#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 |
AST_Smoother | |
#define | ast_smoother_feed(s, f) __ast_smoother_feed(s, f, 0) |
#define | ast_smoother_feed_be(s, f) __ast_smoother_feed(s, f, 0) |
#define | ast_smoother_feed_le(s, f) __ast_smoother_feed(s, f, 1) |
int | __ast_smoother_feed (struct ast_smoother *s, struct ast_frame *f, int swap) |
void | ast_smoother_free (struct ast_smoother *s) |
int | ast_smoother_get_flags (struct ast_smoother *smoother) |
ast_smoother * | ast_smoother_new (int bytes) |
ast_frame * | ast_smoother_read (struct ast_smoother *s) |
void | ast_smoother_reconfigure (struct ast_smoother *s, int bytes) |
Reconfigure an existing smoother to output a different number of bytes per frame. | |
void | ast_smoother_reset (struct ast_smoother *s, int bytes) |
void | ast_smoother_set_flags (struct ast_smoother *smoother, int flags) |
int | ast_smoother_test_flag (struct ast_smoother *s, int flag) |
Defines | |
#define | AST_FORMAT_ADPCM (1ULL << 5) |
#define | AST_FORMAT_ALAW (1ULL << 3) |
#define | AST_FORMAT_AUDIO_MASK 0xFFFF0000FFFFULL |
#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 } |
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. | |
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) |
ast_frame * | ast_frame_enqueue (struct ast_frame *head, struct ast_frame *f, int maxlen, int dupe) |
Appends a frame to the end of a list of frames, truncating the maximum length of the list. | |
void | ast_frame_free (struct ast_frame *fr, int cache) |
Requests a frame to be allocated Frees a frame or list of frames. | |
int | ast_frame_slinear_sum (struct ast_frame *f1, struct ast_frame *f2) |
Sums two frames of audio samples. | |
ast_frame * | ast_frdup (const struct ast_frame *fr) |
Copies a frame. | |
ast_frame * | ast_frisolate (struct ast_frame *fr) |
Makes a frame independent of any static storage. | |
ast_format_list * | ast_get_format_list (size_t *size) |
ast_format_list * | ast_get_format_list_index (int index) |
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 | |
ast_frame | ast_null_frame |
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(), convertcap(), 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(), convertcap(), dahdi_new(), dahdi_read(), dahdi_write(), find_transcoders(), is_encoder(), misdn_read(), oh323_rtp_read(), pcm_seek(), pcm_write(), and start_rtp().
#define AST_FORMAT_AUDIO_MASK 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_closestream(), ast_filehelper(), ast_openstream_full(), ast_openvstream(), ast_playstream(), ast_request(), ast_rtp_read(), ast_translate_available_formats(), ast_translator_best_choice(), ast_writestream(), begin_dial_channel(), complete_trans_path_choice(), filestream_destructor(), 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_G719 (1ULL << 32) |
G.719 (64 kbps assumed)
Definition at line 298 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(), convertcap(), 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(), convertcap(), 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(), convertcap(), 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(), convertcap(), 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 282 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 286 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(), convertcap(), 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(), attempt_reconnect(), audio_audiohook_write_list(), audiohook_read_frame_both(), audiohook_read_frame_single(), background_detect_exec(), bridge_request(), build_conf(), chanspy_exec(), conf_run(), connect_link(), 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(), handle_jack_audio(), handle_recordfile(), handle_speechcreate(), handle_speechrecognize(), iax_frame_wrap(), ices_exec(), init_outgoing(), is_encoder(), isAnsweringMachine(), jack_exec(), jack_hook_callback(), linear_alloc(), linear_generator(), load_module(), load_moh_classes(), local_ast_moh_start(), measurenoise(), mixmonitor_thread(), mp3_exec(), nbs_request(), nbs_xwrite(), NBScat_exec(), ogg_vorbis_read(), ogg_vorbis_write(), oh323_rtp_read(), orig_app(), orig_exten(), 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(), rpt(), rpt_call(), rpt_exec(), rpt_tele_thread(), 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(), transmit_audio(), usbradio_new(), usbradio_read(), usbradio_request(), 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(), 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(), convertcap(), and speex_sample().
#define AST_FORMAT_SPEEX16 (1ULL << 33) |
SpeeX Wideband (16kHz) Free Compression
Definition at line 300 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 293 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 291 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 302 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 296 of file frame.h.
Referenced by add_sdp(), ast_request(), 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(), alarmreceiver_exec(), ast_adsi_transmit_message_full(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), calc_energy(), codec_ast2skinny(), codec_skinny2ast(), conf_run(), convertcap(), dahdi_new(), dahdi_read(), dahdi_translate(), dahdi_write(), find_transcoders(), is_encoder(), load_module(), milliwatt_generate(), oh323_rtp_read(), old_milliwatt_exec(), phone_request(), phone_setup(), phone_write(), pri_dchannel(), send_tone_burst(), start_rtp(), and ulaw_sample().
#define AST_FORMAT_VIDEO_MASK ((((1ULL << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) | 0x7FFF000000000000ULL) |
Definition at line 289 of file frame.h.
Referenced by add_sdp(), ast_filehelper(), ast_openvstream(), ast_request(), ast_rtp_read(), ast_translate_available_formats(), dialog_initialize_rtp(), 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) |
#define ast_frame_byteswap_le | ( | fr | ) | do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data.ptr, __f->data.ptr, __f->samples); } while(0) |
#define AST_FRAME_DTMF AST_FRAME_DTMF_END |
Definition at line 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_bridge_call(), 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(), jingle_handle_dtmf(), keypad_digit(), mgcp_rtp_read(), misdn_bridge(), mp3_exec(), NBScat_exec(), oh323_rtp_read(), phone_exception(), process_ast_dsp(), receive_dtmf_digits(), rpt(), rpt_call(), send_waveform_to_channel(), sip_rtp_read(), speech_background(), unistim_do_senddigit(), unistim_senddigit_end(), volume_callback(), and wait_for_winner().
#define AST_FRAME_SET_BUFFER | ( | fr, | |||
_base, | |||||
_ofs, | |||||
_datalen | ) |
Value:
{ \ (fr)->data.ptr = (char *)_base + (_ofs); \ (fr)->offset = (_ofs); \ (fr)->datalen = (_datalen); \ }
Definition at line 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 550 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_framehook_attach(), 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_wait(), audio_audiohook_write_list(), autoservice_run(), background_detect_exec(), bridge_handle_dtmf(), calc_cost(), channel_spy(), check_goto_on_transfer(), conf_flush(), conf_free(), conf_run(), create_jb(), dahdi_bridge(), dial_exec_full(), dictate_exec(), disa_exec(), disable_t38(), do_idle_thread(), 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(), iax_park_thread(), ices_exec(), isAnsweringMachine(), jack_exec(), jb_empty_and_reset_adaptive(), jb_empty_and_reset_fixed(), jb_get_and_deliver(), launch_asyncagi(), local_bridge_loop(), manage_parkinglot(), masq_park_call(), measurenoise(), moh_files_generator(), monitor_dial(), mp3_exec(), multicast_rtp_write(), NBScat_exec(), read_frame(), receive_dtmf_digits(), receivefax_t38_init(), recordthread(), remote_bridge_loop(), rpt(), run_agi(), send_tone_burst(), send_waveform_to_channel(), sendfax_t38_init(), sendurl_exec(), session_destroy(), speech_background(), spy_generate(), transmit_audio(), transmit_t38(), 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(), process_cn_rfc3389(), send_tone_burst(), send_waveform_to_channel(), siren14read(), siren7read(), slinear_read(), sms_generate(), spandsp_fax_read(), usbradio_read(), vox_read(), and wav_read().
