Asterisk internal frame definitions. More...
#include <sys/time.h>
#include "asterisk/endian.h"
#include "asterisk/linkedlists.h"
#include "asterisk/endian.h"
#include "asterisk/linkedlists.h"
Go to the source code of this file.
Data Structures | |
struct | ast_codec_pref |
struct | ast_control_t38_parameters |
struct | ast_format_list |
Definition of supported media formats (codecs) More... | |
struct | ast_frame |
Data structure associated with a single frame of data. More... | |
struct | ast_option_header |
struct | oprmode |
Defines | |
#define | AST_FORMAT_ADPCM (1 << 5) |
#define | AST_FORMAT_ALAW (1 << 3) |
#define | AST_FORMAT_AUDIO_MASK ((1 << 16)-1) |
#define | AST_FORMAT_G722 (1 << 12) |
#define | AST_FORMAT_G723_1 (1 << 0) |
#define | AST_FORMAT_G726 (1 << 11) |
#define | AST_FORMAT_G726_AAL2 (1 << 4) |
#define | AST_FORMAT_G729A (1 << 8) |
#define | AST_FORMAT_GSM (1 << 1) |
#define | AST_FORMAT_H261 (1 << 18) |
#define | AST_FORMAT_H263 (1 << 19) |
#define | AST_FORMAT_H263_PLUS (1 << 20) |
#define | AST_FORMAT_H264 (1 << 21) |
#define | AST_FORMAT_ILBC (1 << 10) |
#define | AST_FORMAT_JPEG (1 << 16) |
#define | AST_FORMAT_LPC10 (1 << 7) |
#define | AST_FORMAT_MAX_TEXT (1 << 28)) |
#define | AST_FORMAT_MP4_VIDEO (1 << 22) |
#define | AST_FORMAT_PNG (1 << 17) |
#define | AST_FORMAT_SIREN14 (1 << 14) |
#define | AST_FORMAT_SIREN7 (1 << 13) |
#define | AST_FORMAT_SLINEAR (1 << 6) |
#define | AST_FORMAT_SLINEAR16 (1 << 15) |
#define | AST_FORMAT_SPEEX (1 << 9) |
#define | AST_FORMAT_T140 (1 << 27) |
#define | AST_FORMAT_T140RED (1 << 26) |
#define | AST_FORMAT_TEXT_MASK (((1 << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK)) |
#define | AST_FORMAT_ULAW (1 << 2) |
#define | AST_FORMAT_VIDEO_MASK (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) |
#define | ast_frame_byteswap_be(fr) do { ; } while(0) |
#define | ast_frame_byteswap_le(fr) do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data.ptr, __f->data.ptr, __f->samples); } while(0) |
#define | AST_FRAME_DTMF AST_FRAME_DTMF_END |
#define | AST_FRAME_SET_BUFFER(fr, _base, _ofs, _datalen) |
#define | ast_frfree(fr) ast_frame_free(fr, 1) |
#define | AST_FRIENDLY_OFFSET 64 |
Offset into a frame's data buffer. | |
#define | AST_HTML_BEGIN 4 |
#define | AST_HTML_DATA 2 |
#define | AST_HTML_END 8 |
#define | AST_HTML_LDCOMPLETE 16 |
#define | AST_HTML_LINKREJECT 20 |
#define | AST_HTML_LINKURL 18 |
#define | AST_HTML_NOSUPPORT 17 |
#define | AST_HTML_UNLINK 19 |
#define | AST_HTML_URL 1 |
#define | AST_MALLOCD_DATA (1 << 1) |
#define | AST_MALLOCD_HDR (1 << 0) |
#define | AST_MALLOCD_SRC (1 << 2) |
#define | AST_MIN_OFFSET 32 |
#define | AST_MODEM_T38 1 |
#define | AST_MODEM_V150 2 |
#define | AST_OPTION_AUDIO_MODE 4 |
#define | AST_OPTION_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_ECHOCAN 8 |
#define | AST_OPTION_FLAG_ACCEPT 1 |
#define | AST_OPTION_FLAG_ANSWER 5 |
#define | AST_OPTION_FLAG_QUERY 4 |
#define | AST_OPTION_FLAG_REJECT 2 |
#define | AST_OPTION_FLAG_REQUEST 0 |
#define | AST_OPTION_FLAG_WTF 6 |
#define | AST_OPTION_OPRMODE 7 |
#define | AST_OPTION_RELAXDTMF 3 |
#define | AST_OPTION_RXGAIN 6 |
#define | AST_OPTION_T38_STATE 10 |
#define | AST_OPTION_TDD 2 |
#define | AST_OPTION_TONE_VERIFY 1 |
#define | AST_OPTION_TXGAIN 5 |
#define | AST_SMOOTHER_FLAG_BE (1 << 1) |
#define | AST_SMOOTHER_FLAG_G729 (1 << 0) |
Enumerations | |
enum | { AST_FRFLAG_HAS_TIMING_INFO = (1 << 0), AST_FRFLAG_FROM_TRANSLATOR = (1 << 1), AST_FRFLAG_FROM_DSP = (1 << 2) } |
enum | ast_control_frame_type { AST_CONTROL_HANGUP = 1, AST_CONTROL_RING = 2, AST_CONTROL_RINGING = 3, AST_CONTROL_ANSWER = 4, AST_CONTROL_BUSY = 5, AST_CONTROL_TAKEOFFHOOK = 6, AST_CONTROL_OFFHOOK = 7, AST_CONTROL_CONGESTION = 8, AST_CONTROL_FLASH = 9, AST_CONTROL_WINK = 10, AST_CONTROL_OPTION = 11, AST_CONTROL_RADIO_KEY = 12, AST_CONTROL_RADIO_UNKEY = 13, AST_CONTROL_PROGRESS = 14, AST_CONTROL_PROCEEDING = 15, AST_CONTROL_HOLD = 16, AST_CONTROL_UNHOLD = 17, AST_CONTROL_VIDUPDATE = 18, _XXX_AST_CONTROL_T38 = 19, AST_CONTROL_SRCUPDATE = 20, AST_CONTROL_T38_PARAMETERS = 24, AST_CONTROL_SRCCHANGE = 25, 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_frame_type { AST_FRAME_DTMF_END = 1, AST_FRAME_VOICE, AST_FRAME_VIDEO, AST_FRAME_CONTROL, AST_FRAME_NULL, AST_FRAME_IAX, AST_FRAME_TEXT, AST_FRAME_IMAGE, AST_FRAME_HTML, AST_FRAME_CNG, AST_FRAME_MODEM, AST_FRAME_DTMF_BEGIN } |
Frame types. More... | |
Functions | |
char * | ast_codec2str (int codec) |
Get a name from a format Gets a name from a format. | |
int | ast_codec_choose (struct ast_codec_pref *pref, int formats, int find_best) |
Select the best audio format according to preference list from supplied options. If "find_best" is non-zero then if nothing is found, the "Best" format of the format list is selected, otherwise 0 is returned. | |
int | ast_codec_get_len (int format, int samples) |
Returns the number of bytes for the number of samples of the given format. | |
int | ast_codec_get_samples (struct ast_frame *f) |
Returns the number of samples contained in the frame. | |
static int | ast_codec_interp_len (int format) |
Gets duration in ms of interpolation frame for a format. | |
int | ast_codec_pref_append (struct ast_codec_pref *pref, int format) |
Append a audio codec to a preference list, removing it first if it was already there. | |
void | ast_codec_pref_convert (struct ast_codec_pref *pref, char *buf, size_t size, int right) |
Shift an audio codec preference list up or down 65 bytes so that it becomes an ASCII string. | |
struct ast_format_list | ast_codec_pref_getsize (struct ast_codec_pref *pref, int format) |
Get packet size for codec. | |
int | ast_codec_pref_index (struct ast_codec_pref *pref, int index) |
Codec located at a particular place in the preference index. | |
void | ast_codec_pref_init (struct ast_codec_pref *pref) |
Initialize an audio codec preference to "no preference". | |
void | ast_codec_pref_prepend (struct ast_codec_pref *pref, int format, int only_if_existing) |
Prepend an audio codec to a preference list, removing it first if it was already there. | |
void | ast_codec_pref_remove (struct ast_codec_pref *pref, int format) |
Remove audio a codec from a preference list. | |
int | ast_codec_pref_setsize (struct ast_codec_pref *pref, int format, int framems) |
Set packet size for codec. | |
int | ast_codec_pref_string (struct ast_codec_pref *pref, char *buf, size_t size) |
Dump audio codec preference list into a string. | |
static force_inline int | ast_format_rate (int format) |
Get the sample rate for a given format. | |
int | ast_frame_adjust_volume (struct ast_frame *f, int adjustment) |
Adjusts the volume of the audio samples contained in a frame. | |
void | ast_frame_dump (const char *name, struct ast_frame *f, char *prefix) |
struct ast_frame * | ast_frame_enqueue (struct ast_frame *head, struct ast_frame *f, int maxlen, int dupe) |
Appends a frame to the end of a list of frames, truncating the maximum length of the list. | |
void | ast_frame_free (struct ast_frame *fr, int cache) |
Requests a frame to be allocated. | |
int | ast_frame_slinear_sum (struct ast_frame *f1, struct ast_frame *f2) |
Sums two frames of audio samples. | |
struct ast_frame * | ast_frdup (const struct ast_frame *fr) |
Copies a frame. | |
struct ast_frame * | ast_frisolate (struct ast_frame *fr) |
