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_write(), g719read(), g719write(), and process_sdp_a_audio().
#define AST_FORMAT_G722 (1ULL << 12) |
G.722
Definition at line 266 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_raw_write(), au_seek(), g722_sample(), pcm_read(), and rtp_get_rate().
#define AST_FORMAT_G723_1 (1ULL << 0) |
G.723.1 compression
Definition at line 242 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(), dahdi_destroy(), dahdi_translate(), g723_read(), g723_write(), load_module(), phone_request(), phone_setup(), phone_write(), and start_rtp().
#define AST_FORMAT_G726 (1ULL << 11) |
ADPCM (G.726, 32kbps, RFC3551 codeword packing)
Definition at line 264 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_codecs_payloads_set_rtpmap_type_rate(), g726_read(), g726_sample(), and g726_write().
#define AST_FORMAT_G726_AAL2 (1ULL << 4) |
ADPCM (G.726, 32kbps, AAL2 codeword packing)
Definition at line 250 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_codecs_payloads_set_rtpmap_type_rate(), ast_rtp_lookup_mime_subtype2(), codec_ast2skinny(), codec_skinny2ast(), and setup_rtp_connection().
#define AST_FORMAT_G729A (1ULL << 8) |
G.729A audio
Definition at line 258 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(), dahdi_destroy(), dahdi_translate(), g729_read(), g729_write(), load_module(), phone_request(), phone_setup(), phone_write(), and start_rtp().
#define AST_FORMAT_GSM (1ULL << 1) |
GSM compression
Definition at line 244 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), gsm_read(), gsm_sample(), gsm_write(), wav_read(), and wav_write().
#define AST_FORMAT_H261 (1ULL << 18) |
H.261 Video
Definition at line 280 of file frame.h.
Referenced by codec_ast2skinny(), codec_skinny2ast(), and h261_encap().
#define AST_FORMAT_H263 (1ULL << 19) |
H.263 Video
Definition at line 283 of file frame.h.
Referenced by codec_ast2skinny(), codec_skinny2ast(), h263_encap(), h263_read(), and h263_write().
#define AST_FORMAT_H263_PLUS (1ULL << 20) |
#define AST_FORMAT_H264 (1ULL << 21) |
H.264 Video
Definition at line 287 of file frame.h.
Referenced by h264_encap(), h264_read(), and h264_write().
#define AST_FORMAT_ILBC (1ULL << 10) |
iLBC Free Compression
Definition at line 262 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(), ilbc_read(), ilbc_sample(), and ilbc_write().
#define AST_FORMAT_JPEG (1ULL << 16) |
JPEG Images
Definition at line 276 of file frame.h.
Referenced by jpeg_read_image(), and jpeg_write_image().
#define AST_FORMAT_LPC10 (1ULL << 7) |
LPC10, 180 samples/frame
Definition at line 256 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_samples(), and lpc10_sample().
#define AST_FORMAT_MP4_VIDEO (1ULL << 22) |
#define AST_FORMAT_PNG (1ULL << 17) |
#define AST_FORMAT_RESERVED (1ULL << 63) |
#define AST_FORMAT_SIREN14 (1ULL << 14) |
G.722.1 Annex C (also known as Siren14, 48kbps assumed)
Definition at line 270 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_write(), process_sdp_a_audio(), siren14read(), and siren14write().
#define AST_FORMAT_SIREN7 (1ULL << 13) |
G.722.1 (also known as Siren7, 32kbps assumed)
Definition at line 268 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_write(), process_sdp_a_audio(), siren7read(), and siren7write().
#define AST_FORMAT_SLINEAR (1ULL << 6) |
Raw 16-bit Signed Linear (8000 Hz) PCM
Definition at line 254 of file frame.h.
Referenced by __ast_play_and_record(), _moh_class_malloc(), action_originate(), agent_new(), alsa_new(), alsa_read(), alsa_request(), ast_audiohook_read_frame(), ast_best_codec(), ast_channel_make_compatible_helper(), ast_channel_start_silence_generator(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_call_progress(), ast_dsp_noise(), ast_dsp_process(), ast_dsp_silence(), ast_frame_adjust_volume(), ast_frame_slinear_sum(), ast_rtp_read(), ast_slinfactory_init(), ast_slinfactory_init_rate(), ast_speech_new(), ast_write(), audio_audiohook_write_list(), audiohook_read_frame_both(), audiohook_read_frame_single(), background_detect_exec(), bridge_request(), build_conf(), chanspy_exec(), conf_run(), dahdi_read(), dahdi_translate(), dahdi_write(), dahdiscan_exec(), dictate_exec(), do_notify(), do_waiting(), eagi_exec(), extenspy_exec(), fax_generator_generate(), find_transcoders(), generic_fax_exec(), generic_recall(), get_rate_change_result(), handle_jack_audio(), handle_recordfile(), handle_speechcreate(), handle_speechrecognize(), iax_frame_wrap(), ices_exec(), is_encoder(), isAnsweringMachine(), jack_exec(), jack_hook_callback(), linear_alloc(), linear_generator(), load_module(), load_moh_classes(), local_ast_moh_start(), measurenoise(), meetme_menu_admin_extended(), mixmonitor_thread(), mp3_exec(), nbs_request(), nbs_xwrite(), NBScat_exec(), new_outgoing(), ogg_vorbis_read(), ogg_vorbis_write(), oh323_rtp_read(), orig_app(), orig_exten(), originate_exec(), oss_new(), oss_read(), oss_request(), parkandannounce_exec(), phone_new(), phone_read(), phone_request(), phone_setup(), phone_write(), pitchshift_cb(), play_sound_file(), playtones_alloc(), playtones_generator(), record_exec(), send_waveform_to_channel(), silence_generator_generate(), slin8_sample(), slinear_read(), slinear_write(), socket_process(), softmix_bridge_join(), softmix_bridge_write(), spandsp_fax_read(), speech_background(), spy_generate(), tonepair_alloc(), tonepair_generator(), transmit_audio(), wav_read(), and wav_write().
#define AST_FORMAT_SLINEAR16 (1ULL << 15) |
Raw 16-bit Signed Linear (16000 Hz) PCM
Definition at line 272 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_read(), ast_slinfactory_init_rate(), console_new(), get_rate_change_result(), pitchshift_cb(), slin16_sample(), slinear_read(), slinear_write(), softmix_bridge_join(), softmix_bridge_write(), stream_monitor(), wav_open(), wav_read(), wav_rewrite(), and wav_write().
#define AST_FORMAT_SPEEX (1ULL << 9) |
SpeeX Free Compression
Definition at line 260 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_samples(), ast_rtp_write(), and speex_sample().
#define AST_FORMAT_SPEEX16 (1ULL << 33) |
SpeeX Wideband (16kHz) Free Compression
Definition at line 301 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_write(), and speex16_sample().
#define AST_FORMAT_T140 (1ULL << 27) |
T.140 Text format - ITU T.140, RFC 4103
Definition at line 294 of file frame.h.
Referenced by add_tcodec_to_sdp(), ast_rtp_read(), and ast_write().
#define AST_FORMAT_T140RED (1ULL << 26) |
T.140 RED Text format RFC 4103
Definition at line 292 of file frame.h.
Referenced by add_tcodec_to_sdp(), ast_rtp_read(), process_sdp(), and rtp_red_init().
#define AST_FORMAT_TESTLAW (1ULL << 47) |
Raw mu-law data (G.711)
Definition at line 303 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), and ast_dsp_process().
#define AST_FORMAT_TEXT_MASK (((1ULL << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK)) |
Definition at line 297 of file frame.h.
Referenced by add_sdp(), ast_request(), show_codecs(), sip_new(), and sip_rtp_read().
#define AST_FORMAT_ULAW (1ULL << 2) |
Raw mu-law data (G.711)
Definition at line 246 of file frame.h.
Referenced by __adsi_transmit_messages(), adsi_careful_send(), adsi_transmit_message_full(), alarmreceiver_exec(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), calc_energy(), codec_ast2skinny(), codec_skinny2ast(), conf_run(), 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(), and ulaw_sample().
#define AST_FORMAT_VIDEO_MASK ((((1ULL << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) | 0x7FFF000000000000ULL) |
Definition at line 290 of file frame.h.
Referenced by add_sdp(), ast_filehelper(), ast_openvstream(), ast_request(), ast_rtp_read(), ast_translate_available_formats(), dialog_initialize_rtp(), filestream_close(), func_channel_read(), gtalk_new(), gtalk_rtp_read(), jingle_new(), jingle_rtp_read(), show_codecs(), sip_new(), and sip_rtp_read().
#define ast_frame_byteswap_be | ( | fr | ) | do { ; } while(0) |
Definition at line 586 of file frame.h.
Referenced by ast_rtp_read(), and socket_process().
#define ast_frame_byteswap_le | ( | fr | ) | do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data.ptr, __f->data.ptr, __f->samples); } while(0) |
Definition at line 585 of file frame.h.
Referenced by phone_read().
#define AST_FRAME_DTMF AST_FRAME_DTMF_END |
Definition at line 128 of file frame.h.
Referenced by __adsi_transmit_messages(), __analog_ss_thread(), __ast_play_and_record(), action_atxfer(), action_dahdidialoffhook(), agent_ack_sleep(), analog_ss_thread(), ast_audiohook_write_list(), ast_dsp_process(), ast_generic_bridge(), ast_jb_put(), background_detect_exec(), cb_events(), channel_spy(), cli_console_dial(), conf_run(), console_dial(), dahdi_bridge(), dictate_exec(), disa_exec(), do_immediate_setup(), echo_exec(), eivr_comm(), feature_request_and_dial(), gtalk_handle_dtmf(), handle_recordfile(), handle_request(), handle_request_info(), handle_speechrecognize(), iax2_bridge(), jingle_handle_dtmf(), mgcp_rtp_read(), misdn_bridge(), mp3_exec(), NBScat_exec(), oh323_rtp_read(), phone_exception(), process_ast_dsp(), receive_dtmf_digits(), record_exec(), send_waveform_to_channel(), sip_rtp_read(), speech_background(), unistim_do_senddigit(), unistim_senddigit_end(), volume_callback(), wait_for_answer(), and wait_for_winner().