#define AST_HTML_BEGIN 4 |
#define AST_HTML_DATA 2 |
#define AST_HTML_END 8 |
#define AST_HTML_LDCOMPLETE 16 |
Load is complete
Definition at line 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 |
#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 420 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 487 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 451 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 483 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 475 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 443 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 479 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 |
#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 461 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 465 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 471 of file frame.h.
Referenced by ast_channel_make_compatible_helper(), and sip_setoption().
#define AST_OPTION_OPRMODE 7 |
Definition at line 436 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 416 of file frame.h.
Referenced by ast_bridge_call(), dahdi_setoption(), iax2_setoption(), and rpt().
#define AST_OPTION_RXGAIN 6 |
Set channel receive gain Option data is a single signed char representing number of decibels (dB) to set gain to (on top of any gain specified in channel driver)
Definition at line 430 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 492 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 491 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 457 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 412 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 408 of file frame.h.
Referenced by analog_hangup(), ast_bridge_call(), conf_run(), dahdi_hangup(), dahdi_setoption(), iax2_setoption(), rpt(), and rpt_exec().
#define AST_OPTION_TXGAIN 5 |
Set channel transmit gain Option data is a single signed char representing number of decibels (dB) to set gain to (on top of any gain specified in channel driver)
Definition at line 425 of file frame.h.
Referenced by common_exec(), dahdi_setoption(), func_channel_write_real(), iax2_setoption(), reset_volumes(), and set_listen_volume().
#define AST_SMOOTHER_FLAG_BE (1 << 1) |
#define AST_SMOOTHER_FLAG_G729 (1 << 0) |
Definition at line 394 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 | |
AST_CONTROL_CC | T38 state change request/notification with parameters 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. |
Definition at line 306 of file frame.h.
00306 { 00307 AST_CONTROL_HANGUP = 1, /*!< Other end has hungup */ 00308 AST_CONTROL_RING = 2, /*!< Local ring */ 00309 AST_CONTROL_RINGING = 3, /*!< Remote end is ringing */ 00310 AST_CONTROL_ANSWER = 4, /*!< Remote end has answered */ 00311 AST_CONTROL_BUSY = 5, /*!< Remote end is busy */ 00312 AST_CONTROL_TAKEOFFHOOK = 6, /*!< Make it go off hook */ 00313 AST_CONTROL_OFFHOOK = 7, /*!< Line is off hook */ 00314 AST_CONTROL_CONGESTION = 8, /*!< Congestion (circuits busy) */ 00315 AST_CONTROL_FLASH = 9, /*!< Flash hook */ 00316 AST_CONTROL_WINK = 10, /*!< Wink */ 00317 AST_CONTROL_OPTION = 11, /*!< Set a low-level option */ 00318 AST_CONTROL_RADIO_KEY = 12, /*!< Key Radio */ 00319 AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */ 00320 AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */ 00321 AST_CONTROL_PROCEEDING = 15, /*!< Indicate CALL PROCEEDING */ 00322 AST_CONTROL_HOLD = 16, /*!< Indicate call is placed on hold */ 00323 AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold */ 00324 AST_CONTROL_VIDUPDATE = 18, /*!< Indicate video frame update */ 00325 _XXX_AST_CONTROL_T38 = 19, /*!< T38 state change request/notification \deprecated This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead. */ 00326 AST_CONTROL_SRCUPDATE = 20, /*!< Indicate source of media has changed */ 00327 AST_CONTROL_TRANSFER = 21, /*!< Indicate status of a transfer request */ 00328 AST_CONTROL_CONNECTED_LINE = 22,/*!< Indicate connected line has changed */ 00329 AST_CONTROL_REDIRECTING = 23, /*!< Indicate redirecting id has changed */ 00330 AST_CONTROL_T38_PARAMETERS = 24, /*! T38 state change request/notification with parameters */ 00331 AST_CONTROL_CC = 25, /*!< Indication that Call completion service is possible */ 00332 AST_CONTROL_SRCCHANGE = 26, /*!< Media source has changed and requires a new RTP SSRC */ 00333 AST_CONTROL_READ_ACTION = 27, /*!< Tell ast_read to take a specific action */ 00334 AST_CONTROL_AOC = 28, /*!< Advice of Charge with encoded generic AOC payload */ 00335 AST_CONTROL_END_OF_Q = 29, /*!< Indicate that this position was the end of the channel queue for a softhangup. */ 00336 };
enum ast_control_t38 |
Definition at line 355 of file frame.h.
00355 { 00356 AST_T38_REQUEST_NEGOTIATE = 1, /*!< Request T38 on a channel (voice to fax) */ 00357 AST_T38_REQUEST_TERMINATE, /*!< Terminate T38 on a channel (fax to voice) */ 00358 AST_T38_NEGOTIATED, /*!< T38 negotiated (fax mode) */ 00359 AST_T38_TERMINATED, /*!< T38 terminated (back to voice) */ 00360 AST_T38_REFUSED, /*!< T38 refused for some reason (usually rejected by remote end) */ 00361 AST_T38_REQUEST_PARMS, /*!< request far end T.38 parameters for a channel in 'negotiating' state */ 00362 };
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 364 of file frame.h.
00364 { 00365 AST_T38_RATE_2400 = 0, 00366 AST_T38_RATE_4800, 00367 AST_T38_RATE_7200, 00368 AST_T38_RATE_9600, 00369 AST_T38_RATE_12000, 00370 AST_T38_RATE_14400, 00371 };
Definition at line 373 of file frame.h.
00373 { 00374 AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF = 0, 00375 AST_T38_RATE_MANAGEMENT_LOCAL_TCF, 00376 };
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 389 of file frame.h.
00389 { 00390 AST_TRANSFER_SUCCESS = 0, /*!< Transfer request on the channel worked */ 00391 AST_TRANSFER_FAILED, /*!< Transfer request on the channel failed */ 00392 };
Definition at line 338 of file frame.h.
00338 { 00339 AST_FRAME_READ_ACTION_CONNECTED_LINE_MACRO, 00340 };
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 204 of file frame.c.
References AST_FRAME_VOICE, ast_getformatname(), ast_log(), AST_MIN_OFFSET, AST_SMOOTHER_FLAG_G729, ast_swapcopy_samples(), f, ast_smoother::flags, ast_smoother::format, ast_smoother::len, LOG_WARNING, ast_smoother::opt, ast_smoother::opt_needs_swap, ast_smoother::samplesperbyte, ast_smoother::size, smoother_frame_feed(), and SMOOTHER_SIZE.
00205 { 00206 if (f->frametype != AST_FRAME_VOICE) { 00207 ast_log(LOG_WARNING, "Huh? Can't smooth a non-voice frame!\n"); 00208 return -1; 00209 } 00210 if (!s->format) { 00211 s->format = f->subclass.codec; 00212 s->samplesperbyte = (float)f->samples / (float)f->datalen; 00213 } else if (s->format != f->subclass.codec) { 00214 ast_log(LOG_WARNING, "Smoother was working on %s format frames, now trying to feed %s?\n", 00215 ast_getformatname(s->format), ast_getformatname(f->subclass.codec)); 00216 return -1; 00217 } 00218 if (s->len + f->datalen > SMOOTHER_SIZE) { 00219 ast_log(LOG_WARNING, "Out of smoother space\n"); 00220 return -1; 00221 } 00222 if (((f->datalen == s->size) || 00223 ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) && 00224 !s->opt && 00225 !s->len && 00226 (f->offset >= AST_MIN_OFFSET)) { 00227 /* Optimize by sending the frame we just got 00228 on the next read, thus eliminating the douple 00229 copy */ 00230 if (swap) 00231 ast_swapcopy_samples(f->data.ptr, f->data.ptr, f->samples); 00232 s->opt = f; 00233 s->opt_needs_swap = swap ? 1 : 0; 00234 return 0; 00235 } 00236 00237 return smoother_frame_feed(s, f, swap); 00238 }
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 651 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, and ast_format_list::desc.