Makes a frame independent of any static storage. | |
struct ast_format_list * | ast_get_format_list (size_t *size) |
struct ast_format_list * | ast_get_format_list_index (int index) |
int | ast_getformatbyname (const char *name) |
Gets a format from a name. | |
char * | ast_getformatname (int format) |
Get the name of a format. | |
char * | ast_getformatname_multiple (char *buf, size_t size, int format) |
Get the names of a set of formats. | |
int | ast_parse_allow_disallow (struct ast_codec_pref *pref, int *mask, const char *list, int allowing) |
Parse an "allow" or "deny" line in a channel or device configuration and update the capabilities mask and pref if provided. Video codecs are not added to codec preference lists, since we can not transcode. | |
void | ast_swapcopy_samples (void *dst, const void *src, int samples) |
Variables | |
struct ast_frame | ast_null_frame |
AST_Smoother | |
#define | ast_smoother_feed(s, f) __ast_smoother_feed(s, f, 0) |
#define | ast_smoother_feed_be(s, f) __ast_smoother_feed(s, f, 0) |
#define | ast_smoother_feed_le(s, f) __ast_smoother_feed(s, f, 1) |
struct ast_smoother * | ast_smoother_new (int bytes) |
void | ast_smoother_set_flags (struct ast_smoother *smoother, int flags) |
int | ast_smoother_get_flags (struct ast_smoother *smoother) |
int | ast_smoother_test_flag (struct ast_smoother *s, int flag) |
void | ast_smoother_free (struct ast_smoother *s) |
void | ast_smoother_reset (struct ast_smoother *s, int bytes) |
void | ast_smoother_reconfigure (struct ast_smoother *s, int bytes) |
Reconfigure an existing smoother to output a different number of bytes per frame. | |
int | __ast_smoother_feed (struct ast_smoother *s, struct ast_frame *f, int swap) |
struct ast_frame * | ast_smoother_read (struct ast_smoother *s) |
Asterisk internal frame definitions.
Definition in file frame.h.
#define AST_FORMAT_ADPCM (1 << 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 (1 << 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 ((1 << 16)-1) |
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_codec_choose(), ast_filehelper(), ast_openstream_full(), ast_openvstream(), ast_parse_allow_disallow(), ast_playstream(), ast_request(), ast_rtp_read(), ast_translate_available_formats(), ast_translator_best_choice(), ast_writestream(), begin_dial_channel(), filestream_destructor(), func_channel_read(), generator_force(), gtalk_rtp_read(), jingle_rtp_read(), oh323_request(), phone_read(), process_sdp(), set_format(), sip_call(), sip_request_call(), sip_rtp_read(), sip_write(), skinny_request(), transmit_connect_with_sdp(), and transmit_modify_with_sdp().
#define AST_FORMAT_G722 (1 << 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 (1 << 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 (1 << 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_set_rtpmap_type_rate(), g726_read(), g726_sample(), and g726_write().
#define AST_FORMAT_G726_AAL2 (1 << 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_lookup_mime_subtype(), ast_rtp_set_rtpmap_type_rate(), codec_ast2skinny(), codec_skinny2ast(), and setup_rtp_connection().
#define AST_FORMAT_G729A (1 << 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 (1 << 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 (1 << 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 (1 << 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 (1 << 20) |
#define AST_FORMAT_H264 (1 << 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 (1 << 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 (1 << 16) |
JPEG Images
Definition at line 276 of file frame.h.
Referenced by jpeg_read_image(), and jpeg_write_image().
#define AST_FORMAT_LPC10 (1 << 7) |
LPC10, 180 samples/frame
Definition at line 256 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_samples(), and lpc10_sample().
#define AST_FORMAT_MP4_VIDEO (1 << 22) |
#define AST_FORMAT_PNG (1 << 17) |
#define AST_FORMAT_SIREN14 (1 << 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 (1 << 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 (1 << 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(), dictate_exec(), do_waiting(), eagi_exec(), extenspy_exec(), fax_generator_generate(), find_transcoders(), handle_jack_audio(), handle_recordfile(), handle_speechcreate(), handle_speechrecognize(), iax_frame_wrap(), ices_exec(), init_outgoing(), is_encoder(), isAnsweringMachine(), jack_exec(), jack_hook_callback(), linear_alloc(), linear_generator(), 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(), play_sound_file(), playtones_alloc(), playtones_generator(), record_exec(), 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(), speech_background(), spy_generate(), tonepair_alloc(), tonepair_generator(), transmit_audio(), usbradio_new(), usbradio_read(), usbradio_request(), wav_read(), and wav_write().
#define AST_FORMAT_SLINEAR16 (1 << 15) |
Raw 16-bit Signed Linear (16000 Hz) PCM
Definition at line 272 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_slinfactory_init_rate(), console_new(), slin16_sample(), slinear_read(), slinear_write(), softmix_bridge_join(), softmix_bridge_write(), and stream_monitor().
#define AST_FORMAT_SPEEX (1 << 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_T140 (1 << 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 (1 << 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_TEXT_MASK (((1 << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK)) |
Definition at line 296 of file frame.h.
Referenced by add_sdp(), ast_request(), check_peer_ok(), sip_new(), and sip_rtp_read().
#define AST_FORMAT_ULAW (1 << 2) |
Raw mu-law data (G.711)
Definition at line 246 of file frame.h.
Referenced by __adsi_transmit_messages(), _ast_adsi_transmit_message_full(), adsi_careful_send(), alarmreceiver_exec(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), calc_energy(), codec_ast2skinny(), codec_skinny2ast(), conf_run(), convertcap(), dahdi_new(), dahdi_read(), dahdi_translate(), dahdi_write(), find_transcoders(), is_encoder(), load_module(), milliwatt_generate(), oh323_rtp_read(), old_milliwatt_exec(), phone_request(), phone_setup(), phone_write(), send_tone_burst(), start_rtp(), and ulaw_sample().
#define AST_FORMAT_VIDEO_MASK (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) |
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(), check_peer_ok(), create_addr_from_peer(), func_channel_read(), gtalk_new(), gtalk_rtp_read(), jingle_new(), jingle_rtp_read(), sip_new(), and sip_rtp_read().
#define ast_frame_byteswap_be | ( | fr | ) | do { ; } while(0) |
Definition at line 506 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 505 of file frame.h.
Referenced by phone_read().
#define AST_FRAME_DTMF AST_FRAME_DTMF_END |
Definition at line 125 of file frame.h.
Referenced by __adsi_transmit_messages(), __ast_play_and_record(), action_atxfer(), action_dahdidialoffhook(), agent_ack_sleep(), ast_audiohook_write_list(), ast_bridge_call(), ast_dsp_process(), ast_generic_bridge(), ast_jb_put(), background_detect_exec(), cb_events(), channel_spy(), cli_console_dial(), conf_exec(), 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(), rpt(), rpt_call(), rpt_exec(), send_waveform_to_channel(), sip_rtp_read(), speech_background(), ss_thread(), unistim_do_senddigit(), unistim_senddigit_end(), volume_callback(), wait_for_answer(), and wait_for_winner().