#define AST_FRAME_SET_BUFFER | ( | fr, | |||
_base, | |||||
_ofs, | |||||
_datalen | ) |
{ \ (fr)->data.ptr = (char *)_base + (_ofs); \ (fr)->offset = (_ofs); \ (fr)->datalen = (_datalen); \ }
Set the various field of a frame to point to a buffer. Typically you set the base address of the buffer, the offset as AST_FRIENDLY_OFFSET, and the datalen as the amount of bytes queued. The remaining things (to be done manually) is set the number of samples, which cannot be derived from the datalen unless you know the number of bits per sample.
Definition at line 183 of file frame.h.
Referenced by fax_generator_generate(), g719read(), g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), ilbc_read(), ogg_vorbis_read(), pcm_read(), siren14read(), siren7read(), slinear_read(), spandsp_fax_read(), t38_tx_packet_handler(), vox_read(), and wav_read().
#define ast_frfree | ( | fr | ) | ast_frame_free(fr, 1) |
Definition at line 553 of file frame.h.
Referenced by __adsi_transmit_messages(), __analog_ss_thread(), __ast_answer(), __ast_play_and_record(), __ast_queue_frame(), __ast_read(), __ast_request_and_dial(), adsi_careful_send(), agent_ack_sleep(), agent_read(), analog_ss_thread(), ast_audiohook_read_frame(), ast_autoservice_stop(), ast_bridge_call(), ast_bridge_handle_trip(), ast_channel_clear_softhangup(), ast_channel_destructor(), ast_dsp_process(), ast_framehook_attach(), ast_generic_bridge(), ast_indicate_data(), ast_jb_destroy(), ast_jb_put(), ast_queue_cc_frame(), 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_transfer(), ast_translate(), ast_udptl_bridge(), ast_waitfordigit_full(), ast_write(), ast_writestream(), async_agi_read_frame(), async_wait(), audio_audiohook_write_list(), autoservice_run(), background_detect_exec(), bridge_handle_dtmf(), calc_cost(), channel_spy(), check_bridge(), conf_flush(), conf_free(), conf_run(), create_jb(), dahdi_bridge(), dahdi_read(), dial_exec_full(), dictate_exec(), disa_exec(), disable_t38(), do_waiting(), echo_exec(), eivr_comm(), feature_request_and_dial(), find_cache(), framehook_detach_and_destroy(), gen_generate(), generic_fax_exec(), handle_cli_file_convert(), handle_recordfile(), handle_speechrecognize(), iax2_bridge(), ices_exec(), isAnsweringMachine(), jack_exec(), jb_empty_and_reset_adaptive(), jb_empty_and_reset_fixed(), jb_get_and_deliver(), local_bridge_loop(), manage_parked_call(), measurenoise(), moh_files_generator(), monitor_dial(), mp3_exec(), multicast_rtp_write(), NBScat_exec(), read_frame(), receive_dtmf_digits(), receivefax_t38_init(), record_exec(), recordthread(), remote_bridge_loop(), run_agi(), send_tone_burst(), send_waveform_to_channel(), sendfax_t38_init(), sendurl_exec(), session_destroy(), sip_read(), sip_rtp_read(), speech_background(), spy_generate(), 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 204 of file frame.h.
Referenced by __get_from_jb(), adjust_frame_for_plc(), 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(), g719read(), 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_cn_rfc3389(), send_tone_burst(), send_waveform_to_channel(), siren14read(), siren7read(), slinear_read(), sms_generate(), spandsp_fax_read(), tonepair_generator(), 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 230 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 232 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 222 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 210 of file frame.h.
Referenced by __frame_free(), ast_cc_build_frame(), ast_frisolate(), and create_video_frame().
#define AST_MALLOCD_HDR (1 << 0) |
Need the header be free'd?
Definition at line 208 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 212 of file frame.h.
Referenced by __frame_free(), ast_frisolate(), and speex_callback().
#define AST_MIN_OFFSET 32 |
Definition at line 205 of file frame.h.
Referenced by __ast_smoother_feed().
#define AST_MODEM_T38 1 |
T.38 Fax-over-IP
Definition at line 216 of file frame.h.
Referenced by ast_frame_dump(), ast_udptl_write(), generic_fax_exec(), 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 Option data is a single signed char value 0 or 1
Definition at line 423 of file frame.h.
Referenced by ast_bridge_call(), dahdi_hangup(), dahdi_setoption(), and iax2_setoption().
#define AST_OPTION_CC_AGENT_TYPE 17 |
Get the CC agent type from the channel (Read only) Option data is a character buffer of suitable length
Definition at line 490 of file frame.h.
Referenced by ast_channel_get_cc_agent_type(), and dahdi_queryoption().
#define AST_OPTION_CHANNEL_WRITE 9 |
Handle channel write data If a channel needs to process the data from a func_channel write operation after func_channel_write executes, it can define the setoption callback and process this option. A pointer to an ast_chan_write_info_t will be passed.
Definition at line 454 of file frame.h.
Referenced by func_channel_write(), and local_setoption().
#define AST_OPTION_DEVICE_NAME 16 |
Get the device name from the channel (Read only) Option data is a character buffer of suitable length
Definition at line 486 of file frame.h.
Referenced by ast_channel_get_device_name(), and sip_queryoption().
#define AST_OPTION_DIGIT_DETECT 14 |
Get or set the digit detection state of the channel Option data is a single signed char value 0 or 1
Definition at line 478 of file frame.h.
Referenced by ast_bridge_call(), dahdi_queryoption(), dahdi_setoption(), iax2_setoption(), rcvfax_exec(), sip_queryoption(), sip_setoption(), and sndfax_exec().
#define AST_OPTION_ECHOCAN 8 |
Explicitly enable or disable echo cancelation for the given channel Option data is a single signed char value 0 or 1
Definition at line 446 of file frame.h.
Referenced by dahdi_setoption().
#define AST_OPTION_FAX_DETECT 15 |
Get or set the fax tone detection state of the channel Option data is a single signed char value 0 or 1
Definition at line 482 of file frame.h.
Referenced by ast_bridge_call(), dahdi_queryoption(), dahdi_setoption(), iax2_setoption(), rcvfax_exec(), and sndfax_exec().
#define AST_OPTION_FLAG_REQUEST 0 |
Definition at line 401 of file frame.h.
Referenced by ast_bridge_call(), and iax2_setoption().
#define AST_OPTION_FORMAT_READ 11 |
Request that the channel driver deliver frames in a specific format Option data is a format_t
Definition at line 464 of file frame.h.
Referenced by set_format(), and sip_setoption().
#define AST_OPTION_FORMAT_WRITE 12 |
Request that the channel driver be prepared to accept frames in a specific format Option data is a format_t
Definition at line 468 of file frame.h.
Referenced by set_format(), and sip_setoption().
#define AST_OPTION_MAKE_COMPATIBLE 13 |
Request that the channel driver make two channels of the same tech type compatible if possible Option data is an ast_channel
Definition at line 474 of file frame.h.
Referenced by ast_channel_make_compatible_helper(), and sip_setoption().
#define AST_OPTION_OPRMODE 7 |
Definition at line 439 of file frame.h.
Referenced by dahdi_setoption(), dial_exec_full(), and iax2_setoption().
#define AST_OPTION_RELAXDTMF 3 |
Relax the parameters for DTMF reception (mainly for radio use) Option data is a single signed char value 0 or 1
Definition at line 419 of file frame.h.
Referenced by ast_bridge_call(), dahdi_setoption(), and iax2_setoption().
#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 433 of file frame.h.
Referenced by dahdi_setoption(), func_channel_write_real(), iax2_setoption(), play_record_review(), reset_volumes(), set_talk_volume(), and vm_forwardoptions().
#define AST_OPTION_SECURE_MEDIA 19 |
Definition at line 495 of file frame.h.
Referenced by iax2_queryoption(), iax2_setoption(), set_security_requirements(), sip_queryoption(), and sip_setoption().
#define AST_OPTION_SECURE_SIGNALING 18 |
Get or set the security options on a channel Option data is an integer value of 0 or 1
Definition at line 494 of file frame.h.
Referenced by iax2_queryoption(), iax2_setoption(), set_security_requirements(), sip_queryoption(), and sip_setoption().
#define AST_OPTION_T38_STATE 10 |
Definition at line 460 of file frame.h.
Referenced by ast_channel_get_t38_state(), local_queryoption(), and sip_queryoption().
#define AST_OPTION_TDD 2 |
Put a compatible channel into TDD (TTY for the hearing-impared) mode Option data is a single signed char value 0 or 1
Definition at line 415 of file frame.h.
Referenced by analog_hangup(), ast_bridge_call(), dahdi_hangup(), dahdi_setoption(), handle_tddmode(), and iax2_setoption().
#define AST_OPTION_TONE_VERIFY 1 |
Verify touchtones by muting audio transmission (and reception) and verify the tone is still present Option data is a single signed char value 0 or 1
Definition at line 411 of file frame.h.
Referenced by analog_hangup(), ast_bridge_call(), conf_run(), dahdi_hangup(), dahdi_setoption(), iax2_setoption(), 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 428 of file frame.h.
Referenced by common_exec(), dahdi_setoption(), func_channel_write_real(), iax2_setoption(), reset_volumes(), and set_listen_volume().
Definition at line 656 of file frame.h.
Referenced by ast_rtp_write(), and generic_fax_exec().
Definition at line 661 of file frame.h.
Referenced by ast_rtp_write().
#define AST_SMOOTHER_FLAG_BE (1 << 1) |
Definition at line 398 of file frame.h.
Referenced by ast_rtp_write().
#define AST_SMOOTHER_FLAG_G729 (1 << 0) |
Definition at line 397 of file frame.h.
Referenced by __ast_smoother_feed(), ast_smoother_read(), and smoother_frame_feed().
anonymous enum |
Definition at line 130 of file frame.h.