Referenced by moh_alloc(), and show_codec_n().
00652 { 00653 int x; 00654 char *ret = "unknown"; 00655 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00656 if (AST_FORMAT_LIST[x].bits == codec) { 00657 ret = AST_FORMAT_LIST[x].desc; 00658 break; 00659 } 00660 } 00661 return ret; 00662 }
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 1224 of file frame.c.
References ARRAY_LEN, ast_best_codec(), ast_debug, AST_FORMAT_LIST, ast_format_list::bits, and ast_codec_pref::order.
Referenced by __oh323_new(), gtalk_new(), jingle_new(), process_sdp(), sip_new(), and socket_process().
01225 { 01226 int x, slot; 01227 format_t ret = 0; 01228 01229 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01230 slot = pref->order[x]; 01231 01232 if (!slot) 01233 break; 01234 if (formats & AST_FORMAT_LIST[slot-1].bits) { 01235 ret = AST_FORMAT_LIST[slot-1].bits; 01236 break; 01237 } 01238 } 01239 if (ret & AST_FORMAT_AUDIO_MASK) 01240 return ret; 01241 01242 ast_debug(4, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec"); 01243 01244 return find_best ? ast_best_codec(formats) : 0; 01245 }
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 1507 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().
01508 { 01509 int len = 0; 01510 01511 /* XXX Still need speex, and lpc10 XXX */ 01512 switch(format) { 01513 case AST_FORMAT_G723_1: 01514 len = (samples / 240) * 20; 01515 break; 01516 case AST_FORMAT_ILBC: 01517 len = (samples / 240) * 50; 01518 break; 01519 case AST_FORMAT_GSM: 01520 len = (samples / 160) * 33; 01521 break; 01522 case AST_FORMAT_G729A: 01523 len = samples / 8; 01524 break; 01525 case AST_FORMAT_SLINEAR: 01526 case AST_FORMAT_SLINEAR16: 01527 len = samples * 2; 01528 break; 01529 case AST_FORMAT_ULAW: 01530 case AST_FORMAT_ALAW: 01531 case AST_FORMAT_TESTLAW: 01532 len = samples; 01533 break; 01534 case AST_FORMAT_G722: 01535 case AST_FORMAT_ADPCM: 01536 case AST_FORMAT_G726: 01537 case AST_FORMAT_G726_AAL2: 01538 len = samples / 2; 01539 break; 01540 case AST_FORMAT_SIREN7: 01541 /* 16,000 samples per second at 32kbps is 4,000 bytes per second */ 01542 len = samples / (16000 / 4000); 01543 break; 01544 case AST_FORMAT_SIREN14: 01545 /* 32,000 samples per second at 48kbps is 6,000 bytes per second */ 01546 len = (int) samples / ((float) 32000 / 6000); 01547 break; 01548 case AST_FORMAT_G719: 01549 /* 48,000 samples per second at 64kbps is 8,000 bytes per second */ 01550 len = (int) samples / ((float) 48000 / 8000); 01551 break; 01552 default: 01553 ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format)); 01554 } 01555 01556 return len; 01557 }
int ast_codec_get_samples | ( | struct ast_frame * | f | ) |
Returns the number of samples contained in the frame.
Definition at line 1445 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(), f, g723_samples(), LOG_WARNING, and speex_samples().
Referenced by ast_rtp_read(), isAnsweringMachine(), moh_generate(), schedule_delivery(), socket_process(), and socket_process_meta().
01446 { 01447 int samples = 0; 01448 char tmp[64]; 01449 01450 switch (f->subclass.codec) { 01451 case AST_FORMAT_SPEEX: 01452 samples = speex_samples(f->data.ptr, f->datalen); 01453 break; 01454 case AST_FORMAT_SPEEX16: 01455 samples = 2 * speex_samples(f->data.ptr, f->datalen); 01456 break; 01457 case AST_FORMAT_G723_1: 01458 samples = g723_samples(f->data.ptr, f->datalen); 01459 break; 01460 case AST_FORMAT_ILBC: 01461 samples = 240 * (f->datalen / 50); 01462 break; 01463 case AST_FORMAT_GSM: 01464 samples = 160 * (f->datalen / 33); 01465 break; 01466 case AST_FORMAT_G729A: 01467 samples = f->datalen * 8; 01468 break; 01469 case AST_FORMAT_SLINEAR: 01470 case AST_FORMAT_SLINEAR16: 01471 samples = f->datalen / 2; 01472 break; 01473 case AST_FORMAT_LPC10: 01474 /* assumes that the RTP packet contains one LPC10 frame */ 01475 samples = 22 * 8; 01476 samples += (((char *)(f->data.ptr))[7] & 0x1) * 8; 01477 break; 01478 case AST_FORMAT_ULAW: 01479 case AST_FORMAT_ALAW: 01480 case AST_FORMAT_TESTLAW: 01481 samples = f->datalen; 01482 break; 01483 case AST_FORMAT_G722: 01484 case AST_FORMAT_ADPCM: 01485 case AST_FORMAT_G726: 01486 case AST_FORMAT_G726_AAL2: 01487 samples = f->datalen * 2; 01488 break; 01489 case AST_FORMAT_SIREN7: 01490 /* 16,000 samples per second at 32kbps is 4,000 bytes per second */ 01491 samples = f->datalen * (16000 / 4000); 01492 break; 01493 case AST_FORMAT_SIREN14: 01494 /* 32,000 samples per second at 48kbps is 6,000 bytes per second */ 01495 samples = (int) f->datalen * ((float) 32000 / 6000); 01496 break; 01497 case AST_FORMAT_G719: 01498 /* 48,000 samples per second at 64kbps is 8,000 bytes per second */ 01499 samples = (int) f->datalen * ((float) 48000 / 8000); 01500 break; 01501 default: 01502 ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), f->subclass.codec)); 01503 } 01504 return samples; 01505 }
static int ast_codec_interp_len | ( | format_t | format | ) | [inline, static] |
Gets duration in ms of interpolation frame for a format.
Definition at line 749 of file frame.h.
References AST_FORMAT_ILBC.
Referenced by __get_from_jb(), and jb_get_and_deliver().
00750 { 00751 return (format == AST_FORMAT_ILBC) ? 30 : 20; 00752 }
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 1084 of file frame.c.
References ARRAY_LEN, ast_codec_pref_remove(), AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow().
01085 { 01086 int x, newindex = 0; 01087 01088 ast_codec_pref_remove(pref, format); 01089 01090 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01091 if (AST_FORMAT_LIST[x].bits == format) { 01092 newindex = x + 1; 01093 break; 01094 } 01095 } 01096 01097 if (newindex) { 01098 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01099 if (!pref->order[x]) { 01100 pref->order[x] = newindex; 01101 break; 01102 } 01103 } 01104 } 01105 01106 return x; 01107 }
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 987 of file frame.c.