#define AST_FRAME_SET_BUFFER | ( | fr, | |
_base, | |||
_ofs, | |||
_datalen | |||
) |
{ \ (fr)->data.ptr = (char *)_base + (_ofs); \ (fr)->offset = (_ofs); \ (fr)->datalen = (_datalen); \ }
Set the various field of a frame to point to a buffer. Typically you set the base address of the buffer, the offset as AST_FRIENDLY_OFFSET, and the datalen as the amount of bytes queued. The remaining things (to be done manually) is set the number of samples, which cannot be derived from the datalen unless you know the number of bits per sample.
Definition at line 183 of file frame.h.
Referenced by fax_generator_generate(), g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), ilbc_read(), ogg_vorbis_read(), pcm_read(), siren14read(), siren7read(), slinear_read(), t38_tx_packet_handler(), vox_read(), and wav_read().
#define ast_frfree | ( | fr | ) | ast_frame_free(fr, 1) |
Definition at line 473 of file frame.h.
Referenced by __adsi_transmit_messages(), __ast_answer(), __ast_play_and_record(), __ast_queue_frame(), __ast_read(), __ast_request_and_dial(), adsi_careful_send(), agent_ack_sleep(), agent_read(), ast_audiohook_read_frame(), ast_autoservice_stop(), ast_bridge_call(), ast_bridge_handle_trip(), ast_channel_clear_softhangup(), ast_channel_free(), ast_dsp_process(), ast_generic_bridge(), ast_jb_destroy(), ast_jb_put(), ast_readaudio_callback(), ast_readvideo_callback(), ast_recvtext(), ast_rtp_write(), ast_safe_sleep_conditional(), ast_send_image(), ast_slinfactory_destroy(), ast_slinfactory_feed(), ast_slinfactory_flush(), ast_slinfactory_read(), ast_tonepair(), ast_translate(), ast_udptl_bridge(), ast_waitfordigit_full(), ast_write(), ast_writestream(), async_wait(), audio_audiohook_write_list(), autoservice_run(), background_detect_exec(), bridge_handle_dtmf(), bridge_native_loop(), bridge_p2p_loop(), calc_cost(), channel_spy(), check_goto_on_transfer(), conf_exec(), conf_flush(), conf_free(), conf_run(), create_jb(), dahdi_bridge(), dial_exec_full(), dictate_exec(), disa_exec(), do_waiting(), echo_exec(), eivr_comm(), feature_request_and_dial(), find_cache(), gen_generate(), handle_cli_file_convert(), handle_recordfile(), handle_speechrecognize(), iax2_bridge(), iax_park_thread(), ices_exec(), isAnsweringMachine(), jack_exec(), jb_empty_and_reset_adaptive(), jb_empty_and_reset_fixed(), jb_get_and_deliver(), launch_asyncagi(), manage_parkinglot(), masq_park_call(), measurenoise(), moh_files_generator(), monitor_dial(), mp3_exec(), NBScat_exec(), read_frame(), receive_dtmf_digits(), record_exec(), recordthread(), rpt(), rpt_exec(), run_agi(), send_tone_burst(), send_waveform_to_channel(), sendurl_exec(), speech_background(), spy_generate(), ss_thread(), transmit_audio(), transmit_t38(), wait_for_answer(), wait_for_hangup(), wait_for_winner(), waitforring_exec(), and waitstream_core().
#define AST_FRIENDLY_OFFSET 64 |
Offset into a frame's data buffer.
By providing some "empty" space prior to the actual data of an ast_frame, this gives any consumer of the frame ample space to prepend other necessary information without having to create a new buffer.
As an example, RTP can use the data from an ast_frame and simply prepend the RTP header information into the space provided by AST_FRIENDLY_OFFSET instead of having to create a new buffer with the necessary space allocated.
Definition at line 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(), g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), iax_frame_wrap(), ilbc_read(), jb_get_and_deliver(), linear_generator(), milliwatt_generate(), moh_generate(), mohalloc(), mp3_exec(), NBScat_exec(), newpvt(), ogg_vorbis_read(), oss_read(), pcm_read(), phone_read(), playtones_generator(), process_rfc3389(), send_tone_burst(), send_waveform_to_channel(), siren14read(), siren7read(), slinear_read(), sms_generate(), tonepair_generator(), usbradio_read(), vox_read(), and wav_read().
#define AST_HTML_BEGIN 4 |
#define AST_HTML_DATA 2 |
#define AST_HTML_END 8 |
#define AST_HTML_LDCOMPLETE 16 |
Load is complete
Definition at line 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_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(), t38_tx_packet_handler(), transmit_t38(), and udptl_rx_packet().
#define AST_MODEM_V150 2 |
#define AST_OPTION_AUDIO_MODE 4 |
Set (or clear) Audio (Not-Clear) Mode
Definition at line 380 of file frame.h.
Referenced by dahdi_hangup(), and dahdi_setoption().
#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 409 of file frame.h.
Referenced by func_channel_write(), and local_setoption().
#define AST_OPTION_ECHOCAN 8 |
Explicitly enable or disable echo cancelation for the given channel
Definition at line 402 of file frame.h.
Referenced by dahdi_setoption().
#define AST_OPTION_FLAG_REQUEST 0 |
Definition at line 362 of file frame.h.
Referenced by ast_bridge_call(), and iax2_setoption().
#define AST_OPTION_OPRMODE 7 |
Definition at line 399 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)
Definition at line 377 of file frame.h.
Referenced by dahdi_setoption(), and rpt().
#define AST_OPTION_RXGAIN 6 |
Set channel receive gain Option data is a single signed char representing number of decibels (dB) to set gain to (on top of any gain specified in channel driver)
Definition at line 396 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_T38_STATE 10 |
Definition at line 415 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
Definition at line 374 of file frame.h.
Referenced by dahdi_hangup(), dahdi_setoption(), and handle_tddmode().
#define AST_OPTION_TONE_VERIFY 1 |
Verify touchtones by muting audio transmission (and reception) and verify the tone is still present
Definition at line 371 of file frame.h.
Referenced by conf_run(), dahdi_hangup(), dahdi_setoption(), rpt(), rpt_exec(), and try_calling().
#define AST_OPTION_TXGAIN 5 |
Set channel transmit gain Option data is a single signed char representing number of decibels (dB) to set gain to (on top of any gain specified in channel driver)
Definition at line 388 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 576 of file frame.h.
Referenced by ast_rtp_write().
Definition at line 581 of file frame.h.
Referenced by ast_rtp_write().
#define AST_SMOOTHER_FLAG_BE (1 << 1) |
Definition at line 359 of file frame.h.
Referenced by ast_rtp_write().
#define AST_SMOOTHER_FLAG_G729 (1 << 0) |
Definition at line 358 of file frame.h.
Referenced by __ast_smoother_feed(), ast_smoother_read(), and smoother_frame_feed().
anonymous enum |
Definition at line 127 of file frame.h.
{ /*! This frame contains valid timing information */ AST_FRFLAG_HAS_TIMING_INFO = (1 << 0), /*! This frame came from a translator and is still the original frame. * The translator can not be free'd if the frame inside of it still has * this flag set. */ AST_FRFLAG_FROM_TRANSLATOR = (1 << 1), /*! This frame came from a dsp and is still the original frame. * The dsp cannot be free'd if the frame inside of it still has * this flag set. */ AST_FRFLAG_FROM_DSP = (1 << 2), };
AST_CONTROL_HANGUP |
Other end has hungup |
AST_CONTROL_RING |
Local ring |
AST_CONTROL_RINGING |
Remote end is ringing |
AST_CONTROL_ANSWER |
Remote end has answered |
AST_CONTROL_BUSY |
Remote end is busy |
AST_CONTROL_TAKEOFFHOOK |
Make it go off hook |
AST_CONTROL_OFFHOOK |
Line is off hook |
AST_CONTROL_CONGESTION |
Congestion (circuits busy) |
AST_CONTROL_FLASH |
Flash hook |
AST_CONTROL_WINK |
Wink |
AST_CONTROL_OPTION |
Set a low-level option |
AST_CONTROL_RADIO_KEY |
Key Radio |
AST_CONTROL_RADIO_UNKEY |
Un-Key Radio |
AST_CONTROL_PROGRESS |
Indicate PROGRESS |
AST_CONTROL_PROCEEDING |
Indicate CALL PROCEEDING |
AST_CONTROL_HOLD |
Indicate call is placed on hold |
AST_CONTROL_UNHOLD |
Indicate call is left from hold |
AST_CONTROL_VIDUPDATE |
Indicate video frame update |
_XXX_AST_CONTROL_T38 |
T38 state change request/notification
|
AST_CONTROL_SRCUPDATE |
Indicate source of media has changed |
AST_CONTROL_T38_PARAMETERS |
T38 state change request/notification with parameters |
AST_CONTROL_SRCCHANGE |
Media source has changed and requires a new RTP SSRC |
AST_CONTROL_END_OF_Q |
Indicate that this position was the end of the channel queue for a softhangup. |
Definition at line 298 of file frame.h.