00130 { 00131 /*! This frame contains valid timing information */ 00132 AST_FRFLAG_HAS_TIMING_INFO = (1 << 0), 00133 };
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_TRANSFER |
Indicate status of a transfer request |
AST_CONTROL_CONNECTED_LINE |
Indicate connected line has changed |
AST_CONTROL_REDIRECTING |
Indicate redirecting id has changed |
AST_CONTROL_T38_PARAMETERS |
T38 state change request/notification with parameters |
AST_CONTROL_CC |
Indication that Call completion service is possible |
AST_CONTROL_SRCCHANGE |
Media source has changed and requires a new RTP SSRC |
AST_CONTROL_READ_ACTION |
Tell ast_read to take a specific action |
AST_CONTROL_AOC |
Advice of Charge with encoded generic AOC payload |
AST_CONTROL_END_OF_Q |
Indicate that this position was the end of the channel queue for a softhangup. |
AST_CONTROL_INCOMPLETE |
Indication that the extension dialed is incomplete |
AST_CONTROL_UPDATE_RTP_PEER |
Interrupt the bridge and have it update the peer |
Definition at line 307 of file frame.h.
00307 { 00308 AST_CONTROL_HANGUP = 1, /*!< Other end has hungup */ 00309 AST_CONTROL_RING = 2, /*!< Local ring */ 00310 AST_CONTROL_RINGING = 3, /*!< Remote end is ringing */ 00311 AST_CONTROL_ANSWER = 4, /*!< Remote end has answered */ 00312 AST_CONTROL_BUSY = 5, /*!< Remote end is busy */ 00313 AST_CONTROL_TAKEOFFHOOK = 6, /*!< Make it go off hook */ 00314 AST_CONTROL_OFFHOOK = 7, /*!< Line is off hook */ 00315 AST_CONTROL_CONGESTION = 8, /*!< Congestion (circuits busy) */ 00316 AST_CONTROL_FLASH = 9, /*!< Flash hook */ 00317 AST_CONTROL_WINK = 10, /*!< Wink */ 00318 AST_CONTROL_OPTION = 11, /*!< Set a low-level option */ 00319 AST_CONTROL_RADIO_KEY = 12, /*!< Key Radio */ 00320 AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */ 00321 AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */ 00322 AST_CONTROL_PROCEEDING = 15, /*!< Indicate CALL PROCEEDING */ 00323 AST_CONTROL_HOLD = 16, /*!< Indicate call is placed on hold */ 00324 AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold */ 00325 AST_CONTROL_VIDUPDATE = 18, /*!< Indicate video frame update */ 00326 _XXX_AST_CONTROL_T38 = 19, /*!< T38 state change request/notification \deprecated This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead. */ 00327 AST_CONTROL_SRCUPDATE = 20, /*!< Indicate source of media has changed */ 00328 AST_CONTROL_TRANSFER = 21, /*!< Indicate status of a transfer request */ 00329 AST_CONTROL_CONNECTED_LINE = 22,/*!< Indicate connected line has changed */ 00330 AST_CONTROL_REDIRECTING = 23, /*!< Indicate redirecting id has changed */ 00331 AST_CONTROL_T38_PARAMETERS = 24,/*!< T38 state change request/notification with parameters */ 00332 AST_CONTROL_CC = 25, /*!< Indication that Call completion service is possible */ 00333 AST_CONTROL_SRCCHANGE = 26, /*!< Media source has changed and requires a new RTP SSRC */ 00334 AST_CONTROL_READ_ACTION = 27, /*!< Tell ast_read to take a specific action */ 00335 AST_CONTROL_AOC = 28, /*!< Advice of Charge with encoded generic AOC payload */ 00336 AST_CONTROL_END_OF_Q = 29, /*!< Indicate that this position was the end of the channel queue for a softhangup. */ 00337 AST_CONTROL_INCOMPLETE = 30, /*!< Indication that the extension dialed is incomplete */ 00338 AST_CONTROL_UPDATE_RTP_PEER = 31, /*!< Interrupt the bridge and have it update the peer */ 00339 };
enum ast_control_t38 |
AST_T38_REQUEST_NEGOTIATE |
Request T38 on a channel (voice to fax) |
AST_T38_REQUEST_TERMINATE |
Terminate T38 on a channel (fax to voice) |
AST_T38_NEGOTIATED |
T38 negotiated (fax mode) |
AST_T38_TERMINATED |
T38 terminated (back to voice) |
AST_T38_REFUSED |
T38 refused for some reason (usually rejected by remote end) |
AST_T38_REQUEST_PARMS |
request far end T.38 parameters for a channel in 'negotiating' state |
Definition at line 358 of file frame.h.
00358 { 00359 AST_T38_REQUEST_NEGOTIATE = 1, /*!< Request T38 on a channel (voice to fax) */ 00360 AST_T38_REQUEST_TERMINATE, /*!< Terminate T38 on a channel (fax to voice) */ 00361 AST_T38_NEGOTIATED, /*!< T38 negotiated (fax mode) */ 00362 AST_T38_TERMINATED, /*!< T38 terminated (back to voice) */ 00363 AST_T38_REFUSED, /*!< T38 refused for some reason (usually rejected by remote end) */ 00364 AST_T38_REQUEST_PARMS, /*!< request far end T.38 parameters for a channel in 'negotiating' state */ 00365 };
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 367 of file frame.h.
00367 { 00368 AST_T38_RATE_2400 = 0, 00369 AST_T38_RATE_4800, 00370 AST_T38_RATE_7200, 00371 AST_T38_RATE_9600, 00372 AST_T38_RATE_12000, 00373 AST_T38_RATE_14400, 00374 };
Definition at line 376 of file frame.h.
00376 { 00377 AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF = 0, 00378 AST_T38_RATE_MANAGEMENT_LOCAL_TCF, 00379 };
enum ast_control_transfer |
AST_TRANSFER_SUCCESS |
Transfer request on the channel worked |
AST_TRANSFER_FAILED |
Transfer request on the channel failed |
Definition at line 392 of file frame.h.
00392 { 00393 AST_TRANSFER_SUCCESS = 0, /*!< Transfer request on the channel worked */ 00394 AST_TRANSFER_FAILED, /*!< Transfer request on the channel failed */ 00395 };
Definition at line 341 of file frame.h.
00341 { 00342 AST_FRAME_READ_ACTION_CONNECTED_LINE_MACRO, 00343 };
enum ast_frame_type |
Frame types.
Definition at line 101 of file frame.h.
00101 { 00102 /*! DTMF end event, subclass is the digit */ 00103 AST_FRAME_DTMF_END = 1, 00104 /*! Voice data, subclass is AST_FORMAT_* */ 00105 AST_FRAME_VOICE, 00106 /*! Video frame, maybe?? :) */ 00107 AST_FRAME_VIDEO, 00108 /*! A control frame, subclass is AST_CONTROL_* */ 00109 AST_FRAME_CONTROL, 00110 /*! An empty, useless frame */ 00111 AST_FRAME_NULL, 00112 /*! Inter Asterisk Exchange private frame type */ 00113 AST_FRAME_IAX, 00114 /*! Text messages */ 00115 AST_FRAME_TEXT, 00116 /*! Image Frames */ 00117 AST_FRAME_IMAGE, 00118 /*! HTML Frame */ 00119 AST_FRAME_HTML, 00120 /*! Comfort Noise frame (subclass is level of CNG in -dBov), 00121 body may include zero or more 8-bit quantization coefficients */ 00122 AST_FRAME_CNG, 00123 /*! Modem-over-IP data streams */ 00124 AST_FRAME_MODEM, 00125 /*! DTMF begin event, subclass is the digit */ 00126 AST_FRAME_DTMF_BEGIN, 00127 };
int __ast_smoother_feed | ( | struct ast_smoother * | s, | |
struct ast_frame * | f, | |||
int | swap | |||
) |
Definition at line 208 of file frame.c.
References AST_FRAME_VOICE, ast_getformatname(), ast_log(), AST_MIN_OFFSET, AST_SMOOTHER_FLAG_G729, ast_swapcopy_samples(), ast_frame_subclass::codec, ast_frame::data, ast_frame::datalen, ast_smoother::flags, ast_smoother::format, ast_frame::frametype, ast_smoother::len, LOG_WARNING, ast_frame::offset, ast_smoother::opt, ast_smoother::opt_needs_swap, ast_frame::ptr, ast_frame::samples, ast_smoother::samplesperbyte, ast_smoother::size, smoother_frame_feed(), SMOOTHER_SIZE, and ast_frame::subclass.
00209 { 00210 if (f->frametype != AST_FRAME_VOICE) { 00211 ast_log(LOG_WARNING, "Huh? Can't smooth a non-voice frame!\n"); 00212 return -1; 00213 } 00214 if (!s->format) { 00215 s->format = f->subclass.codec; 00216 s->samplesperbyte = (float)f->samples / (float)f->datalen; 00217 } else if (s->format != f->subclass.codec) { 00218 ast_log(LOG_WARNING, "Smoother was working on %s format frames, now trying to feed %s?\n", 00219 ast_getformatname(s->format), ast_getformatname(f->subclass.codec)); 00220 return -1; 00221 } 00222 if (s->len + f->datalen > SMOOTHER_SIZE) { 00223 ast_log(LOG_WARNING, "Out of smoother space\n"); 00224 return -1; 00225 } 00226 if (((f->datalen == s->size) || 00227 ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) && 00228 !s->opt && 00229 !s->len && 00230 (f->offset >= AST_MIN_OFFSET)) { 00231 /* Optimize by sending the frame we just got 00232 on the next read, thus eliminating the douple 00233 copy */ 00234 if (swap) 00235 ast_swapcopy_samples(f->data.ptr, f->data.ptr, f->samples); 00236 s->opt = f; 00237 s->opt_needs_swap = swap ? 1 : 0; 00238 return 0; 00239 } 00240 00241 return smoother_frame_feed(s, f, swap); 00242 }
char* ast_codec2str | ( | format_t | 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 660 of file frame.c.
References ARRAY_LEN, and ast_format_list::desc.
Referenced by moh_alloc(), show_codec_n(), and show_codecs().