References ast_codec_pref::order.
Referenced by check_access(), create_addr(), dump_prefs(), and socket_process().
00988 { 00989 int x, differential = (int) 'A', mem; 00990 char *from, *to; 00991 00992 if (right) { 00993 from = pref->order; 00994 to = buf; 00995 mem = size; 00996 } else { 00997 to = pref->order; 00998 from = buf; 00999 mem = sizeof(format_t) * 8; 01000 } 01001 01002 memset(to, 0, mem); 01003 for (x = 0; x < sizeof(format_t) * 8; x++) { 01004 if (!from[x]) 01005 break; 01006 to[x] = right ? (from[x] + differential) : (from[x] - differential); 01007 } 01008 }
struct ast_format_list ast_codec_pref_getsize | ( | struct ast_codec_pref * | pref, | |
format_t | format | |||
) |
Get packet size for codec.
Definition at line 1185 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_format_list::bits, ast_format_list::cur_ms, ast_format_list::def_ms, format, ast_format_list::inc_ms, ast_format_list::max_ms, and ast_format_list::min_ms.
Referenced by add_codec_to_sdp(), ast_rtp_write(), handle_open_receive_channel_ack_message(), skinny_set_rtp_peer(), and transmit_connect().
01186 { 01187 int x, idx = -1, framems = 0; 01188 struct ast_format_list fmt = { 0, }; 01189 01190 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01191 if (AST_FORMAT_LIST[x].bits == format) { 01192 fmt = AST_FORMAT_LIST[x]; 01193 idx = x; 01194 break; 01195 } 01196 } 01197 01198 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01199 if (pref->order[x] == (idx + 1)) { 01200 framems = pref->framing[x]; 01201 break; 01202 } 01203 } 01204 01205 /* size validation */ 01206 if (!framems) 01207 framems = AST_FORMAT_LIST[idx].def_ms; 01208 01209 if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */ 01210 framems -= framems % AST_FORMAT_LIST[idx].inc_ms; 01211 01212 if (framems < AST_FORMAT_LIST[idx].min_ms) 01213 framems = AST_FORMAT_LIST[idx].min_ms; 01214 01215 if (framems > AST_FORMAT_LIST[idx].max_ms) 01216 framems = AST_FORMAT_LIST[idx].max_ms; 01217 01218 fmt.cur_ms = framems; 01219 01220 return fmt; 01221 }
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 1046 of file frame.c.
References AST_FORMAT_LIST, 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().
01047 { 01048 int slot = 0; 01049 01050 if ((idx >= 0) && (idx < sizeof(pref->order))) { 01051 slot = pref->order[idx]; 01052 } 01053 01054 return slot ? AST_FORMAT_LIST[slot - 1].bits : 0; 01055 }
void ast_codec_pref_init | ( | struct ast_codec_pref * | pref | ) |
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 1110 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_codec_pref::framing, and ast_codec_pref::order.
Referenced by create_addr().
01111 { 01112 int x, newindex = 0; 01113 01114 /* First step is to get the codecs "index number" */ 01115 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01116 if (AST_FORMAT_LIST[x].bits == format) { 01117 newindex = x + 1; 01118 break; 01119 } 01120 } 01121 /* Done if its unknown */ 01122 if (!newindex) 01123 return; 01124 01125 /* Now find any existing occurrence, or the end */ 01126 for (x = 0; x < sizeof(format_t) * 8; x++) { 01127 if (!pref->order[x] || pref->order[x] == newindex) 01128 break; 01129 } 01130 01131 if (only_if_existing && !pref->order[x]) 01132 return; 01133 01134 /* Move down to make space to insert - either all the way to the end, 01135 or as far as the existing location (which will be overwritten) */ 01136 for (; x > 0; x--) { 01137 pref->order[x] = pref->order[x - 1]; 01138 pref->framing[x] = pref->framing[x - 1]; 01139 } 01140 01141 /* And insert the new entry */ 01142 pref->order[0] = newindex; 01143 pref->framing[0] = 0; /* ? */ 01144 }
void ast_codec_pref_remove | ( | struct ast_codec_pref * | pref, | |
format_t | format | |||
) |
Remove audio a codec from a preference list.
Definition at line 1058 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_codec_pref_append(), and ast_parse_allow_disallow().
01059 { 01060 struct ast_codec_pref oldorder; 01061 int x, y = 0; 01062 int slot; 01063 int size; 01064 01065 if (!pref->order[0]) 01066 return; 01067 01068 memcpy(&oldorder, pref, sizeof(oldorder)); 01069 memset(pref, 0, sizeof(*pref)); 01070 01071 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01072 slot = oldorder.order[x]; 01073 size = oldorder.framing[x]; 01074 if (! slot) 01075 break; 01076 if (AST_FORMAT_LIST[slot-1].bits != format) { 01077 pref->order[y] = slot; 01078 pref->framing[y++] = size; 01079 } 01080 } 01081 }
int ast_codec_pref_setsize | ( | struct ast_codec_pref * | pref, | |
format_t | format, | |||
int | framems | |||
) |
Set packet size for codec.
Definition at line 1147 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_format_list::def_ms, ast_codec_pref::framing, ast_format_list::inc_ms, ast_format_list::max_ms, ast_format_list::min_ms, and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow(), and process_sdp_a_audio().
01148 { 01149 int x, idx = -1; 01150 01151 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01152 if (AST_FORMAT_LIST[x].bits == format) { 01153 idx = x; 01154 break; 01155 } 01156 } 01157 01158 if (idx < 0) 01159 return -1; 01160 01161 /* size validation */ 01162 if (!framems) 01163 framems = AST_FORMAT_LIST[idx].def_ms; 01164 01165 if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */ 01166 framems -= framems % AST_FORMAT_LIST[idx].inc_ms; 01167 01168 if (framems < AST_FORMAT_LIST[idx].min_ms) 01169 framems = AST_FORMAT_LIST[idx].min_ms; 01170 01171 if (framems > AST_FORMAT_LIST[idx].max_ms) 01172 framems = AST_FORMAT_LIST[idx].max_ms; 01173 01174 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01175 if (pref->order[x] == (idx + 1)) { 01176 pref->framing[x] = framems; 01177 break; 01178 } 01179 } 01180 01181 return x; 01182 }
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 1010 of file frame.c.
References ast_codec_pref_index(), and ast_getformatname().
Referenced by dump_prefs(), and socket_process().
01011 { 01012 int x; 01013 format_t codec; 01014 size_t total_len, slen; 01015 char *formatname; 01016 01017 memset(buf, 0, size); 01018 total_len = size; 01019 buf[0] = '('; 01020 total_len--; 01021 for (x = 0; x < sizeof(format_t) * 8; x++) { 01022 if (total_len <= 0) 01023 break; 01024 if (!(codec = ast_codec_pref_index(pref,x))) 01025 break; 01026 if ((formatname = ast_getformatname(codec))) { 01027 slen = strlen(formatname); 01028 if (slen > total_len) 01029 break; 01030 strncat(buf, formatname, total_len - 1); /* safe */ 01031 total_len -= slen; 01032 } 01033 if (total_len && x < sizeof(format_t) * 8 - 1 && ast_codec_pref_index(pref, x + 1)) { 01034 strncat(buf, "|", total_len - 1); /* safe */ 01035 total_len--; 01036 } 01037 } 01038 if (total_len) { 01039 strncat(buf, ")", total_len - 1); /* safe */ 01040 total_len--; 01041 } 01042 01043 return size - total_len; 01044 }
static force_inline int ast_format_rate | ( | format_t | format | ) | [static] |
Get the sample rate for a given format.