{ AST_CONTROL_HANGUP = 1, /*!< Other end has hungup */ AST_CONTROL_RING = 2, /*!< Local ring */ AST_CONTROL_RINGING = 3, /*!< Remote end is ringing */ AST_CONTROL_ANSWER = 4, /*!< Remote end has answered */ AST_CONTROL_BUSY = 5, /*!< Remote end is busy */ AST_CONTROL_TAKEOFFHOOK = 6, /*!< Make it go off hook */ AST_CONTROL_OFFHOOK = 7, /*!< Line is off hook */ AST_CONTROL_CONGESTION = 8, /*!< Congestion (circuits busy) */ AST_CONTROL_FLASH = 9, /*!< Flash hook */ AST_CONTROL_WINK = 10, /*!< Wink */ AST_CONTROL_OPTION = 11, /*!< Set a low-level option */ AST_CONTROL_RADIO_KEY = 12, /*!< Key Radio */ AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */ AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */ AST_CONTROL_PROCEEDING = 15, /*!< Indicate CALL PROCEEDING */ AST_CONTROL_HOLD = 16, /*!< Indicate call is placed on hold */ AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold */ AST_CONTROL_VIDUPDATE = 18, /*!< Indicate video frame update */ _XXX_AST_CONTROL_T38 = 19, /*!< T38 state change request/notification \deprecated This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead. */ AST_CONTROL_SRCUPDATE = 20, /*!< Indicate source of media has changed */ AST_CONTROL_T38_PARAMETERS = 24, /*!< T38 state change request/notification with parameters */ AST_CONTROL_SRCCHANGE = 25, /*!< Media source has changed and requires a new RTP SSRC */ AST_CONTROL_END_OF_Q = 29, /*!< Indicate that this position was the end of the channel queue for a softhangup. */ };
enum ast_control_t38 |
Definition at line 324 of file frame.h.
{ AST_T38_REQUEST_NEGOTIATE = 1, /*!< 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 */ };
enum ast_control_t38_rate |
enum ast_frame_type |
Frame types.
Definition at line 98 of file frame.h.
{ /*! DTMF end event, subclass is the digit */ AST_FRAME_DTMF_END = 1, /*! Voice data, subclass is AST_FORMAT_* */ AST_FRAME_VOICE, /*! Video frame, maybe?? :) */ AST_FRAME_VIDEO, /*! A control frame, subclass is AST_CONTROL_* */ AST_FRAME_CONTROL, /*! An empty, useless frame */ AST_FRAME_NULL, /*! Inter Asterisk Exchange private frame type */ AST_FRAME_IAX, /*! Text messages */ AST_FRAME_TEXT, /*! Image Frames */ AST_FRAME_IMAGE, /*! HTML Frame */ AST_FRAME_HTML, /*! Comfort Noise frame (subclass is level of CNG in -dBov), body may include zero or more 8-bit quantization coefficients */ AST_FRAME_CNG, /*! Modem-over-IP data streams */ AST_FRAME_MODEM, /*! DTMF begin event, subclass is the digit */ AST_FRAME_DTMF_BEGIN, };
int __ast_smoother_feed | ( | struct ast_smoother * | s, |
struct ast_frame * | f, | ||
int | swap | ||
) |
Definition at line 201 of file frame.c.
References AST_FRAME_VOICE, ast_log(), AST_MIN_OFFSET, AST_SMOOTHER_FLAG_G729, ast_swapcopy_samples(), ast_frame::data, ast_frame::datalen, f, 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.
{ if (f->frametype != AST_FRAME_VOICE) { ast_log(LOG_WARNING, "Huh? Can't smooth a non-voice frame!\n"); return -1; } if (!s->format) { s->format = f->subclass; s->samplesperbyte = (float)f->samples / (float)f->datalen; } else if (s->format != f->subclass) { ast_log(LOG_WARNING, "Smoother was working on %d format frames, now trying to feed %d?\n", s->format, f->subclass); return -1; } if (s->len + f->datalen > SMOOTHER_SIZE) { ast_log(LOG_WARNING, "Out of smoother space\n"); return -1; } if (((f->datalen == s->size) || ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) && !s->opt && !s->len && (f->offset >= AST_MIN_OFFSET)) { /* Optimize by sending the frame we just got on the next read, thus eliminating the douple copy */ if (swap) ast_swapcopy_samples(f->data.ptr, f->data.ptr, f->samples); s->opt = f; s->opt_needs_swap = swap ? 1 : 0; return 0; } return smoother_frame_feed(s, f, swap); }
char* ast_codec2str | ( | int | codec | ) |
Get a name from a format Gets a name from a format.
codec | codec number (1,2,4,8,16,etc.) |
Definition at line 654 of file frame.c.
References ARRAY_LEN, and ast_format_list::desc.
Referenced by moh_alloc(), show_codec_n(), and show_codecs().
{ int x; char *ret = "unknown"; for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { if (AST_FORMAT_LIST[x].bits == codec) { ret = AST_FORMAT_LIST[x].desc; break; } } return ret; }
int ast_codec_choose | ( | struct ast_codec_pref * | pref, |
int | formats, | ||
int | find_best | ||
) |
Select the best audio format according to preference list from supplied options. If "find_best" is non-zero then if nothing is found, the "Best" format of the format list is selected, otherwise 0 is returned.
Definition at line 1211 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().
{ int x, ret = 0, slot; for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { slot = pref->order[x]; if (!slot) break; if (formats & AST_FORMAT_LIST[slot-1].bits) { ret = AST_FORMAT_LIST[slot-1].bits; break; } } if (ret & AST_FORMAT_AUDIO_MASK) return ret; ast_debug(4, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec"); return find_best ? ast_best_codec(formats) : 0; }
int ast_codec_get_len | ( | int | format, |
int | samples | ||
) |
Returns the number of bytes for the number of samples of the given format.
Definition at line 1486 of file frame.c.
References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G722, AST_FORMAT_G723_1, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_ILBC, AST_FORMAT_SIREN14, AST_FORMAT_SIREN7, AST_FORMAT_SLINEAR, AST_FORMAT_SLINEAR16, AST_FORMAT_ULAW, ast_getformatname(), ast_log(), len(), and LOG_WARNING.
Referenced by moh_generate(), and monmp3thread().
{ int len = 0; /* XXX Still need speex, and lpc10 XXX */ switch(format) { case AST_FORMAT_G723_1: len = (samples / 240) * 20; break; case AST_FORMAT_ILBC: len = (samples / 240) * 50; break; case AST_FORMAT_GSM: len = (samples / 160) * 33; break; case AST_FORMAT_G729A: len = samples / 8; break; case AST_FORMAT_SLINEAR: case AST_FORMAT_SLINEAR16: len = samples * 2; break; case AST_FORMAT_ULAW: case AST_FORMAT_ALAW: len = samples; break; case AST_FORMAT_G722: case AST_FORMAT_ADPCM: case AST_FORMAT_G726: case AST_FORMAT_G726_AAL2: len = samples / 2; break; case AST_FORMAT_SIREN7: /* 16,000 samples per second at 32kbps is 4,000 bytes per second */ len = samples / (16000 / 4000); break; case AST_FORMAT_SIREN14: /* 32,000 samples per second at 48kbps is 6,000 bytes per second */ len = (int) samples / ((float) 32000 / 6000); break; default: ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format)); } return len; }
int ast_codec_get_samples | ( | struct ast_frame * | f | ) |
Returns the number of samples contained in the frame.
Definition at line 1433 of file frame.c.
References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G722, AST_FORMAT_G723_1, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_ILBC, AST_FORMAT_LPC10, AST_FORMAT_SIREN14, AST_FORMAT_SIREN7, AST_FORMAT_SLINEAR, AST_FORMAT_SLINEAR16, AST_FORMAT_SPEEX, AST_FORMAT_ULAW, ast_getformatname(), ast_log(), ast_frame::data, ast_frame::datalen, g723_samples(), LOG_WARNING, ast_frame::ptr, speex_samples(), and ast_frame::subclass.