00661 { 00662 int x; 00663 char *ret = "unknown"; 00664 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00665 if (AST_FORMAT_LIST[x].bits == codec) { 00666 ret = AST_FORMAT_LIST[x].desc; 00667 break; 00668 } 00669 } 00670 return ret; 00671 }
format_t ast_codec_choose | ( | struct ast_codec_pref * | pref, | |
format_t | 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 1249 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().
01250 { 01251 int x, slot; 01252 format_t ret = 0; 01253 01254 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01255 slot = pref->order[x]; 01256 01257 if (!slot) 01258 break; 01259 if (formats & AST_FORMAT_LIST[slot-1].bits) { 01260 ret = AST_FORMAT_LIST[slot-1].bits; 01261 break; 01262 } 01263 } 01264 if (ret & AST_FORMAT_AUDIO_MASK) 01265 return ret; 01266 01267 ast_debug(4, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec"); 01268 01269 return find_best ? ast_best_codec(formats) : 0; 01270 }
int ast_codec_get_len | ( | format_t | format, | |
int | samples | |||
) |
Returns the number of bytes for the number of samples of the given format.
Definition at line 1532 of file frame.c.
References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G719, AST_FORMAT_G722, AST_FORMAT_G723_1, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_ILBC, AST_FORMAT_SIREN14, AST_FORMAT_SIREN7, AST_FORMAT_SLINEAR, AST_FORMAT_SLINEAR16, AST_FORMAT_TESTLAW, AST_FORMAT_ULAW, ast_getformatname(), ast_log(), len(), and LOG_WARNING.
Referenced by moh_generate(), and monmp3thread().
01533 { 01534 int len = 0; 01535 01536 /* XXX Still need speex, and lpc10 XXX */ 01537 switch(format) { 01538 case AST_FORMAT_G723_1: 01539 len = (samples / 240) * 20; 01540 break; 01541 case AST_FORMAT_ILBC: 01542 len = (samples / 240) * 50; 01543 break; 01544 case AST_FORMAT_GSM: 01545 len = (samples / 160) * 33; 01546 break; 01547 case AST_FORMAT_G729A: 01548 len = samples / 8; 01549 break; 01550 case AST_FORMAT_SLINEAR: 01551 case AST_FORMAT_SLINEAR16: 01552 len = samples * 2; 01553 break; 01554 case AST_FORMAT_ULAW: 01555 case AST_FORMAT_ALAW: 01556 case AST_FORMAT_TESTLAW: 01557 len = samples; 01558 break; 01559 case AST_FORMAT_G722: 01560 case AST_FORMAT_ADPCM: 01561 case AST_FORMAT_G726: 01562 case AST_FORMAT_G726_AAL2: 01563 len = samples / 2; 01564 break; 01565 case AST_FORMAT_SIREN7: 01566 /* 16,000 samples per second at 32kbps is 4,000 bytes per second */ 01567 len = samples / (16000 / 4000); 01568 break; 01569 case AST_FORMAT_SIREN14: 01570 /* 32,000 samples per second at 48kbps is 6,000 bytes per second */ 01571 len = (int) samples / ((float) 32000 / 6000); 01572 break; 01573 case AST_FORMAT_G719: 01574 /* 48,000 samples per second at 64kbps is 8,000 bytes per second */ 01575 len = (int) samples / ((float) 48000 / 8000); 01576 break; 01577 default: 01578 ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format)); 01579 } 01580 01581 return len; 01582 }
int ast_codec_get_samples | ( | struct ast_frame * | f | ) |
Returns the number of samples contained in the frame.
Definition at line 1470 of file frame.c.
References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G719, AST_FORMAT_G722, AST_FORMAT_G723_1, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_ILBC, AST_FORMAT_LPC10, AST_FORMAT_SIREN14, AST_FORMAT_SIREN7, AST_FORMAT_SLINEAR, AST_FORMAT_SLINEAR16, AST_FORMAT_SPEEX, AST_FORMAT_SPEEX16, AST_FORMAT_TESTLAW, AST_FORMAT_ULAW, ast_getformatname_multiple(), ast_log(), ast_frame_subclass::codec, ast_frame::data, ast_frame::datalen, g723_samples(), LOG_WARNING, ast_frame::ptr, speex_samples(), and ast_frame::subclass.
Referenced by ast_rtp_read(), dahdi_encoder_frameout(), isAnsweringMachine(), moh_generate(), schedule_delivery(), socket_process(), and socket_process_meta().
01471 { 01472 int samples = 0; 01473 char tmp[64]; 01474 01475 switch (f->subclass.codec) { 01476 case AST_FORMAT_SPEEX: 01477 samples = speex_samples(f->data.ptr, f->datalen); 01478 break; 01479 case AST_FORMAT_SPEEX16: 01480 samples = 2 * speex_samples(f->data.ptr, f->datalen); 01481 break; 01482 case AST_FORMAT_G723_1: 01483 samples = g723_samples(f->data.ptr, f->datalen); 01484 break; 01485 case AST_FORMAT_ILBC: 01486 samples = 240 * (f->datalen / 50); 01487 break; 01488 case AST_FORMAT_GSM: 01489 samples = 160 * (f->datalen / 33); 01490 break; 01491 case AST_FORMAT_G729A: 01492 samples = f->datalen * 8; 01493 break; 01494 case AST_FORMAT_SLINEAR: 01495 case AST_FORMAT_SLINEAR16: 01496 samples = f->datalen / 2; 01497 break; 01498 case AST_FORMAT_LPC10: 01499 /* assumes that the RTP packet contains one LPC10 frame */ 01500 samples = 22 * 8; 01501 samples += (((char *)(f->data.ptr))[7] & 0x1) * 8; 01502 break; 01503 case AST_FORMAT_ULAW: 01504 case AST_FORMAT_ALAW: 01505 case AST_FORMAT_TESTLAW: 01506 samples = f->datalen; 01507 break; 01508 case AST_FORMAT_G722: 01509 case AST_FORMAT_ADPCM: 01510 case AST_FORMAT_G726: 01511 case AST_FORMAT_G726_AAL2: 01512 samples = f->datalen * 2; 01513 break; 01514 case AST_FORMAT_SIREN7: 01515 /* 16,000 samples per second at 32kbps is 4,000 bytes per second */ 01516 samples = f->datalen * (16000 / 4000); 01517 break; 01518 case AST_FORMAT_SIREN14: 01519 /* 32,000 samples per second at 48kbps is 6,000 bytes per second */ 01520 samples = (int) f->datalen * ((float) 32000 / 6000); 01521 break; 01522 case AST_FORMAT_G719: 01523 /* 48,000 samples per second at 64kbps is 8,000 bytes per second */ 01524 samples = (int) f->datalen * ((float) 48000 / 8000); 01525 break; 01526 default: 01527 ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), f->subclass.codec)); 01528 } 01529 return samples; 01530 }
static int ast_codec_interp_len | ( | format_t | format | ) | [inline, static] |
Gets duration in ms of interpolation frame for a format.
Definition at line 752 of file frame.h.
References AST_FORMAT_ILBC.
Referenced by __get_from_jb(), and jb_get_and_deliver().
00753 { 00754 return (format == AST_FORMAT_ILBC) ? 30 : 20; 00755 }
int ast_codec_pref_append | ( | struct ast_codec_pref * | pref, | |
format_t | format | |||
) |
Append a audio codec to a preference list, removing it first if it was already there.
Definition at line 1099 of file frame.c.
References ARRAY_LEN, ast_codec_pref_remove(), and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow().
01100 { 01101 int x, newindex = 0; 01102 01103 ast_codec_pref_remove(pref, format); 01104 01105 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01106 if (AST_FORMAT_LIST[x].bits == format) { 01107 newindex = x + 1; 01108 break; 01109 } 01110 } 01111 01112 if (newindex) { 01113 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01114 if (!pref->order[x]) { 01115 pref->order[x] = newindex; 01116 break; 01117 } 01118 } 01119 } 01120 01121 return x; 01122 }
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.
pref | A codec preference list structure | |
buf | A string denoting codec preference, appropriate for use in line transmission | |
size | Size of buf | |
right | Boolean: if 0, convert from buf to pref; if 1, convert from pref to buf. |
Definition at line 1002 of file frame.c.
References ast_codec_pref::order.
Referenced by check_access(), create_addr(), dump_prefs(), and socket_process().
01003 { 01004 int x, differential = (int) 'A', mem; 01005 char *from, *to; 01006 01007 if (right) { 01008 from = pref->order; 01009 to = buf; 01010 mem = size; 01011 } else { 01012 to = pref->order; 01013 from = buf; 01014 mem = sizeof(format_t) * 8; 01015 } 01016 01017 memset(to, 0, mem); 01018 for (x = 0; x < sizeof(format_t) * 8; x++) { 01019 if (!from[x]) 01020 break; 01021 to[x] = right ? (from[x] + differential) : (from[x] - differential); 01022 } 01023 }
struct ast_format_list ast_codec_pref_getsize | ( | struct ast_codec_pref * | pref, | |
format_t | format | |||
) | [read] |
Get packet size for codec.
Definition at line 1205 of file frame.c.
References ARRAY_LEN, ast_getformatname(), ast_log(), AST_LOG_WARNING, 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_instance_bridge(), ast_rtp_write(), handle_open_receive_channel_ack_message(), skinny_set_rtp_peer(), and transmit_connect().
01206 { 01207 int x, idx = -1, framems = 0; 01208 struct ast_format_list fmt = { 0, }; 01209 01210 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01211 if (AST_FORMAT_LIST[x].bits == format) { 01212 fmt = AST_FORMAT_LIST[x]; 01213 idx = x; 01214 break; 01215 } 01216 } 01217 01218 if (idx < 0) { 01219 ast_log(AST_LOG_WARNING, "Format %s unknown; unable to get preferred codec packet size\n", ast_getformatname(format)); 01220 return fmt; 01221 } 01222 01223 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01224 if (pref->order[x] == (idx + 1)) { 01225 framems = pref->framing[x]; 01226 break; 01227 } 01228 } 01229 01230 /* size validation */ 01231 if (!framems) 01232 framems = AST_FORMAT_LIST[idx].def_ms; 01233 01234 if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */ 01235 framems -= framems % AST_FORMAT_LIST[idx].inc_ms; 01236 01237 if (framems < AST_FORMAT_LIST[idx].min_ms) 01238 framems = AST_FORMAT_LIST[idx].min_ms; 01239 01240 if (framems > AST_FORMAT_LIST[idx].max_ms) 01241 framems = AST_FORMAT_LIST[idx].max_ms; 01242 01243 fmt.cur_ms = framems; 01244 01245 return fmt; 01246 }
format_t ast_codec_pref_index | ( | struct ast_codec_pref * | pref, | |
int | index | |||
) |
Codec located at a particular place in the preference index.