Definition at line 776 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_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().
00777 { 00778 switch (format) { 00779 case AST_FORMAT_G722: 00780 case AST_FORMAT_SLINEAR16: 00781 case AST_FORMAT_SIREN7: 00782 case AST_FORMAT_SPEEX16: 00783 return 16000; 00784 case AST_FORMAT_SIREN14: 00785 return 32000; 00786 case AST_FORMAT_G719: 00787 return 48000; 00788 default: 00789 return 8000; 00790 } 00791 }
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 1559 of file frame.c.
References AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_slinear_saturated_divide(), ast_slinear_saturated_multiply(), and f.
Referenced by audiohook_read_frame_single(), audiohook_volume_callback(), conf_run(), and volume_callback().
01560 { 01561 int count; 01562 short *fdata = f->data.ptr; 01563 short adjust_value = abs(adjustment); 01564 01565 if ((f->frametype != AST_FRAME_VOICE) || (f->subclass.codec != AST_FORMAT_SLINEAR)) 01566 return -1; 01567 01568 if (!adjustment) 01569 return 0; 01570 01571 for (count = 0; count < f->samples; count++) { 01572 if (adjustment > 0) { 01573 ast_slinear_saturated_multiply(&fdata[count], &adjust_value); 01574 } else if (adjustment < 0) { 01575 ast_slinear_saturated_divide(&fdata[count], &adjust_value); 01576 } 01577 } 01578 01579 return 0; 01580 }
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 1604 of file frame.c.
References AST_LIST_NEXT, ast_frame::data, ast_frame::datalen, ast_frame::next, and ast_frame::ptr.
Referenced by ast_audiohook_write_frame(), and mute_callback().
01605 { 01606 struct ast_frame *next; 01607 01608 for (next = AST_LIST_NEXT(frame, frame_list); 01609 frame; 01610 frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) { 01611 memset(frame->data.ptr, 0, frame->datalen); 01612 } 01613 return 0; 01614 }
void ast_frame_dump | ( | const char * | name, | |
struct ast_frame * | f, | |||
char * | prefix | |||
) |
Dump a frame for debugging purposes
Definition at line 769 of file frame.c.
References AST_CONTROL_ANSWER, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_FLASH, AST_CONTROL_HANGUP, AST_CONTROL_HOLD, AST_CONTROL_OFFHOOK, AST_CONTROL_OPTION, AST_CONTROL_RADIO_KEY, AST_CONTROL_RADIO_UNKEY, AST_CONTROL_RING, AST_CONTROL_RINGING, AST_CONTROL_T38_PARAMETERS, AST_CONTROL_TAKEOFFHOOK, AST_CONTROL_UNHOLD, AST_CONTROL_WINK, ast_copy_string(), AST_FRAME_CONTROL, AST_FRAME_DTMF_BEGIN, AST_FRAME_DTMF_END, AST_FRAME_HTML, AST_FRAME_IAX, AST_FRAME_IMAGE, AST_FRAME_MODEM, AST_FRAME_NULL, AST_FRAME_TEXT, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_getformatname(), AST_HTML_BEGIN, AST_HTML_DATA, AST_HTML_END, AST_HTML_LDCOMPLETE, AST_HTML_LINKREJECT, AST_HTML_LINKURL, AST_HTML_NOSUPPORT, AST_HTML_UNLINK, AST_HTML_URL, AST_MODEM_T38, AST_MODEM_V150, ast_strlen_zero(), AST_T38_NEGOTIATED, AST_T38_REFUSED, AST_T38_REQUEST_NEGOTIATE, AST_T38_REQUEST_TERMINATE, AST_T38_TERMINATED, ast_verbose, COLOR_BLACK, COLOR_BRCYAN, COLOR_BRGREEN, COLOR_BRMAGENTA, COLOR_BRRED, COLOR_YELLOW, f, ast_control_t38_parameters::request_response, and term_color().
Referenced by __ast_read(), and ast_write().
00770 { 00771 const char noname[] = "unknown"; 00772 char ftype[40] = "Unknown Frametype"; 00773 char cft[80]; 00774 char subclass[40] = "Unknown Subclass"; 00775 char csub[80]; 00776 char moreinfo[40] = ""; 00777 char cn[60]; 00778 char cp[40]; 00779 char cmn[40]; 00780 const char *message = "Unknown"; 00781 00782 if (!name) 00783 name = noname; 00784 00785 00786 if (!f) { 00787 ast_verbose("%s [ %s (NULL) ] [%s]\n", 00788 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00789 term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00790 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00791 return; 00792 } 00793 /* XXX We should probably print one each of voice and video when the format changes XXX */ 00794 if (f->frametype == AST_FRAME_VOICE) 00795 return; 00796 if (f->frametype == AST_FRAME_VIDEO) 00797 return; 00798 switch(f->frametype) { 00799 case AST_FRAME_DTMF_BEGIN: 00800 strcpy(ftype, "DTMF Begin"); 00801 subclass[0] = f->subclass.integer; 00802 subclass[1] = '\0'; 00803 break; 00804 case AST_FRAME_DTMF_END: 00805 strcpy(ftype, "DTMF End"); 00806 subclass[0] = f->subclass.integer; 00807 subclass[1] = '\0'; 00808 break; 00809 case AST_FRAME_CONTROL: 00810 strcpy(ftype, "Control"); 00811 switch (f->subclass.integer) { 00812 case AST_CONTROL_HANGUP: 00813 strcpy(subclass, "Hangup"); 00814 break; 00815 case AST_CONTROL_RING: 00816 strcpy(subclass, "Ring"); 00817 break; 00818 case AST_CONTROL_RINGING: 00819 strcpy(subclass, "Ringing"); 00820 break; 00821 case AST_CONTROL_ANSWER: 00822 strcpy(subclass, "Answer"); 00823 break; 00824 case AST_CONTROL_BUSY: 00825 strcpy(subclass, "Busy"); 00826 break; 00827 case AST_CONTROL_TAKEOFFHOOK: 00828 strcpy(subclass, "Take Off Hook"); 00829 break; 00830 case AST_CONTROL_OFFHOOK: 00831 strcpy(subclass, "Line Off Hook"); 00832 break; 00833 case AST_CONTROL_CONGESTION: 00834 strcpy(subclass, "Congestion"); 00835 break; 00836 case AST_CONTROL_FLASH: 00837 strcpy(subclass, "Flash"); 00838 break; 00839 case AST_CONTROL_WINK: 00840 strcpy(subclass, "Wink"); 00841 break; 00842 case AST_CONTROL_OPTION: 00843 strcpy(subclass, "Option"); 00844 break; 00845 case AST_CONTROL_RADIO_KEY: 00846 strcpy(subclass, "Key Radio"); 00847 break; 00848 case AST_CONTROL_RADIO_UNKEY: 00849 strcpy(subclass, "Unkey Radio"); 00850 break; 00851 case AST_CONTROL_HOLD: 00852 strcpy(subclass, "Hold"); 00853 break; 00854 case AST_CONTROL_UNHOLD: 00855 strcpy(subclass, "Unhold"); 00856 break; 00857 case AST_CONTROL_T38_PARAMETERS: 00858 if (f->datalen != sizeof(struct ast_control_t38_parameters)) { 00859 message = "Invalid"; 00860 } else { 00861 struct ast_control_t38_parameters *parameters = f->data.ptr; 00862 enum ast_control_t38 state = parameters->request_response; 00863 if (state == AST_T38_REQUEST_NEGOTIATE) 00864 message = "Negotiation Requested"; 00865 else if (state == AST_T38_REQUEST_TERMINATE) 00866 message = "Negotiation Request Terminated"; 00867 else if (state == AST_T38_NEGOTIATED) 00868 message = "Negotiated"; 00869 else if (state == AST_T38_TERMINATED) 00870 message = "Terminated"; 00871 else if (state == AST_T38_REFUSED) 00872 message = "Refused"; 00873 } 00874 snprintf(subclass, sizeof(subclass), "T38_Parameters/%s", message); 00875 break; 00876 case -1: 00877 strcpy(subclass, "Stop generators"); 00878 break; 00879 default: 00880 snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass.integer); 00881 } 00882 break; 00883 case AST_FRAME_NULL: 00884 strcpy(ftype, "Null Frame"); 00885 strcpy(subclass, "N/A"); 00886 break; 00887 case AST_FRAME_IAX: 00888 /* Should never happen */ 00889 strcpy(ftype, "IAX Specific"); 00890 snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass.integer); 00891 break; 00892 case AST_FRAME_TEXT: 00893 strcpy(ftype, "Text"); 00894 strcpy(subclass, "N/A"); 00895 ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); 00896 break; 00897 case AST_FRAME_IMAGE: 00898 strcpy(ftype, "Image"); 00899 snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass.codec)); 00900 break; 00901 case AST_FRAME_HTML: 00902 strcpy(ftype, "HTML"); 00903 switch (f->subclass.integer) { 00904 case AST_HTML_URL: 00905 strcpy(subclass, "URL"); 00906 ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); 00907 break; 00908 case AST_HTML_DATA: 00909 strcpy(subclass, "Data"); 00910 break; 00911 case AST_HTML_BEGIN: 00912 strcpy(subclass, "Begin"); 00913 break; 00914 case AST_HTML_END: 00915 strcpy(subclass, "End"); 00916 break; 00917 case AST_HTML_LDCOMPLETE: 00918 strcpy(subclass, "Load Complete"); 00919 break; 00920 case AST_HTML_NOSUPPORT: 00921 strcpy(subclass, "No Support"); 00922 break; 00923 case AST_HTML_LINKURL: 00924 strcpy(subclass, "Link URL"); 00925 ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); 00926 break; 00927 case AST_HTML_UNLINK: 00928 strcpy(subclass, "Unlink"); 00929 break; 00930 case AST_HTML_LINKREJECT: 00931 strcpy(subclass, "Link Reject"); 00932 break; 00933 default: 00934 snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass.integer); 00935 break; 00936 } 00937 break; 00938 case AST_FRAME_MODEM: 00939 strcpy(ftype, "Modem"); 00940 switch (f->subclass.integer) { 00941 case AST_MODEM_T38: 00942 strcpy(subclass, "T.38"); 00943 break; 00944 case AST_MODEM_V150: 00945 strcpy(subclass, "V.150"); 00946 break; 00947 default: 00948 snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass.integer); 00949 break; 00950 } 00951 break; 00952 default: 00953 snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype); 00954 } 00955 if (!ast_strlen_zero(moreinfo)) 00956 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n", 00957 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00958 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00959 f->frametype, 00960 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00961 f->subclass.integer, 00962 term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)), 00963 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00964 else 00965 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%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(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00972 }
struct ast_frame* ast_frame_enqueue | ( | struct ast_frame * | head, | |
struct ast_frame * | f, | |||
int | maxlen, | |||
int | dupe | |||
) |
Appends a frame to the end of a list of frames, truncating the maximum length of the list.
void ast_frame_free | ( | struct ast_frame * | fr, | |
int | cache | |||
) |
Requests a frame to be allocated Frees a frame or list of frames.
fr | Frame to free, or head of list to free | |
cache | Whether to consider this frame for frame caching |
Definition at line 371 of file frame.c.
References __frame_free(), AST_LIST_NEXT, and ast_frame::next.
Referenced by mixmonitor_thread().
00372 { 00373 struct ast_frame *next; 00374 00375 for (next = AST_LIST_NEXT(frame, frame_list); 00376 frame; 00377 frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) { 00378 __frame_free(frame, cache); 00379 } 00380 }
Sums two frames of audio samples.
f1 | The first frame (which will contain the result) | |
f2 | The second frame |
Definition at line 1582 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.
01583 { 01584 int count; 01585 short *data1, *data2; 01586 01587 if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass.codec != AST_FORMAT_SLINEAR)) 01588 return -1; 01589 01590 if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass.codec != AST_FORMAT_SLINEAR)) 01591 return -1; 01592 01593 if (f1->samples != f2->samples) 01594 return -1; 01595 01596 for (count = 0, data1 = f1->data.ptr, data2 = f2->data.ptr; 01597 count < f1->samples; 01598 count++, data1++, data2++) 01599 ast_slinear_saturated_add(data1, data2); 01600 01601 return 0; 01602 }
Copies a frame.
fr | frame to copy Duplicates a frame -- should only rarely be used, typically frisolate is good enough |
Definition at line 470 of file frame.c.
References ast_calloc_cache, ast_copy_flags, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, AST_LIST_REMOVE_CURRENT, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, AST_MALLOCD_HDR, ast_threadstorage_get(), ast_frame_subclass::codec, ast_frame::data, ast_frame::datalen, ast_frame::delivery, f, frame_cache, frames, ast_frame::frametype, len(), ast_frame::len, ast_frame::mallocd, ast_frame::mallocd_hdr_len, ast_frame::offset, ast_frame::ptr, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, ast_frame::ts, and ast_frame::uint32.
Referenced by __ast_queue_frame(), ast_frisolate(), ast_indicate_data(), ast_jb_put(), ast_rtp_write(), ast_slinfactory_feed(), audiohook_read_frame_single(), autoservice_run(), multicast_rtp_write(), process_dtmf_rfc2833(), recordthread(), and rpt().
00471 { 00472 struct ast_frame *out = NULL; 00473 int len, srclen = 0; 00474 void *buf = NULL; 00475 00476 #if !defined(LOW_MEMORY) 00477 struct ast_frame_cache *frames; 00478 #endif 00479 00480 /* Start with standard stuff */ 00481 len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00482 /* If we have a source, add space for it */ 00483 /* 00484 * XXX Watch out here - if we receive a src which is not terminated 00485 * properly, we can be easily attacked. Should limit the size we deal with. 00486 */ 00487 if (f->src) 00488 srclen = strlen(f->src); 00489 if (srclen > 0) 00490 len += srclen + 1; 00491 00492 #if !defined(LOW_MEMORY) 00493 if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) { 00494 AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) { 00495 if (out->mallocd_hdr_len >= len) { 00496 size_t mallocd_len = out->mallocd_hdr_len; 00497 00498 AST_LIST_REMOVE_CURRENT(frame_list); 00499 memset(out, 0, sizeof(*out)); 00500 out->mallocd_hdr_len = mallocd_len; 00501 buf = out; 00502 frames->size--; 00503 break; 00504 } 00505 } 00506 AST_LIST_TRAVERSE_SAFE_END; 00507 } 00508 #endif 00509 00510 if (!buf) { 00511 if (!(buf = ast_calloc_cache(1, len))) 00512 return NULL; 00513 out = buf; 00514 out->mallocd_hdr_len = len; 00515 } 00516 00517 out->frametype = f->frametype; 00518 out->subclass.codec = f->subclass.codec; 00519 out->datalen = f->datalen; 00520 out->samples = f->samples; 00521 out->delivery = f->delivery; 00522 /* Set us as having malloc'd header only, so it will eventually 00523 get freed. */ 00524 out->mallocd = AST_MALLOCD_HDR; 00525 out->offset = AST_FRIENDLY_OFFSET; 00526 if (out->datalen) { 00527 out->data.ptr = buf + sizeof(*out) + AST_FRIENDLY_OFFSET; 00528 memcpy(out->data.ptr, f->data.ptr, out->datalen); 00529 } else { 00530 out->data.uint32 = f->data.uint32; 00531 } 00532 if (srclen > 0) { 00533 /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */ 00534 char *src; 00535 out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00536 src = (char *) out->src; 00537 /* Must have space since we allocated for it */ 00538 strcpy(src, f->src); 00539 } 00540 ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO); 00541 out->ts = f->ts; 00542 out->len = f->len; 00543 out->seqno = f->seqno; 00544 return out; 00545 }
Makes a frame independent of any static storage.