Referenced by ast_rtp_read(), isAnsweringMachine(), moh_generate(), schedule_delivery(), socket_process(), and socket_process_meta().
{ int samples = 0; switch(f->subclass) { case AST_FORMAT_SPEEX: samples = speex_samples(f->data.ptr, f->datalen); break; case AST_FORMAT_G723_1: samples = g723_samples(f->data.ptr, f->datalen); break; case AST_FORMAT_ILBC: samples = 240 * (f->datalen / 50); break; case AST_FORMAT_GSM: samples = 160 * (f->datalen / 33); break; case AST_FORMAT_G729A: samples = f->datalen * 8; break; case AST_FORMAT_SLINEAR: case AST_FORMAT_SLINEAR16: samples = f->datalen / 2; break; case AST_FORMAT_LPC10: /* assumes that the RTP packet contains one LPC10 frame */ samples = 22 * 8; samples += (((char *)(f->data.ptr))[7] & 0x1) * 8; break; case AST_FORMAT_ULAW: case AST_FORMAT_ALAW: samples = f->datalen; break; case AST_FORMAT_G722: case AST_FORMAT_ADPCM: case AST_FORMAT_G726: case AST_FORMAT_G726_AAL2: samples = f->datalen * 2; break; case AST_FORMAT_SIREN7: /* 16,000 samples per second at 32kbps is 4,000 bytes per second */ samples = f->datalen * (16000 / 4000); break; case AST_FORMAT_SIREN14: /* 32,000 samples per second at 48kbps is 6,000 bytes per second */ samples = (int) f->datalen * ((float) 32000 / 6000); break; default: ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass)); } return samples; }
static int ast_codec_interp_len | ( | int | format | ) | [inline, static] |
Gets duration in ms of interpolation frame for a format.
Definition at line 672 of file frame.h.
References AST_FORMAT_ILBC.
Referenced by __get_from_jb(), and jb_get_and_deliver().
{ return (format == AST_FORMAT_ILBC) ? 30 : 20; }
int ast_codec_pref_append | ( | struct ast_codec_pref * | pref, |
int | format | ||
) |
Append a audio codec to a preference list, removing it first if it was already there.
Definition at line 1071 of file frame.c.
References ARRAY_LEN, ast_codec_pref_remove(), and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow().
{ int x, newindex = 0; ast_codec_pref_remove(pref, format); for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { if (AST_FORMAT_LIST[x].bits == format) { newindex = x + 1; break; } } if (newindex) { for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { if (!pref->order[x]) { pref->order[x] = newindex; break; } } } return x; }
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 974 of file frame.c.
References buf, and ast_codec_pref::order.
Referenced by check_access(), create_addr(), dump_prefs(), and socket_process().
{ int x, differential = (int) 'A', mem; char *from, *to; if (right) { from = pref->order; to = buf; mem = size; } else { to = pref->order; from = buf; mem = 32; } memset(to, 0, mem); for (x = 0; x < 32 ; x++) { if (!from[x]) break; to[x] = right ? (from[x] + differential) : (from[x] - differential); } }
struct ast_format_list ast_codec_pref_getsize | ( | struct ast_codec_pref * | pref, |
int | format | ||
) | [read] |
Get packet size for codec.
Definition at line 1172 of file frame.c.
References ARRAY_LEN, ast_format_list::bits, ast_format_list::cur_ms, ast_format_list::def_ms, format, ast_format_list::inc_ms, ast_format_list::max_ms, and ast_format_list::min_ms.
Referenced by add_codec_to_sdp(), ast_rtp_bridge(), ast_rtp_codec_setpref(), ast_rtp_write(), handle_open_receive_channel_ack_message(), skinny_set_rtp_peer(), and transmit_connect().
{ int x, idx = -1, framems = 0; struct ast_format_list fmt = { 0, }; for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { if (AST_FORMAT_LIST[x].bits == format) { fmt = AST_FORMAT_LIST[x]; idx = x; break; } } for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { if (pref->order[x] == (idx + 1)) { framems = pref->framing[x]; break; } } /* size validation */ if (!framems) framems = AST_FORMAT_LIST[idx].def_ms; if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */ framems -= framems % AST_FORMAT_LIST[idx].inc_ms; if (framems < AST_FORMAT_LIST[idx].min_ms) framems = AST_FORMAT_LIST[idx].min_ms; if (framems > AST_FORMAT_LIST[idx].max_ms) framems = AST_FORMAT_LIST[idx].max_ms; fmt.cur_ms = framems; return fmt; }
int ast_codec_pref_index | ( | struct ast_codec_pref * | pref, |
int | index | ||
) |
Codec located at a particular place in the preference index.
Definition at line 1032 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().
{ int slot = 0; if ((idx >= 0) && (idx < sizeof(pref->order))) { slot = pref->order[idx]; } return slot ? AST_FORMAT_LIST[slot - 1].bits : 0; }
void ast_codec_pref_init | ( | struct ast_codec_pref * | pref | ) |
Initialize an audio codec preference to "no preference".
void ast_codec_pref_prepend | ( | struct ast_codec_pref * | pref, |
int | format, | ||
int | only_if_existing | ||
) |
Prepend an audio codec to a preference list, removing it first if it was already there.
Definition at line 1097 of file frame.c.
References ARRAY_LEN, ast_codec_pref::framing, and ast_codec_pref::order.
Referenced by create_addr().
{ int x, newindex = 0; /* First step is to get the codecs "index number" */ for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { if (AST_FORMAT_LIST[x].bits == format) { newindex = x + 1; break; } } /* Done if its unknown */ if (!newindex) return; /* Now find any existing occurrence, or the end */ for (x = 0; x < 32; x++) { if (!pref->order[x] || pref->order[x] == newindex) break; } if (only_if_existing && !pref->order[x]) return; /* Move down to make space to insert - either all the way to the end, or as far as the existing location (which will be overwritten) */ for (; x > 0; x--) { pref->order[x] = pref->order[x - 1]; pref->framing[x] = pref->framing[x - 1]; } /* And insert the new entry */ pref->order[0] = newindex; pref->framing[0] = 0; /* ? */ }
void ast_codec_pref_remove | ( | struct ast_codec_pref * | pref, |
int | format | ||
) |
Remove audio a codec from a preference list.
Definition at line 1044 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().
{ struct ast_codec_pref oldorder; int x, y = 0; int slot; int size; if (!pref->order[0]) return; memcpy(&oldorder, pref, sizeof(oldorder)); memset(pref, 0, sizeof(*pref)); for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { slot = oldorder.order[x]; size = oldorder.framing[x]; if (! slot) break; if (AST_FORMAT_LIST[slot-1].bits != format) { pref->order[y] = slot; pref->framing[y++] = size; } } }
int ast_codec_pref_setsize | ( | struct ast_codec_pref * | pref, |
int | format, | ||
int | framems | ||
) |
Set packet size for codec.
Definition at line 1134 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().
{ int x, idx = -1; for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { if (AST_FORMAT_LIST[x].bits == format) { idx = x; break; } } if (idx < 0) return -1; /* size validation */ if (!framems) framems = AST_FORMAT_LIST[idx].def_ms; if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */ framems -= framems % AST_FORMAT_LIST[idx].inc_ms; if (framems < AST_FORMAT_LIST[idx].min_ms) framems = AST_FORMAT_LIST[idx].min_ms; if (framems > AST_FORMAT_LIST[idx].max_ms) framems = AST_FORMAT_LIST[idx].max_ms; for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { if (pref->order[x] == (idx + 1)) { pref->framing[x] = framems; break; } } return x; }
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 997 of file frame.c.
References ast_codec_pref_index(), and ast_getformatname().
Referenced by dump_prefs(), and socket_process().
{ int x, codec; size_t total_len, slen; char *formatname; memset(buf,0,size); total_len = size; buf[0] = '('; total_len--; for(x = 0; x < 32 ; x++) { if (total_len <= 0) break; if (!(codec = ast_codec_pref_index(pref,x))) break; if ((formatname = ast_getformatname(codec))) { slen = strlen(formatname); if (slen > total_len) break; strncat(buf, formatname, total_len - 1); /* safe */ total_len -= slen; } if (total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) { strncat(buf, "|", total_len - 1); /* safe */ total_len--; } } if (total_len) { strncat(buf, ")", total_len - 1); /* safe */ total_len--; } return size - total_len; }
static force_inline int ast_format_rate | ( | int | format | ) | [static] |
Get the sample rate for a given format.