Definition at line 1061 of file frame.c.
References ast_format_list::bits, and ast_codec_pref::order.
Referenced by _sip_show_peer(), _skinny_show_line(), add_sdp(), ast_codec_pref_string(), function_iaxpeer(), function_sippeer(), gtalk_invite(), handle_cli_iax2_show_peer(), jingle_accept_call(), print_codec_to_cli(), and socket_process().
01062 { 01063 int slot = 0; 01064 01065 if ((idx >= 0) && (idx < sizeof(pref->order))) { 01066 slot = pref->order[idx]; 01067 } 01068 01069 return slot ? AST_FORMAT_LIST[slot - 1].bits : 0; 01070 }
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, | |
format_t | format, | |||
int | only_if_existing | |||
) |
Prepend an audio codec to a preference list, removing it first if it was already there.
Definition at line 1125 of file frame.c.
References ARRAY_LEN, ast_codec_pref::framing, and ast_codec_pref::order.
Referenced by create_addr().
01126 { 01127 int x, newindex = 0; 01128 01129 /* First step is to get the codecs "index number" */ 01130 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01131 if (AST_FORMAT_LIST[x].bits == format) { 01132 newindex = x + 1; 01133 break; 01134 } 01135 } 01136 /* Done if its unknown */ 01137 if (!newindex) 01138 return; 01139 01140 /* Now find any existing occurrence, or the end */ 01141 for (x = 0; x < sizeof(format_t) * 8; x++) { 01142 if (!pref->order[x] || pref->order[x] == newindex) 01143 break; 01144 } 01145 01146 /* If we failed to find any occurrence, set to the end */ 01147 if (x == sizeof(format_t) * 8) { 01148 --x; 01149 } 01150 01151 if (only_if_existing && !pref->order[x]) 01152 return; 01153 01154 /* Move down to make space to insert - either all the way to the end, 01155 or as far as the existing location (which will be overwritten) */ 01156 for (; x > 0; x--) { 01157 pref->order[x] = pref->order[x - 1]; 01158 pref->framing[x] = pref->framing[x - 1]; 01159 } 01160 01161 /* And insert the new entry */ 01162 pref->order[0] = newindex; 01163 pref->framing[0] = 0; /* ? */ 01164 }
void ast_codec_pref_remove | ( | struct ast_codec_pref * | pref, | |
format_t | format | |||
) |
Remove audio a codec from a preference list.
Definition at line 1073 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().
01074 { 01075 struct ast_codec_pref oldorder; 01076 int x, y = 0; 01077 int slot; 01078 int size; 01079 01080 if (!pref->order[0]) 01081 return; 01082 01083 memcpy(&oldorder, pref, sizeof(oldorder)); 01084 memset(pref, 0, sizeof(*pref)); 01085 01086 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01087 slot = oldorder.order[x]; 01088 size = oldorder.framing[x]; 01089 if (! slot) 01090 break; 01091 if (AST_FORMAT_LIST[slot-1].bits != format) { 01092 pref->order[y] = slot; 01093 pref->framing[y++] = size; 01094 } 01095 } 01096 }
int ast_codec_pref_setsize | ( | struct ast_codec_pref * | pref, | |
format_t | format, | |||
int | framems | |||
) |
Set packet size for codec.
Definition at line 1167 of file frame.c.
References ARRAY_LEN, ast_format_list::def_ms, ast_codec_pref::framing, ast_format_list::inc_ms, ast_format_list::max_ms, ast_format_list::min_ms, and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow(), and process_sdp_a_audio().
01168 { 01169 int x, idx = -1; 01170 01171 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01172 if (AST_FORMAT_LIST[x].bits == format) { 01173 idx = x; 01174 break; 01175 } 01176 } 01177 01178 if (idx < 0) 01179 return -1; 01180 01181 /* size validation */ 01182 if (!framems) 01183 framems = AST_FORMAT_LIST[idx].def_ms; 01184 01185 if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */ 01186 framems -= framems % AST_FORMAT_LIST[idx].inc_ms; 01187 01188 if (framems < AST_FORMAT_LIST[idx].min_ms) 01189 framems = AST_FORMAT_LIST[idx].min_ms; 01190 01191 if (framems > AST_FORMAT_LIST[idx].max_ms) 01192 framems = AST_FORMAT_LIST[idx].max_ms; 01193 01194 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01195 if (pref->order[x] == (idx + 1)) { 01196 pref->framing[x] = framems; 01197 break; 01198 } 01199 } 01200 01201 return x; 01202 }
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 1025 of file frame.c.
References ast_codec_pref_index(), and ast_getformatname().
Referenced by dump_prefs(), and socket_process().
01026 { 01027 int x; 01028 format_t codec; 01029 size_t total_len, slen; 01030 char *formatname; 01031 01032 memset(buf, 0, size); 01033 total_len = size; 01034 buf[0] = '('; 01035 total_len--; 01036 for (x = 0; x < sizeof(format_t) * 8; x++) { 01037 if (total_len <= 0) 01038 break; 01039 if (!(codec = ast_codec_pref_index(pref,x))) 01040 break; 01041 if ((formatname = ast_getformatname(codec))) { 01042 slen = strlen(formatname); 01043 if (slen > total_len) 01044 break; 01045 strncat(buf, formatname, total_len - 1); /* safe */ 01046 total_len -= slen; 01047 } 01048 if (total_len && x < sizeof(format_t) * 8 - 1 && ast_codec_pref_index(pref, x + 1)) { 01049 strncat(buf, "|", total_len - 1); /* safe */ 01050 total_len--; 01051 } 01052 } 01053 if (total_len) { 01054 strncat(buf, ")", total_len - 1); /* safe */ 01055 total_len--; 01056 } 01057 01058 return size - total_len; 01059 }
static force_inline int ast_format_rate | ( | format_t | format | ) | [static] |
Get the sample rate for a given format.
Definition at line 779 of file frame.h.
References AST_FORMAT_G719, AST_FORMAT_G722, AST_FORMAT_SIREN14, AST_FORMAT_SIREN7, AST_FORMAT_SLINEAR16, and AST_FORMAT_SPEEX16.
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_translator_best_choice(), ast_write(), calc_cost(), calc_timestamp(), generator_force(), get_rate_change_result(), handle_cli_core_show_translation(), pitch_shift(), rtp_get_rate(), and schedule_delivery().
00780 { 00781 switch (format) { 00782 case AST_FORMAT_G722: 00783 case AST_FORMAT_SLINEAR16: 00784 case AST_FORMAT_SIREN7: 00785 case AST_FORMAT_SPEEX16: 00786 return 16000; 00787 case AST_FORMAT_SIREN14: 00788 return 32000; 00789 case AST_FORMAT_G719: 00790 return 48000; 00791 default: 00792 return 8000; 00793 } 00794 }
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 1584 of file frame.c.
References AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_slinear_saturated_divide(), ast_slinear_saturated_multiply(), ast_frame_subclass::codec, ast_frame::data, ast_frame::frametype, ast_frame::ptr, ast_frame::samples, and ast_frame::subclass.
Referenced by audiohook_read_frame_single(), audiohook_volume_callback(), conf_run(), and volume_callback().
01585 { 01586 int count; 01587 short *fdata = f->data.ptr; 01588 short adjust_value = abs(adjustment); 01589 01590 if ((f->frametype != AST_FRAME_VOICE) || (f->subclass.codec != AST_FORMAT_SLINEAR)) 01591 return -1; 01592 01593 if (!adjustment) 01594 return 0; 01595 01596 for (count = 0; count < f->samples; count++) { 01597 if (adjustment > 0) { 01598 ast_slinear_saturated_multiply(&fdata[count], &adjust_value); 01599 } else if (adjustment < 0) { 01600 ast_slinear_saturated_divide(&fdata[count], &adjust_value); 01601 } 01602 } 01603 01604 return 0; 01605 }
int ast_frame_clear | ( | struct ast_frame * | frame | ) |
Clear all audio samples from an ast_frame. The frame must be AST_FRAME_VOICE and AST_FORMAT_SLINEAR.
Definition at line 1629 of file frame.c.
References AST_LIST_NEXT, ast_frame::data, ast_frame::datalen, and ast_frame::ptr.
Referenced by ast_audiohook_write_frame(), and mute_callback().
01630 { 01631 struct ast_frame *next; 01632 01633 for (next = AST_LIST_NEXT(frame, frame_list); 01634 frame; 01635 frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) { 01636 memset(frame->data.ptr, 0, frame->datalen); 01637 } 01638 return 0; 01639 }
void ast_frame_dump | ( | const char * | name, | |
struct ast_frame * | f, | |||
char * | prefix | |||
) |
Dump a frame for debugging purposes
Definition at line 778 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, ast_frame_subclass::codec, COLOR_BLACK, COLOR_BRCYAN, COLOR_BRGREEN, COLOR_BRMAGENTA, COLOR_BRRED, COLOR_YELLOW, ast_frame::data, ast_frame::datalen, ast_frame::frametype, ast_frame_subclass::integer, ast_frame::ptr, ast_control_t38_parameters::request_response, ast_frame::subclass, and term_color().
Referenced by __ast_read(), and ast_write().