fr | frame to act upon Take a frame, and if it's not been malloc'd, make a malloc'd copy and if the data hasn't been malloced then make the data malloc'd. If you need to store frames, say for queueing, then you should call this function. |
Definition at line 387 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_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().
00388 { 00389 struct ast_frame *out; 00390 void *newdata; 00391 00392 /* if none of the existing frame is malloc'd, let ast_frdup() do it 00393 since it is more efficient 00394 */ 00395 if (fr->mallocd == 0) { 00396 return ast_frdup(fr); 00397 } 00398 00399 /* if everything is already malloc'd, we are done */ 00400 if ((fr->mallocd & (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) == 00401 (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) { 00402 return fr; 00403 } 00404 00405 if (!(fr->mallocd & AST_MALLOCD_HDR)) { 00406 /* Allocate a new header if needed */ 00407 if (!(out = ast_frame_header_new())) { 00408 return NULL; 00409 } 00410 out->frametype = fr->frametype; 00411 out->subclass.codec = fr->subclass.codec; 00412 out->datalen = fr->datalen; 00413 out->samples = fr->samples; 00414 out->offset = fr->offset; 00415 /* Copy the timing data */ 00416 ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO); 00417 if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) { 00418 out->ts = fr->ts; 00419 out->len = fr->len; 00420 out->seqno = fr->seqno; 00421 } 00422 } else { 00423 out = fr; 00424 } 00425 00426 if (!(fr->mallocd & AST_MALLOCD_SRC) && fr->src) { 00427 if (!(out->src = ast_strdup(fr->src))) { 00428 if (out != fr) { 00429 ast_free(out); 00430 } 00431 return NULL; 00432 } 00433 } else { 00434 out->src = fr->src; 00435 fr->src = NULL; 00436 fr->mallocd &= ~AST_MALLOCD_SRC; 00437 } 00438 00439 if (!(fr->mallocd & AST_MALLOCD_DATA)) { 00440 if (!fr->datalen) { 00441 out->data.uint32 = fr->data.uint32; 00442 out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC; 00443 return out; 00444 } 00445 if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) { 00446 if (out->src != fr->src) { 00447 ast_free((void *) out->src); 00448 } 00449 if (out != fr) { 00450 ast_free(out); 00451 } 00452 return NULL; 00453 } 00454 newdata += AST_FRIENDLY_OFFSET; 00455 out->offset = AST_FRIENDLY_OFFSET; 00456 out->datalen = fr->datalen; 00457 memcpy(newdata, fr->data.ptr, fr->datalen); 00458 out->data.ptr = newdata; 00459 } else { 00460 out->data = fr->data; 00461 memset(&fr->data, 0, sizeof(fr->data)); 00462 fr->mallocd &= ~AST_MALLOCD_DATA; 00463 } 00464 00465 out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA; 00466 00467 return out; 00468 }
struct ast_format_list* ast_get_format_list | ( | size_t * | size | ) |
Definition at line 563 of file frame.c.
References ARRAY_LEN, and AST_FORMAT_LIST.
Referenced by ast_data_add_codecs(), complete_trans_path_choice(), and handle_cli_core_show_translation().
00564 { 00565 *size = ARRAY_LEN(AST_FORMAT_LIST); 00566 return AST_FORMAT_LIST; 00567 }
struct ast_format_list* ast_get_format_list_index | ( | int | index | ) |
Definition at line 558 of file frame.c.
References AST_FORMAT_LIST.
00559 { 00560 return &AST_FORMAT_LIST[idx]; 00561 }
format_t ast_getformatbyname | ( | const char * | name | ) |
Gets a format from a name.
name | string of format |
Definition at line 632 of file frame.c.
References ARRAY_LEN, ast_expand_codec_alias(), AST_FORMAT_LIST, ast_format_list::bits, and format.
Referenced by ast_parse_allow_disallow(), iax_template_parse(), load_moh_classes(), local_ast_moh_start(), and try_suggested_sip_codec().
00633 { 00634 int x, all; 00635 format_t format = 0; 00636 00637 all = strcasecmp(name, "all") ? 0 : 1; 00638 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00639 if (all || 00640 !strcasecmp(AST_FORMAT_LIST[x].name,name) || 00641 !strcasecmp(AST_FORMAT_LIST[x].name, ast_expand_codec_alias(name))) { 00642 format |= AST_FORMAT_LIST[x].bits; 00643 if (!all) 00644 break; 00645 } 00646 } 00647 00648 return format; 00649 }
char* ast_getformatname | ( | format_t | format | ) |
Get the name of a format.
format | id of format |
Definition at line 569 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_format_list::bits, and ast_format_list::name.
Referenced by __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_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_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(), 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().
00570 { 00571 int x; 00572 char *ret = "unknown"; 00573 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00574 if (AST_FORMAT_LIST[x].bits == format) { 00575 ret = AST_FORMAT_LIST[x].name; 00576 break; 00577 } 00578 } 00579 return ret; 00580 }
char* ast_getformatname_multiple | ( | char * | buf, | |
size_t | size, | |||
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 582 of file frame.c.
References ARRAY_LEN, ast_copy_string(), AST_FORMAT_LIST, ast_format_list::bits, len(), and name.
Referenced by __ast_read(), _sip_show_peer(), _skinny_show_device(), _skinny_show_line(), add_sdp(), alsa_request(), ast_best_codec(), ast_bridge_new(), ast_codec_get_samples(), ast_request(), ast_streamfile(), bridge_make_compatible(), console_request(), find_best_technology(), 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(), 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(), smart_bridge_operation(), socket_process(), start_rtp(), unistim_new(), unistim_request(), and unistim_write().
00583 { 00584 int x; 00585 unsigned len; 00586 char *start, *end = buf; 00587 00588 if (!size) 00589 return buf; 00590 snprintf(end, size, "0x%llx (", (unsigned long long) format); 00591 len = strlen(end); 00592 end += len; 00593 size -= len; 00594 start = end; 00595 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00596 if (AST_FORMAT_LIST[x].bits & format) { 00597 snprintf(end, size, "%s|", AST_FORMAT_LIST[x].name); 00598 len = strlen(end); 00599 end += len; 00600 size -= len; 00601 } 00602 } 00603 if (start == end) 00604 ast_copy_string(start, "nothing)", size); 00605 else if (size > 1) 00606 *(end - 1) = ')'; 00607 return buf; 00608 }
int ast_parse_allow_disallow | ( | struct ast_codec_pref * | pref, | |
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 1247 of file frame.c.