Definition at line 699 of file frame.h.
References AST_FORMAT_G722, AST_FORMAT_SIREN14, AST_FORMAT_SIREN7, and AST_FORMAT_SLINEAR16.
Referenced by __ast_read(), __get_from_jb(), ast_read_generator_actions(), ast_readaudio_callback(), ast_readvideo_callback(), ast_rtp_read(), ast_smoother_read(), ast_translate(), ast_write(), calc_cost(), calc_timestamp(), generator_force(), rtp_get_rate(), and schedule_delivery().
{ switch (format) { case AST_FORMAT_G722: case AST_FORMAT_SLINEAR16: case AST_FORMAT_SIREN7: return 16000; case AST_FORMAT_SIREN14: return 32000; default: return 8000; } }
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 1533 of file frame.c.
References AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_slinear_saturated_divide(), ast_slinear_saturated_multiply(), ast_frame::data, ast_frame::frametype, ast_frame::ptr, ast_frame::samples, and ast_frame::subclass.
Referenced by audiohook_read_frame_single(), audiohook_volume_callback(), conf_run(), and volume_callback().
{ int count; short *fdata = f->data.ptr; short adjust_value = abs(adjustment); if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR)) return -1; if (!adjustment) return 0; for (count = 0; count < f->samples; count++) { if (adjustment > 0) { ast_slinear_saturated_multiply(&fdata[count], &adjust_value); } else if (adjustment < 0) { ast_slinear_saturated_divide(&fdata[count], &adjust_value); } } return 0; }
void ast_frame_dump | ( | const char * | name, |
struct ast_frame * | f, | ||
char * | prefix | ||
) |
Dump a frame for debugging purposes
Definition at line 756 of file frame.c.
References AST_CONTROL_ANSWER, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_FLASH, AST_CONTROL_HANGUP, AST_CONTROL_HOLD, AST_CONTROL_OFFHOOK, AST_CONTROL_OPTION, AST_CONTROL_RADIO_KEY, AST_CONTROL_RADIO_UNKEY, AST_CONTROL_RING, AST_CONTROL_RINGING, AST_CONTROL_T38_PARAMETERS, AST_CONTROL_TAKEOFFHOOK, AST_CONTROL_UNHOLD, AST_CONTROL_WINK, ast_copy_string(), AST_FRAME_CONTROL, AST_FRAME_DTMF_BEGIN, AST_FRAME_DTMF_END, AST_FRAME_HTML, AST_FRAME_IAX, AST_FRAME_IMAGE, AST_FRAME_MODEM, AST_FRAME_NULL, AST_FRAME_TEXT, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_getformatname(), AST_HTML_BEGIN, AST_HTML_DATA, AST_HTML_END, AST_HTML_LDCOMPLETE, AST_HTML_LINKREJECT, AST_HTML_LINKURL, AST_HTML_NOSUPPORT, AST_HTML_UNLINK, AST_HTML_URL, AST_MODEM_T38, AST_MODEM_V150, ast_strlen_zero(), AST_T38_NEGOTIATED, AST_T38_REFUSED, AST_T38_REQUEST_NEGOTIATE, AST_T38_REQUEST_TERMINATE, AST_T38_TERMINATED, ast_verbose(), COLOR_BLACK, COLOR_BRCYAN, COLOR_BRGREEN, COLOR_BRMAGENTA, COLOR_BRRED, COLOR_YELLOW, ast_frame::data, ast_frame::datalen, ast_frame::frametype, ast_frame::ptr, ast_control_t38_parameters::request_response, ast_frame::subclass, and term_color().
Referenced by __ast_read(), and ast_write().
{ const char noname[] = "unknown"; char ftype[40] = "Unknown Frametype"; char cft[80]; char subclass[40] = "Unknown Subclass"; char csub[80]; char moreinfo[40] = ""; char cn[60]; char cp[40]; char cmn[40]; const char *message = "Unknown"; if (!name) name = noname; if (!f) { ast_verbose("%s [ %s (NULL) ] [%s]\n", term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); return; } /* XXX We should probably print one each of voice and video when the format changes XXX */ if (f->frametype == AST_FRAME_VOICE) return; if (f->frametype == AST_FRAME_VIDEO) return; switch(f->frametype) { case AST_FRAME_DTMF_BEGIN: strcpy(ftype, "DTMF Begin"); subclass[0] = f->subclass; subclass[1] = '\0'; break; case AST_FRAME_DTMF_END: strcpy(ftype, "DTMF End"); subclass[0] = f->subclass; subclass[1] = '\0'; break; case AST_FRAME_CONTROL: strcpy(ftype, "Control"); switch(f->subclass) { case AST_CONTROL_HANGUP: strcpy(subclass, "Hangup"); break; case AST_CONTROL_RING: strcpy(subclass, "Ring"); break; case AST_CONTROL_RINGING: strcpy(subclass, "Ringing"); break; case AST_CONTROL_ANSWER: strcpy(subclass, "Answer"); break; case AST_CONTROL_BUSY: strcpy(subclass, "Busy"); break; case AST_CONTROL_TAKEOFFHOOK: strcpy(subclass, "Take Off Hook"); break; case AST_CONTROL_OFFHOOK: strcpy(subclass, "Line Off Hook"); break; case AST_CONTROL_CONGESTION: strcpy(subclass, "Congestion"); break; case AST_CONTROL_FLASH: strcpy(subclass, "Flash"); break; case AST_CONTROL_WINK: strcpy(subclass, "Wink"); break; case AST_CONTROL_OPTION: strcpy(subclass, "Option"); break; case AST_CONTROL_RADIO_KEY: strcpy(subclass, "Key Radio"); break; case AST_CONTROL_RADIO_UNKEY: strcpy(subclass, "Unkey Radio"); break; case AST_CONTROL_HOLD: strcpy(subclass, "Hold"); break; case AST_CONTROL_UNHOLD: strcpy(subclass, "Unhold"); break; case AST_CONTROL_T38_PARAMETERS: if (f->datalen != sizeof(struct ast_control_t38_parameters)) { message = "Invalid"; } else { struct ast_control_t38_parameters *parameters = f->data.ptr; enum ast_control_t38 state = parameters->request_response; if (state == AST_T38_REQUEST_NEGOTIATE) message = "Negotiation Requested"; else if (state == AST_T38_REQUEST_TERMINATE) message = "Negotiation Request Terminated"; else if (state == AST_T38_NEGOTIATED) message = "Negotiated"; else if (state == AST_T38_TERMINATED) message = "Terminated"; else if (state == AST_T38_REFUSED) message = "Refused"; } snprintf(subclass, sizeof(subclass), "T38_Parameters/%s", message); break; case -1: strcpy(subclass, "Stop generators"); break; default: snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass); } break; case AST_FRAME_NULL: strcpy(ftype, "Null Frame"); strcpy(subclass, "N/A"); break; case AST_FRAME_IAX: /* Should never happen */ strcpy(ftype, "IAX Specific"); snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass); break; case AST_FRAME_TEXT: strcpy(ftype, "Text"); strcpy(subclass, "N/A"); ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); break; case AST_FRAME_IMAGE: strcpy(ftype, "Image"); snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass)); break; case AST_FRAME_HTML: strcpy(ftype, "HTML"); switch(f->subclass) { case AST_HTML_URL: strcpy(subclass, "URL"); ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); break; case AST_HTML_DATA: strcpy(subclass, "Data"); break; case AST_HTML_BEGIN: strcpy(subclass, "Begin"); break; case AST_HTML_END: strcpy(subclass, "End"); break; case AST_HTML_LDCOMPLETE: strcpy(subclass, "Load Complete"); break; case AST_HTML_NOSUPPORT: strcpy(subclass, "No Support"); break; case AST_HTML_LINKURL: strcpy(subclass, "Link URL"); ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); break; case AST_HTML_UNLINK: strcpy(subclass, "Unlink"); break; case AST_HTML_LINKREJECT: strcpy(subclass, "Link Reject"); break; default: snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass); break; } break; case AST_FRAME_MODEM: strcpy(ftype, "Modem"); switch (f->subclass) { case AST_MODEM_T38: strcpy(subclass, "T.38"); break; case AST_MODEM_V150: strcpy(subclass, "V.150"); break; default: snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass); break; } break; default: snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype); } if (!ast_strlen_zero(moreinfo)) ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n", term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), f->frametype, term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), f->subclass, term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)), term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); else ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n", term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), f->frametype, term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), f->subclass, term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); }
struct ast_frame* ast_frame_enqueue | ( | struct ast_frame * | head, |
struct ast_frame * | f, | ||
int | maxlen, | ||
int | dupe | ||
) | [read] |
Appends a frame to the end of a list of frames, truncating the maximum length of the list.
void ast_frame_free | ( | struct ast_frame * | fr, |
int | cache | ||
) |
Requests a frame to be allocated.
source | Request a frame be allocated. source is an optional source of the frame, len is the requested length, or "0" if the caller will supply the buffer |
Frees a frame or list of frames
fr | Frame to free, or head of list to free |
cache | Whether to consider this frame for frame caching |
Definition at line 373 of file frame.c.