00779 { 00780 const char noname[] = "unknown"; 00781 char ftype[40] = "Unknown Frametype"; 00782 char cft[80]; 00783 char subclass[40] = "Unknown Subclass"; 00784 char csub[80]; 00785 char moreinfo[40] = ""; 00786 char cn[60]; 00787 char cp[40]; 00788 char cmn[40]; 00789 const char *message = "Unknown"; 00790 00791 if (!name) 00792 name = noname; 00793 00794 00795 if (!f) { 00796 ast_verbose("%s [ %s (NULL) ] [%s]\n", 00797 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00798 term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00799 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00800 return; 00801 } 00802 /* XXX We should probably print one each of voice and video when the format changes XXX */ 00803 if (f->frametype == AST_FRAME_VOICE) 00804 return; 00805 if (f->frametype == AST_FRAME_VIDEO) 00806 return; 00807 switch(f->frametype) { 00808 case AST_FRAME_DTMF_BEGIN: 00809 strcpy(ftype, "DTMF Begin"); 00810 subclass[0] = f->subclass.integer; 00811 subclass[1] = '\0'; 00812 break; 00813 case AST_FRAME_DTMF_END: 00814 strcpy(ftype, "DTMF End"); 00815 subclass[0] = f->subclass.integer; 00816 subclass[1] = '\0'; 00817 break; 00818 case AST_FRAME_CONTROL: 00819 strcpy(ftype, "Control"); 00820 switch (f->subclass.integer) { 00821 case AST_CONTROL_HANGUP: 00822 strcpy(subclass, "Hangup"); 00823 break; 00824 case AST_CONTROL_RING: 00825 strcpy(subclass, "Ring"); 00826 break; 00827 case AST_CONTROL_RINGING: 00828 strcpy(subclass, "Ringing"); 00829 break; 00830 case AST_CONTROL_ANSWER: 00831 strcpy(subclass, "Answer"); 00832 break; 00833 case AST_CONTROL_BUSY: 00834 strcpy(subclass, "Busy"); 00835 break; 00836 case AST_CONTROL_TAKEOFFHOOK: 00837 strcpy(subclass, "Take Off Hook"); 00838 break; 00839 case AST_CONTROL_OFFHOOK: 00840 strcpy(subclass, "Line Off Hook"); 00841 break; 00842 case AST_CONTROL_CONGESTION: 00843 strcpy(subclass, "Congestion"); 00844 break; 00845 case AST_CONTROL_FLASH: 00846 strcpy(subclass, "Flash"); 00847 break; 00848 case AST_CONTROL_WINK: 00849 strcpy(subclass, "Wink"); 00850 break; 00851 case AST_CONTROL_OPTION: 00852 strcpy(subclass, "Option"); 00853 break; 00854 case AST_CONTROL_RADIO_KEY: 00855 strcpy(subclass, "Key Radio"); 00856 break; 00857 case AST_CONTROL_RADIO_UNKEY: 00858 strcpy(subclass, "Unkey Radio"); 00859 break; 00860 case AST_CONTROL_HOLD: 00861 strcpy(subclass, "Hold"); 00862 break; 00863 case AST_CONTROL_UNHOLD: 00864 strcpy(subclass, "Unhold"); 00865 break; 00866 case AST_CONTROL_T38_PARAMETERS: 00867 if (f->datalen != sizeof(struct ast_control_t38_parameters)) { 00868 message = "Invalid"; 00869 } else { 00870 struct ast_control_t38_parameters *parameters = f->data.ptr; 00871 enum ast_control_t38 state = parameters->request_response; 00872 if (state == AST_T38_REQUEST_NEGOTIATE) 00873 message = "Negotiation Requested"; 00874 else if (state == AST_T38_REQUEST_TERMINATE) 00875 message = "Negotiation Request Terminated"; 00876 else if (state == AST_T38_NEGOTIATED) 00877 message = "Negotiated"; 00878 else if (state == AST_T38_TERMINATED) 00879 message = "Terminated"; 00880 else if (state == AST_T38_REFUSED) 00881 message = "Refused"; 00882 } 00883 snprintf(subclass, sizeof(subclass), "T38_Parameters/%s", message); 00884 break; 00885 case -1: 00886 strcpy(subclass, "Stop generators"); 00887 break; 00888 default: 00889 snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass.integer); 00890 } 00891 break; 00892 case AST_FRAME_NULL: 00893 strcpy(ftype, "Null Frame"); 00894 strcpy(subclass, "N/A"); 00895 break; 00896 case AST_FRAME_IAX: 00897 /* Should never happen */ 00898 strcpy(ftype, "IAX Specific"); 00899 snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass.integer); 00900 break; 00901 case AST_FRAME_TEXT: 00902 strcpy(ftype, "Text"); 00903 strcpy(subclass, "N/A"); 00904 ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); 00905 break; 00906 case AST_FRAME_IMAGE: 00907 strcpy(ftype, "Image"); 00908 snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass.codec)); 00909 break; 00910 case AST_FRAME_HTML: 00911 strcpy(ftype, "HTML"); 00912 switch (f->subclass.integer) { 00913 case AST_HTML_URL: 00914 strcpy(subclass, "URL"); 00915 ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); 00916 break; 00917 case AST_HTML_DATA: 00918 strcpy(subclass, "Data"); 00919 break; 00920 case AST_HTML_BEGIN: 00921 strcpy(subclass, "Begin"); 00922 break; 00923 case AST_HTML_END: 00924 strcpy(subclass, "End"); 00925 break; 00926 case AST_HTML_LDCOMPLETE: 00927 strcpy(subclass, "Load Complete"); 00928 break; 00929 case AST_HTML_NOSUPPORT: 00930 strcpy(subclass, "No Support"); 00931 break; 00932 case AST_HTML_LINKURL: 00933 strcpy(subclass, "Link URL"); 00934 ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); 00935 break; 00936 case AST_HTML_UNLINK: 00937 strcpy(subclass, "Unlink"); 00938 break; 00939 case AST_HTML_LINKREJECT: 00940 strcpy(subclass, "Link Reject"); 00941 break; 00942 default: 00943 snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass.integer); 00944 break; 00945 } 00946 break; 00947 case AST_FRAME_MODEM: 00948 strcpy(ftype, "Modem"); 00949 switch (f->subclass.integer) { 00950 case AST_MODEM_T38: 00951 strcpy(subclass, "T.38"); 00952 break; 00953 case AST_MODEM_V150: 00954 strcpy(subclass, "V.150"); 00955 break; 00956 default: 00957 snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass.integer); 00958 break; 00959 } 00960 break; 00961 default: 00962 snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype); 00963 } 00964 if (!ast_strlen_zero(moreinfo)) 00965 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n", 00966 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00967 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00968 f->frametype, 00969 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00970 f->subclass.integer, 00971 term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)), 00972 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00973 else 00974 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n", 00975 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00976 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00977 f->frametype, 00978 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00979 f->subclass.integer, 00980 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00981 }
struct ast_frame* ast_frame_enqueue | ( | struct ast_frame * | head, | |
struct ast_frame * | f, | |||
int | maxlen, | |||
int | dupe | |||
) | [read] |
Appends a frame to the end of a list of frames, truncating the maximum length of the list.
void ast_frame_free | ( | struct ast_frame * | fr, | |
int | cache | |||
) |
Requests a frame to be allocated.
source | Request a frame be allocated. source is an optional source of the frame, len is the requested length, or "0" if the caller will supply the buffer |
Frees a frame or list of frames
fr | Frame to free, or head of list to free | |
cache | Whether to consider this frame for frame caching |
Definition at line 375 of file frame.c.
References __frame_free(), and AST_LIST_NEXT.
Referenced by mixmonitor_thread().
00376 { 00377 struct ast_frame *next; 00378 00379 for (next = AST_LIST_NEXT(frame, frame_list); 00380 frame; 00381 frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) { 00382 __frame_free(frame, cache); 00383 } 00384 }
Sums two frames of audio samples.
f1 | The first frame (which will contain the result) | |
f2 | The second frame |
The frames must be AST_FRAME_VOICE and must contain AST_FORMAT_SLINEAR samples, and must contain the same number of samples.
Definition at line 1607 of file frame.c.
References AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_slinear_saturated_add(), ast_frame_subclass::codec, ast_frame::data, ast_frame::frametype, ast_frame::ptr, ast_frame::samples, and ast_frame::subclass.
01608 { 01609 int count; 01610 short *data1, *data2; 01611 01612 if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass.codec != AST_FORMAT_SLINEAR)) 01613 return -1; 01614 01615 if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass.codec != AST_FORMAT_SLINEAR)) 01616 return -1; 01617 01618 if (f1->samples != f2->samples) 01619 return -1; 01620 01621 for (count = 0, data1 = f1->data.ptr, data2 = f2->data.ptr; 01622 count < f1->samples; 01623 count++, data1++, data2++) 01624 ast_slinear_saturated_add(data1, data2); 01625 01626 return 0; 01627 }
Copies a frame.
fr | frame to copy Duplicates a frame -- should only rarely be used, typically frisolate is good enough |
Definition at line 474 of file frame.c.
References ast_calloc_cache, ast_copy_flags, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, AST_LIST_REMOVE_CURRENT, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, AST_MALLOCD_HDR, ast_threadstorage_get(), ast_frame_subclass::codec, ast_frame::data, ast_frame::datalen, ast_frame::delivery, frame_cache, frames, ast_frame::frametype, ast_frame::len, len(), ast_frame_cache::list, ast_frame::mallocd, ast_frame::mallocd_hdr_len, ast_frame::offset, ast_frame::ptr, ast_frame::samples, ast_frame::seqno, ast_frame_cache::size, ast_frame::src, ast_frame::subclass, ast_frame::ts, and ast_frame::uint32.
Referenced by __ast_queue_frame(), ast_frisolate(), ast_indicate_data(), ast_jb_put(), ast_rtp_write(), ast_slinfactory_feed(), audiohook_read_frame_both(), audiohook_read_frame_single(), autoservice_run(), multicast_rtp_write(), process_dtmf_rfc2833(), and recordthread().