References ast_codec_pref_append(), ast_codec_pref_remove(), ast_codec_pref_setsize(), ast_debug, ast_getformatbyname(), ast_log(), ast_strdupa, format, LOG_WARNING, parse(), and strsep().
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(), set_config(), skinny_unregister(), and update_common_options().
01248 { 01249 int errors = 0, framems = 0; 01250 char *parse = NULL, *this = NULL, *psize = NULL; 01251 format_t format = 0; 01252 01253 parse = ast_strdupa(list); 01254 while ((this = strsep(&parse, ","))) { 01255 framems = 0; 01256 if ((psize = strrchr(this, ':'))) { 01257 *psize++ = '\0'; 01258 ast_debug(1, "Packetization for codec: %s is %s\n", this, psize); 01259 framems = atoi(psize); 01260 if (framems < 0) { 01261 framems = 0; 01262 errors++; 01263 ast_log(LOG_WARNING, "Bad packetization value for codec %s\n", this); 01264 } 01265 } 01266 if (!(format = ast_getformatbyname(this))) { 01267 ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this); 01268 errors++; 01269 continue; 01270 } 01271 01272 if (mask) { 01273 if (allowing) 01274 *mask |= format; 01275 else 01276 *mask &= ~format; 01277 } 01278 01279 /* Set up a preference list for audio. Do not include video in preferences 01280 since we can not transcode video and have to use whatever is offered 01281 */ 01282 if (pref && (format & AST_FORMAT_AUDIO_MASK)) { 01283 if (strcasecmp(this, "all")) { 01284 if (allowing) { 01285 ast_codec_pref_append(pref, format); 01286 ast_codec_pref_setsize(pref, format, framems); 01287 } 01288 else 01289 ast_codec_pref_remove(pref, format); 01290 } else if (!allowing) { 01291 memset(pref, 0, sizeof(*pref)); 01292 } 01293 } 01294 } 01295 return errors; 01296 }
void ast_smoother_free | ( | struct ast_smoother * | s | ) |
Definition at line 290 of file frame.c.
References ast_free.
Referenced by ast_rtp_destroy(), ast_rtp_write(), destroy_session(), and generic_fax_exec().
00291 { 00292 ast_free(s); 00293 }
int ast_smoother_get_flags | ( | struct ast_smoother * | smoother | ) |
struct ast_smoother* ast_smoother_new | ( | int | bytes | ) |
Definition at line 179 of file frame.c.
References ast_malloc, and ast_smoother_reset().
Referenced by ast_rtp_write(), and generic_fax_exec().
00180 { 00181 struct ast_smoother *s; 00182 if (size < 1) 00183 return NULL; 00184 if ((s = ast_malloc(sizeof(*s)))) 00185 ast_smoother_reset(s, size); 00186 return s; 00187 }
struct ast_frame* ast_smoother_read | ( | struct ast_smoother * | s | ) |
Definition at line 240 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_frame::data, ast_smoother::data, ast_frame::datalen, ast_frame::delivery, ast_smoother::delivery, ast_smoother::f, ast_smoother::flags, ast_smoother::format, ast_smoother::framedata, ast_frame::frametype, len(), ast_smoother::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().
00241 { 00242 struct ast_frame *opt; 00243 int len; 00244 00245 /* IF we have an optimization frame, send it */ 00246 if (s->opt) { 00247 if (s->opt->offset < AST_FRIENDLY_OFFSET) 00248 ast_log(LOG_WARNING, "Returning a frame of inappropriate offset (%d).\n", 00249 s->opt->offset); 00250 opt = s->opt; 00251 s->opt = NULL; 00252 return opt; 00253 } 00254 00255 /* Make sure we have enough data */ 00256 if (s->len < s->size) { 00257 /* Or, if this is a G.729 frame with VAD on it, send it immediately anyway */ 00258 if (!((s->flags & AST_SMOOTHER_FLAG_G729) && (s->len % 10))) 00259 return NULL; 00260 } 00261 len = s->size; 00262 if (len > s->len) 00263 len = s->len; 00264 /* Make frame */ 00265 s->f.frametype = AST_FRAME_VOICE; 00266 s->f.subclass.codec = s->format; 00267 s->f.data.ptr = s->framedata + AST_FRIENDLY_OFFSET; 00268 s->f.offset = AST_FRIENDLY_OFFSET; 00269 s->f.datalen = len; 00270 /* Samples will be improper given VAD, but with VAD the concept really doesn't even exist */ 00271 s->f.samples = len * s->samplesperbyte; /* XXX rounding */ 00272 s->f.delivery = s->delivery; 00273 /* Fill Data */ 00274 memcpy(s->f.data.ptr, s->data, len); 00275 s->len -= len; 00276 /* Move remaining data to the front if applicable */ 00277 if (s->len) { 00278 /* In principle this should all be fine because if we are sending 00279 G.729 VAD, the next timestamp will take over anyawy */ 00280 memmove(s->data, s->data + len, s->len); 00281 if (!ast_tvzero(s->delivery)) { 00282 /* If we have delivery time, increment it, otherwise, leave it at 0 */ 00283 s->delivery = ast_tvadd(s->delivery, ast_samp2tv(s->f.samples, ast_format_rate(s->format))); 00284 } 00285 } 00286 /* Return frame */ 00287 return &s->f; 00288 }
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 157 of file frame.c.
References ast_smoother::opt, ast_smoother::opt_needs_swap, ast_smoother::size, and smoother_frame_feed().
00158 { 00159 /* if there is no change, then nothing to do */ 00160 if (s->size == bytes) { 00161 return; 00162 } 00163 /* set the new desired output size */ 00164 s->size = bytes; 00165 /* if there is no 'optimized' frame in the smoother, 00166 * then there is nothing left to do 00167 */ 00168 if (!s->opt) { 00169 return; 00170 } 00171 /* there is an 'optimized' frame here at the old size, 00172 * but it must now be put into the buffer so the data 00173 * can be extracted at the new size 00174 */ 00175 smoother_frame_feed(s, s->opt, s->opt_needs_swap); 00176 s->opt = NULL; 00177 }
void ast_smoother_reset | ( | struct ast_smoother * | s, | |
int | bytes | |||
) |
void ast_smoother_set_flags | ( | struct ast_smoother * | smoother, | |
int | flags | |||
) |
Definition at line 194 of file frame.c.
References ast_smoother::flags.
Referenced by ast_rtp_write().
00195 { 00196 s->flags = flags; 00197 }
int ast_smoother_test_flag | ( | struct ast_smoother * | s, | |
int | flag | |||
) |
Definition at line 199 of file frame.c.
References ast_smoother::flags.
Referenced by ast_rtp_write().
00200 { 00201 return (s->flags & flag); 00202 }
void ast_swapcopy_samples | ( | void * | dst, | |
const void * | src, | |||
int | samples | |||
) |
Definition at line 547 of file frame.c.
Referenced by __ast_smoother_feed(), iax_frame_wrap(), phone_write_buf(), and smoother_frame_feed().
00548 { 00549 int i; 00550 unsigned short *dst_s = dst; 00551 const unsigned short *src_s = src; 00552 00553 for (i = 0; i < samples; i++) 00554 dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8); 00555 }
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 127 of file frame.c.
Referenced by __analog_handle_event(), __ast_channel_masquerade(), __ast_read(), __oh323_rtp_create(), __oh323_update_info(), agent_new(), agent_read(), 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_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().