References __frame_free(), AST_LIST_NEXT, and ast_frame::next.
Referenced by mixmonitor_thread().
{ struct ast_frame *next; for (next = AST_LIST_NEXT(frame, frame_list); frame; frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) { __frame_free(frame, cache); } }
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 1556 of file frame.c.
References AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_slinear_saturated_add(), ast_frame::data, ast_frame::frametype, ast_frame::ptr, ast_frame::samples, and ast_frame::subclass.
{ int count; short *data1, *data2; if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR)) return -1; if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR)) return -1; if (f1->samples != f2->samples) return -1; for (count = 0, data1 = f1->data.ptr, data2 = f2->data.ptr; count < f1->samples; count++, data1++, data2++) ast_slinear_saturated_add(data1, data2); return 0; }
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(), buf, ast_frame::data, ast_frame::datalen, ast_frame::delivery, frames, ast_frame::frametype, ast_frame::len, len(), ast_frame_cache::list, ast_frame::mallocd, ast_frame::mallocd_hdr_len, ast_frame::offset, ast_frame::ptr, ast_frame::samples, ast_frame::seqno, ast_frame_cache::size, ast_frame::src, ast_frame::subclass, ast_frame::ts, and ast_frame::uint32.
Referenced by __ast_queue_frame(), ast_frisolate(), ast_jb_put(), ast_rtp_write(), ast_slinfactory_feed(), audiohook_read_frame_both(), audiohook_read_frame_single(), autoservice_run(), process_rfc2833(), recordthread(), rpt(), and rpt_exec().
{ struct ast_frame *out = NULL; int len, srclen = 0; void *buf = NULL; #if !defined(LOW_MEMORY) struct ast_frame_cache *frames; #endif /* Start with standard stuff */ len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; /* If we have a source, add space for it */ /* * XXX Watch out here - if we receive a src which is not terminated * properly, we can be easily attacked. Should limit the size we deal with. */ if (f->src) srclen = strlen(f->src); if (srclen > 0) len += srclen + 1; #if !defined(LOW_MEMORY) if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) { AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) { if (out->mallocd_hdr_len >= len) { size_t mallocd_len = out->mallocd_hdr_len; AST_LIST_REMOVE_CURRENT(frame_list); memset(out, 0, sizeof(*out)); out->mallocd_hdr_len = mallocd_len; buf = out; frames->size--; break; } } AST_LIST_TRAVERSE_SAFE_END; } #endif if (!buf) { if (!(buf = ast_calloc_cache(1, len))) return NULL; out = buf; out->mallocd_hdr_len = len; } out->frametype = f->frametype; out->subclass = f->subclass; out->datalen = f->datalen; out->samples = f->samples; out->delivery = f->delivery; /* Set us as having malloc'd header only, so it will eventually get freed. */ out->mallocd = AST_MALLOCD_HDR; out->offset = AST_FRIENDLY_OFFSET; if (out->datalen) { out->data.ptr = buf + sizeof(*out) + AST_FRIENDLY_OFFSET; memcpy(out->data.ptr, f->data.ptr, out->datalen); } else { out->data.uint32 = f->data.uint32; } if (srclen > 0) { /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */ char *src; out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; src = (char *) out->src; /* Must have space since we allocated for it */ strcpy(src, f->src); } ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO); out->ts = f->ts; out->len = f->len; out->seqno = f->seqno; return out; }
Makes a frame independent of any static storage.
fr | frame to act upon Take a frame, and if it's not been malloc'd, make a malloc'd copy and if the data hasn't been malloced then make the data malloc'd. If you need to store frames, say for queueing, then you should call this function. |
Definition at line 389 of file frame.c.
References ast_clear_flag, ast_copy_flags, ast_frame_header_new(), ast_frdup(), ast_free, AST_FRFLAG_FROM_DSP, AST_FRFLAG_FROM_TRANSLATOR, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, ast_malloc, AST_MALLOCD_DATA, AST_MALLOCD_HDR, AST_MALLOCD_SRC, ast_strdup, ast_test_flag, ast_frame::data, ast_frame::datalen, ast_frame::frametype, ast_frame::len, ast_frame::mallocd, ast_frame::offset, ast_frame::ptr, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, ast_frame::ts, and ast_frame::uint32.
Referenced by __ast_answer(), ast_rtp_read(), ast_safe_sleep_conditional(), ast_slinfactory_feed(), ast_write(), autoservice_run(), feature_request_and_dial(), jpeg_read_image(), and read_frame().
{ struct ast_frame *out; void *newdata; /* if none of the existing frame is malloc'd, let ast_frdup() do it since it is more efficient */ if (fr->mallocd == 0) { return ast_frdup(fr); } /* if everything is already malloc'd, we are done */ if ((fr->mallocd & (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) == (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) { return fr; } if (!(fr->mallocd & AST_MALLOCD_HDR)) { /* Allocate a new header if needed */ if (!(out = ast_frame_header_new())) { return NULL; } out->frametype = fr->frametype; out->subclass = fr->subclass; out->datalen = fr->datalen; out->samples = fr->samples; out->offset = fr->offset; /* Copy the timing data */ ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO); if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) { out->ts = fr->ts; out->len = fr->len; out->seqno = fr->seqno; } } else { ast_clear_flag(fr, AST_FRFLAG_FROM_TRANSLATOR); ast_clear_flag(fr, AST_FRFLAG_FROM_DSP); out = fr; } if (!(fr->mallocd & AST_MALLOCD_SRC) && fr->src) { if (!(out->src = ast_strdup(fr->src))) { if (out != fr) { ast_free(out); } return NULL; } } else { out->src = fr->src; fr->src = NULL; fr->mallocd &= ~AST_MALLOCD_SRC; } if (!(fr->mallocd & AST_MALLOCD_DATA)) { if (!fr->datalen) { out->data.uint32 = fr->data.uint32; out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC; return out; } if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) { if (out->src != fr->src) { ast_free((void *) out->src); } if (out != fr) { ast_free(out); } return NULL; } newdata += AST_FRIENDLY_OFFSET; out->offset = AST_FRIENDLY_OFFSET; out->datalen = fr->datalen; memcpy(newdata, fr->data.ptr, fr->datalen); out->data.ptr = newdata; } else { out->data = fr->data; memset(&fr->data, 0, sizeof(fr->data)); fr->mallocd &= ~AST_MALLOCD_DATA; } out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA; return out; }
struct ast_format_list* ast_get_format_list | ( | size_t * | size | ) | [read] |
Definition at line 567 of file frame.c.
References ARRAY_LEN, and AST_FORMAT_LIST.
{ *size = ARRAY_LEN(AST_FORMAT_LIST); return AST_FORMAT_LIST; }
struct ast_format_list* ast_get_format_list_index | ( | int | index | ) | [read] |
Definition at line 562 of file frame.c.
{ return &AST_FORMAT_LIST[idx]; }
int ast_getformatbyname | ( | const char * | name | ) |
Gets a format from a name.
name | string of format |
Definition at line 636 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().
{ int x, all, format = 0; all = strcasecmp(name, "all") ? 0 : 1; for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { if (all || !strcasecmp(AST_FORMAT_LIST[x].name,name) || !strcasecmp(AST_FORMAT_LIST[x].name, ast_expand_codec_alias(name))) { format |= AST_FORMAT_LIST[x].bits; if (!all) break; } } return format; }
char* ast_getformatname | ( | int | format | ) |
Get the name of a format.
format | id of format |
Definition at line 573 of file frame.c.
References ARRAY_LEN, ast_format_list::bits, and ast_format_list::name.