00475 { 00476 struct ast_frame *out = NULL; 00477 int len, srclen = 0; 00478 void *buf = NULL; 00479 00480 #if !defined(LOW_MEMORY) 00481 struct ast_frame_cache *frames; 00482 #endif 00483 00484 /* Start with standard stuff */ 00485 len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00486 /* If we have a source, add space for it */ 00487 /* 00488 * XXX Watch out here - if we receive a src which is not terminated 00489 * properly, we can be easily attacked. Should limit the size we deal with. 00490 */ 00491 if (f->src) 00492 srclen = strlen(f->src); 00493 if (srclen > 0) 00494 len += srclen + 1; 00495 00496 #if !defined(LOW_MEMORY) 00497 if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) { 00498 AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) { 00499 if (out->mallocd_hdr_len >= len) { 00500 size_t mallocd_len = out->mallocd_hdr_len; 00501 00502 AST_LIST_REMOVE_CURRENT(frame_list); 00503 memset(out, 0, sizeof(*out)); 00504 out->mallocd_hdr_len = mallocd_len; 00505 buf = out; 00506 frames->size--; 00507 break; 00508 } 00509 } 00510 AST_LIST_TRAVERSE_SAFE_END; 00511 } 00512 #endif 00513 00514 if (!buf) { 00515 if (!(buf = ast_calloc_cache(1, len))) 00516 return NULL; 00517 out = buf; 00518 out->mallocd_hdr_len = len; 00519 } 00520 00521 out->frametype = f->frametype; 00522 out->subclass.codec = f->subclass.codec; 00523 out->datalen = f->datalen; 00524 out->samples = f->samples; 00525 out->delivery = f->delivery; 00526 /* Even though this new frame was allocated from the heap, we can't mark it 00527 * with AST_MALLOCD_HDR, AST_MALLOCD_DATA and AST_MALLOCD_SRC, because that 00528 * would cause ast_frfree() to attempt to individually free each of those 00529 * under the assumption that they were separately allocated. Since this frame 00530 * was allocated in a single allocation, we'll only mark it as if the header 00531 * was heap-allocated; this will result in the entire frame being properly freed. 00532 */ 00533 out->mallocd = AST_MALLOCD_HDR; 00534 out->offset = AST_FRIENDLY_OFFSET; 00535 if (out->datalen) { 00536 out->data.ptr = buf + sizeof(*out) + AST_FRIENDLY_OFFSET; 00537 memcpy(out->data.ptr, f->data.ptr, out->datalen); 00538 } else { 00539 out->data.uint32 = f->data.uint32; 00540 } 00541 if (srclen > 0) { 00542 /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */ 00543 char *src; 00544 out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00545 src = (char *) out->src; 00546 /* Must have space since we allocated for it */ 00547 strcpy(src, f->src); 00548 } 00549 ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO); 00550 out->ts = f->ts; 00551 out->len = f->len; 00552 out->seqno = f->seqno; 00553 return out; 00554 }
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 391 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_subclass::codec, 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_dsp_process(), ast_rtp_read(), ast_safe_sleep_conditional(), ast_slinfactory_feed(), ast_trans_frameout(), ast_write(), autoservice_run(), dahdi_decoder_frameout(), dahdi_encoder_frameout(), feature_request_and_dial(), jpeg_read_image(), read_frame(), spandsp_fax_read(), and t38_tx_packet_handler().
00392 { 00393 struct ast_frame *out; 00394 void *newdata; 00395 00396 /* if none of the existing frame is malloc'd, let ast_frdup() do it 00397 since it is more efficient 00398 */ 00399 if (fr->mallocd == 0) { 00400 return ast_frdup(fr); 00401 } 00402 00403 /* if everything is already malloc'd, we are done */ 00404 if ((fr->mallocd & (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) == 00405 (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) { 00406 return fr; 00407 } 00408 00409 if (!(fr->mallocd & AST_MALLOCD_HDR)) { 00410 /* Allocate a new header if needed */ 00411 if (!(out = ast_frame_header_new())) { 00412 return NULL; 00413 } 00414 out->frametype = fr->frametype; 00415 out->subclass.codec = fr->subclass.codec; 00416 out->datalen = fr->datalen; 00417 out->samples = fr->samples; 00418 out->offset = fr->offset; 00419 /* Copy the timing data */ 00420 ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO); 00421 if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) { 00422 out->ts = fr->ts; 00423 out->len = fr->len; 00424 out->seqno = fr->seqno; 00425 } 00426 } else { 00427 out = fr; 00428 } 00429 00430 if (!(fr->mallocd & AST_MALLOCD_SRC) && fr->src) { 00431 if (!(out->src = ast_strdup(fr->src))) { 00432 if (out != fr) { 00433 ast_free(out); 00434 } 00435 return NULL; 00436 } 00437 } else { 00438 out->src = fr->src; 00439 fr->src = NULL; 00440 fr->mallocd &= ~AST_MALLOCD_SRC; 00441 } 00442 00443 if (!(fr->mallocd & AST_MALLOCD_DATA)) { 00444 if (!fr->datalen) { 00445 out->data.uint32 = fr->data.uint32; 00446 out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC; 00447 return out; 00448 } 00449 if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) { 00450 if (out->src != fr->src) { 00451 ast_free((void *) out->src); 00452 } 00453 if (out != fr) { 00454 ast_free(out); 00455 } 00456 return NULL; 00457 } 00458 newdata += AST_FRIENDLY_OFFSET; 00459 out->offset = AST_FRIENDLY_OFFSET; 00460 out->datalen = fr->datalen; 00461 memcpy(newdata, fr->data.ptr, fr->datalen); 00462 out->data.ptr = newdata; 00463 } else { 00464 out->data = fr->data; 00465 memset(&fr->data, 0, sizeof(fr->data)); 00466 fr->mallocd &= ~AST_MALLOCD_DATA; 00467 } 00468 00469 out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA; 00470 00471 return out; 00472 }
struct ast_format_list* ast_get_format_list | ( | size_t * | size | ) | [read] |
Definition at line 572 of file frame.c.
References ARRAY_LEN.
Referenced by ast_data_add_codecs(), complete_trans_path_choice(), and handle_cli_core_show_translation().
00573 { 00574 *size = ARRAY_LEN(AST_FORMAT_LIST); 00575 return AST_FORMAT_LIST; 00576 }
struct ast_format_list* ast_get_format_list_index | ( | int | index | ) | [read] |
Definition at line 567 of file frame.c.
00568 { 00569 return &AST_FORMAT_LIST[idx]; 00570 }
format_t ast_getformatbyname | ( | const char * | name | ) |
Gets a format from a name.
name | string of format |
Definition at line 641 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().
00642 { 00643 int x, all; 00644 format_t format = 0; 00645 00646 all = strcasecmp(name, "all") ? 0 : 1; 00647 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00648 if (all || 00649 !strcasecmp(AST_FORMAT_LIST[x].name,name) || 00650 !strcasecmp(AST_FORMAT_LIST[x].name, ast_expand_codec_alias(name))) { 00651 format |= AST_FORMAT_LIST[x].bits; 00652 if (!all) 00653 break; 00654 } 00655 } 00656 00657 return format; 00658 }
char* ast_getformatname | ( | format_t | format | ) |
Get the name of a format.
format | id of format |
Definition at line 578 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(), __ast_smoother_feed(), _sip_show_peer(), _skinny_show_line(), add_codec_to_answer(), add_codec_to_sdp(), add_sdp(), add_tcodec_to_sdp(), add_vcodec_to_sdp(), agent_call(), ast_channel_make_compatible_helper(), ast_codec_get_len(), ast_codec_pref_getsize(), ast_codec_pref_string(), ast_do_masquerade(), ast_dsp_process(), ast_frame_dump(), ast_openvstream(), ast_rtp_instance_bridge(), ast_rtp_write(), ast_slinfactory_feed(), ast_stopstream(), ast_streamfile(), ast_translate_path_to_str(), ast_translator_build_path(), ast_unregister_translator(), ast_write(), ast_writestream(), background_detect_exec(), bridge_channel_join(), bridge_make_compatible(), conf_run(), dahdi_read(), dahdi_write(), do_waiting(), dump_versioned_codec(), eagi_exec(), func_channel_read(), function_iaxpeer(), function_sippeer(), g719write(), g726_write(), g729_write(), gsm_write(), gtalk_rtp_read(), gtalk_show_channels(), gtalk_write(), h263_write(), h264_write(), 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(), handle_open_receive_channel_ack_message(), iax2_request(), iax_show_provisioning(), ilbc_write(), isAnsweringMachine(), jack_hook_callback(), jingle_rtp_read(), jingle_show_channels(), jingle_write(), login_exec(), mgcp_rtp_read(), mgcp_write(), misdn_write(), moh_files_release(), moh_release(), nbs_request(), nbs_xwrite(), ogg_vorbis_write(), oh323_rtp_read(), oh323_write(), pcm_write(), phone_setup(), phone_write(), print_codec_to_cli(), print_frame(), process_sdp_a_audio(), rebuild_matrix(), register_translator(), remote_bridge_loop(), set_format(), set_local_capabilities(), set_peer_capabilities(), setup_rtp_connection(), show_codecs(), sip_request_call(), sip_rtp_read(), sip_write(), siren14write(), siren7write(), skinny_new(), skinny_rtp_read(), skinny_set_rtp_peer(), skinny_write(), slinear_write(), socket_process(), start_rtp(), unistim_new(), unistim_request(), unistim_rtp_read(), unistim_write(), vox_write(), and wav_write().
00579 { 00580 int x; 00581 char *ret = "unknown"; 00582 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00583 if (AST_FORMAT_LIST[x].bits == format) { 00584 ret = AST_FORMAT_LIST[x].name; 00585 break; 00586 } 00587 } 00588 return ret; 00589 }
char* ast_getformatname_multiple | ( | char * | buf, | |
size_t | size, | |||
format_t | 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 591 of file frame.c.
References ARRAY_LEN, ast_copy_string(), ast_format_list::bits, len(), and name.
Referenced by __ast_read(), _sip_show_peer(), _skinny_show_device(), _skinny_show_line(), add_sdp(), alsa_request(), ast_best_codec(), ast_codec_get_samples(), ast_request(), ast_streamfile(), ast_write(), bridge_make_compatible(), console_request(), function_iaxpeer(), function_sippeer(), gtalk_is_answered(), gtalk_newcall(), gtalk_write(), handle_capabilities_res_message(), handle_cli_core_show_channeltype(), handle_cli_iax2_show_peer(), handle_showchan(), iax2_bridge(), jingle_write(), mgcp_request(), mgcp_write(), oh323_request(), oh323_write(), oss_request(), phone_request(), process_sdp(), serialize_showchan(), set_format(), setup_rtp_connection(), show_channels_cb(), sip_new(), sip_request_call(), sip_show_channel(), sip_show_settings(), sip_write(), skinny_new(), skinny_request(), skinny_write(), socket_process(), start_rtp(), unistim_new(), unistim_request(), and unistim_write().
00592 { 00593 int x; 00594 unsigned len; 00595 char *start, *end = buf; 00596 00597 if (!size) 00598 return buf; 00599 snprintf(end, size, "0x%llx (", (unsigned long long) format); 00600 len = strlen(end); 00601 end += len; 00602 size -= len; 00603 start = end; 00604 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00605 if (AST_FORMAT_LIST[x].bits & format) { 00606 snprintf(end, size, "%s|", AST_FORMAT_LIST[x].name); 00607 len = strlen(end); 00608 end += len; 00609 size -= len; 00610 } 00611 } 00612 if (start == end) 00613 ast_copy_string(start, "nothing)", size); 00614 else if (size > 1) 00615 *(end - 1) = ')'; 00616 return buf; 00617 }
int ast_parse_allow_disallow | ( | struct ast_codec_pref * | pref, | |
format_t * | 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 1272 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, and parse().
Referenced by action_originate(), apply_outgoing(), build_peer(), build_user(), config_parse_variables(), gtalk_create_member(), gtalk_load_config(), jingle_create_member(), jingle_load_config(), reload_config(), set_config(), skinny_unregister(), and update_common_options().
01273 { 01274 int errors = 0, framems = 0; 01275 char *parse = NULL, *this = NULL, *psize = NULL; 01276 format_t format = 0; 01277 01278 parse = ast_strdupa(list); 01279 while ((this = strsep(&parse, ","))) { 01280 framems = 0; 01281 if ((psize = strrchr(this, ':'))) { 01282 *psize++ = '\0'; 01283 ast_debug(1, "Packetization for codec: %s is %s\n", this, psize); 01284 framems = atoi(psize); 01285 if (framems < 0) { 01286 framems = 0; 01287 errors++; 01288 ast_log(LOG_WARNING, "Bad packetization value for codec %s\n", this); 01289 } 01290 } 01291 if (!(format = ast_getformatbyname(this))) { 01292 ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this); 01293 errors++; 01294 continue; 01295 } 01296 01297 if (mask) { 01298 if (allowing) 01299 *mask |= format; 01300 else 01301 *mask &= ~format; 01302 } 01303 01304 /* Set up a preference list for audio. Do not include video in preferences 01305 since we can not transcode video and have to use whatever is offered 01306 */ 01307 if (pref && (format & AST_FORMAT_AUDIO_MASK)) { 01308 if (strcasecmp(this, "all")) { 01309 if (allowing) { 01310 ast_codec_pref_append(pref, format); 01311 ast_codec_pref_setsize(pref, format, framems); 01312 } 01313 else 01314 ast_codec_pref_remove(pref, format); 01315 } else if (!allowing) { 01316 memset(pref, 0, sizeof(*pref)); 01317 } 01318 } 01319 } 01320 return errors; 01321 }
void ast_smoother_free | ( | struct ast_smoother * | s | ) |
Definition at line 294 of file frame.c.
References ast_free.
Referenced by ast_rtp_destroy(), ast_rtp_write(), destroy_session(), and generic_fax_exec().
00295 { 00296 ast_free(s); 00297 }
int ast_smoother_get_flags | ( | struct ast_smoother * | smoother | ) |
Definition at line 193 of file frame.c.
References ast_smoother::flags.
struct ast_smoother* ast_smoother_new | ( | int | bytes | ) | [read] |
Definition at line 183 of file frame.c.
References ast_malloc, and ast_smoother_reset().
Referenced by ast_rtp_write(), and generic_fax_exec().
00184 { 00185 struct ast_smoother *s; 00186 if (size < 1) 00187 return NULL; 00188 if ((s = ast_malloc(sizeof(*s)))) 00189 ast_smoother_reset(s, size); 00190 return s; 00191 }
struct ast_frame* ast_smoother_read | ( | struct ast_smoother * | s | ) | [read] |
Definition at line 244 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_frame_subclass::codec, ast_smoother::data, ast_frame::data, ast_frame::datalen, ast_smoother::delivery, ast_frame::delivery, ast_smoother::f, ast_smoother::flags, ast_smoother::format, ast_smoother::framedata, ast_frame::frametype, ast_smoother::len, len(), LOG_WARNING, ast_frame::offset, ast_smoother::opt, ast_frame::ptr, ast_frame::samples, ast_smoother::samplesperbyte, ast_smoother::size, and ast_frame::subclass.
Referenced by ast_rtp_write(), and generic_fax_exec().
00245 { 00246 struct ast_frame *opt; 00247 int len; 00248 00249 /* IF we have an optimization frame, send it */ 00250 if (s->opt) { 00251 if (s->opt->offset < AST_FRIENDLY_OFFSET) 00252 ast_log(LOG_WARNING, "Returning a frame of inappropriate offset (%d).\n", 00253 s->opt->offset); 00254 opt = s->opt; 00255 s->opt = NULL; 00256 return opt; 00257 } 00258 00259 /* Make sure we have enough data */ 00260 if (s->len < s->size) { 00261 /* Or, if this is a G.729 frame with VAD on it, send it immediately anyway */ 00262 if (!((s->flags & AST_SMOOTHER_FLAG_G729) && (s->len % 10))) 00263 return NULL; 00264 } 00265 len = s->size; 00266 if (len > s->len) 00267 len = s->len; 00268 /* Make frame */ 00269 s->f.frametype = AST_FRAME_VOICE; 00270 s->f.subclass.codec = s->format; 00271 s->f.data.ptr = s->framedata + AST_FRIENDLY_OFFSET; 00272 s->f.offset = AST_FRIENDLY_OFFSET; 00273 s->f.datalen = len; 00274 /* Samples will be improper given VAD, but with VAD the concept really doesn't even exist */ 00275 s->f.samples = len * s->samplesperbyte; /* XXX rounding */ 00276 s->f.delivery = s->delivery; 00277 /* Fill Data */ 00278 memcpy(s->f.data.ptr, s->data, len); 00279 s->len -= len; 00280 /* Move remaining data to the front if applicable */ 00281 if (s->len) { 00282 /* In principle this should all be fine because if we are sending 00283 G.729 VAD, the next timestamp will take over anyawy */ 00284 memmove(s->data, s->data + len, s->len); 00285 if (!ast_tvzero(s->delivery)) { 00286 /* If we have delivery time, increment it, otherwise, leave it at 0 */ 00287 s->delivery = ast_tvadd(s->delivery, ast_samp2tv(s->f.samples, ast_format_rate(s->format))); 00288 } 00289 } 00290 /* Return frame */ 00291 return &s->f; 00292 }
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 161 of file frame.c.
References ast_smoother::opt, ast_smoother::opt_needs_swap, ast_smoother::size, and smoother_frame_feed().
00162 { 00163 /* if there is no change, then nothing to do */ 00164 if (s->size == bytes) { 00165 return; 00166 } 00167 /* set the new desired output size */ 00168 s->size = bytes; 00169 /* if there is no 'optimized' frame in the smoother, 00170 * then there is nothing left to do 00171 */ 00172 if (!s->opt) { 00173 return; 00174 } 00175 /* there is an 'optimized' frame here at the old size, 00176 * but it must now be put into the buffer so the data 00177 * can be extracted at the new size 00178 */ 00179 smoother_frame_feed(s, s->opt, s->opt_needs_swap); 00180 s->opt = NULL; 00181 }
void ast_smoother_reset | ( | struct ast_smoother * | s, | |
int | bytes | |||
) |
Definition at line 155 of file frame.c.
References ast_smoother::size.
Referenced by ast_smoother_new().
00156 { 00157 memset(s, 0, sizeof(*s)); 00158 s->size = bytes; 00159 }
void ast_smoother_set_flags | ( | struct ast_smoother * | smoother, | |
int | flags | |||
) |
Definition at line 198 of file frame.c.
References ast_smoother::flags.
Referenced by ast_rtp_write().
00199 { 00200 s->flags = flags; 00201 }
int ast_smoother_test_flag | ( | struct ast_smoother * | s, | |
int | flag | |||
) |
Definition at line 203 of file frame.c.
References ast_smoother::flags.
Referenced by ast_rtp_write().
00204 { 00205 return (s->flags & flag); 00206 }
void ast_swapcopy_samples | ( | void * | dst, | |
const void * | src, | |||
int | samples | |||
) |
Definition at line 556 of file frame.c.
Referenced by __ast_smoother_feed(), iax_frame_wrap(), phone_write_buf(), and smoother_frame_feed().
struct ast_frame ast_null_frame |
Queueing a null frame is fairly common, so we declare a global null frame object for this purpose instead of having to declare one on the stack
Definition at line 131 of file frame.c.
Referenced by __analog_handle_event(), __ast_channel_masquerade(), __ast_read(), __oh323_rtp_create(), __oh323_update_info(), agent_read(), agent_request(), ast_channel_setwhentohangup_tv(), ast_do_masquerade(), ast_rtcp_read(), ast_rtp_read(), ast_softhangup_nolock(), ast_udptl_read(), bridge_read(), conf_run(), console_read(), create_dtmf_frame(), dahdi_handle_event(), dahdi_read(), gtalk_rtp_read(), handle_request_invite(), handle_response_invite(), iax2_read(), jingle_rtp_read(), local_read(), mgcp_rtp_read(), multicast_rtp_read(), oh323_read(), oh323_rtp_read(), process_sdp(), sip_read(), sip_rtp_read(), skinny_rtp_read(), spandsp_fax_read(), unistim_rtp_read(), and wakeup_sub().