Referenced by __ast_play_and_record(), __ast_read(), __ast_register_translator(), _sip_show_peer(), _skinny_show_line(), add_codec_to_answer(), add_codec_to_sdp(), add_tcodec_to_sdp(), add_vcodec_to_sdp(), agent_call(), ast_codec_get_len(), ast_codec_get_samples(), ast_codec_pref_string(), ast_dsp_process(), ast_frame_dump(), ast_openvstream(), ast_rtp_write(), ast_slinfactory_feed(), ast_streamfile(), ast_translator_build_path(), ast_unregister_translator(), ast_writestream(), background_detect_exec(), bridge_channel_join(), bridge_make_compatible(), dahdi_read(), do_waiting(), eagi_exec(), func_channel_read(), function_iaxpeer(), function_sippeer(), gtalk_show_channels(), handle_cli_core_show_file_formats(), handle_cli_core_show_translation(), handle_cli_iax2_show_channels(), handle_cli_iax2_show_peer(), handle_cli_moh_show_classes(), handle_core_show_image_formats(), iax2_request(), iax_show_provisioning(), jingle_show_channels(), login_exec(), moh_release(), oh323_rtp_read(), phone_setup(), print_codec_to_cli(), rebuild_matrix(), register_translator(), set_format(), set_local_capabilities(), set_peer_capabilities(), show_codecs(), sip_request_call(), sip_rtp_read(), socket_process(), start_rtp(), unistim_request(), unistim_rtp_read(), and unistim_write().
{ int x; char *ret = "unknown"; for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { if (AST_FORMAT_LIST[x].bits == format) { ret = AST_FORMAT_LIST[x].name; break; } } return ret; }
char* ast_getformatname_multiple | ( | char * | buf, |
size_t | size, | ||
int | format | ||
) |
Get the names of a set of formats.
buf | a buffer for the output string |
size | size of buf (bytes) |
format | the format (combined IDs of codecs) Prints a list of readable codec names corresponding to "format". ex: for format=AST_FORMAT_GSM|AST_FORMAT_SPEEX|AST_FORMAT_ILBC it will return "0x602 (GSM|SPEEX|ILBC)" |
Definition at line 586 of file frame.c.
References ARRAY_LEN, ast_copy_string(), ast_format_list::bits, buf, len(), and name.
Referenced by __ast_read(), _sip_show_peer(), _skinny_show_device(), _skinny_show_line(), add_sdp(), ast_streamfile(), bridge_make_compatible(), function_iaxpeer(), function_sippeer(), gtalk_is_answered(), gtalk_newcall(), handle_cli_iax2_show_peer(), handle_showchan(), iax2_bridge(), process_sdp(), serialize_showchan(), set_format(), show_channels_cb(), sip_new(), sip_request_call(), sip_show_channel(), sip_show_settings(), and sip_write().
{ int x; unsigned len; char *start, *end = buf; if (!size) return buf; snprintf(end, size, "0x%x (", format); len = strlen(end); end += len; size -= len; start = end; for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { if (AST_FORMAT_LIST[x].bits & format) { snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name); len = strlen(end); end += len; size -= len; } } if (start == end) ast_copy_string(start, "nothing)", size); else if (size > 1) *(end -1) = ')'; return buf; }
int ast_parse_allow_disallow | ( | struct ast_codec_pref * | pref, |
int * | mask, | ||
const char * | list, | ||
int | allowing | ||
) |
Parse an "allow" or "deny" line in a channel or device configuration and update the capabilities mask and pref if provided. Video codecs are not added to codec preference lists, since we can not transcode.
Definition at line 1233 of file frame.c.
References ast_codec_pref_append(), ast_codec_pref_remove(), ast_codec_pref_setsize(), ast_debug, AST_FORMAT_AUDIO_MASK, ast_getformatbyname(), ast_log(), ast_strdupa, format, LOG_WARNING, parse(), and strsep().
Referenced by action_originate(), apply_outgoing(), build_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().
{ int errors = 0; char *parse = NULL, *this = NULL, *psize = NULL; int format = 0, framems = 0; parse = ast_strdupa(list); while ((this = strsep(&parse, ","))) { framems = 0; if ((psize = strrchr(this, ':'))) { *psize++ = '\0'; ast_debug(1, "Packetization for codec: %s is %s\n", this, psize); framems = atoi(psize); if (framems < 0) { framems = 0; errors++; ast_log(LOG_WARNING, "Bad packetization value for codec %s\n", this); } } if (!(format = ast_getformatbyname(this))) { ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this); errors++; continue; } if (mask) { if (allowing) *mask |= format; else *mask &= ~format; } /* Set up a preference list for audio. Do not include video in preferences since we can not transcode video and have to use whatever is offered */ if (pref && (format & AST_FORMAT_AUDIO_MASK)) { if (strcasecmp(this, "all")) { if (allowing) { ast_codec_pref_append(pref, format); ast_codec_pref_setsize(pref, format, framems); } else ast_codec_pref_remove(pref, format); } else if (!allowing) { memset(pref, 0, sizeof(*pref)); } } } return errors; }
void ast_smoother_free | ( | struct ast_smoother * | s | ) |
Definition at line 286 of file frame.c.
References ast_free.
Referenced by ast_rtp_destroy(), and ast_rtp_write().
{ ast_free(s); }
int ast_smoother_get_flags | ( | struct ast_smoother * | smoother | ) |
struct ast_smoother* ast_smoother_new | ( | int | bytes | ) | [read] |
Definition at line 176 of file frame.c.
References ast_malloc, ast_smoother_reset(), and s.
Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().
{ struct ast_smoother *s; if (size < 1) return NULL; if ((s = ast_malloc(sizeof(*s)))) ast_smoother_reset(s, size); return s; }
struct ast_frame* ast_smoother_read | ( | struct ast_smoother * | s | ) | [read] |
Definition at line 236 of file frame.c.
References ast_format_rate(), AST_FRAME_VOICE, AST_FRIENDLY_OFFSET, ast_log(), ast_samp2tv(), AST_SMOOTHER_FLAG_G729, ast_tvadd(), ast_tvzero(), ast_smoother::data, ast_frame::data, ast_frame::datalen, ast_smoother::delivery, ast_frame::delivery, ast_smoother::f, ast_smoother::flags, ast_smoother::format, ast_smoother::framedata, ast_frame::frametype, ast_smoother::len, len(), LOG_WARNING, ast_frame::offset, ast_smoother::opt, ast_frame::ptr, ast_frame::samples, ast_smoother::samplesperbyte, ast_smoother::size, and ast_frame::subclass.
Referenced by ast_rtp_write().
{ struct ast_frame *opt; int len; /* IF we have an optimization frame, send it */ if (s->opt) { if (s->opt->offset < AST_FRIENDLY_OFFSET) ast_log(LOG_WARNING, "Returning a frame of inappropriate offset (%d).\n", s->opt->offset); opt = s->opt; s->opt = NULL; return opt; } /* Make sure we have enough data */ if (s->len < s->size) { /* Or, if this is a G.729 frame with VAD on it, send it immediately anyway */ if (!((s->flags & AST_SMOOTHER_FLAG_G729) && (s->len % 10))) return NULL; } len = s->size; if (len > s->len) len = s->len; /* Make frame */ s->f.frametype = AST_FRAME_VOICE; s->f.subclass = s->format; s->f.data.ptr = s->framedata + AST_FRIENDLY_OFFSET; s->f.offset = AST_FRIENDLY_OFFSET; s->f.datalen = len; /* Samples will be improper given VAD, but with VAD the concept really doesn't even exist */ s->f.samples = len * s->samplesperbyte; /* XXX rounding */ s->f.delivery = s->delivery; /* Fill Data */ memcpy(s->f.data.ptr, s->data, len); s->len -= len; /* Move remaining data to the front if applicable */ if (s->len) { /* In principle this should all be fine because if we are sending G.729 VAD, the next timestamp will take over anyawy */ memmove(s->data, s->data + len, s->len); if (!ast_tvzero(s->delivery)) { /* If we have delivery time, increment it, otherwise, leave it at 0 */ s->delivery = ast_tvadd(s->delivery, ast_samp2tv(s->f.samples, ast_format_rate(s->format))); } } /* Return frame */ return &s->f; }
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 154 of file frame.c.
References ast_smoother::opt, ast_smoother::opt_needs_swap, ast_smoother::size, and smoother_frame_feed().
Referenced by ast_rtp_codec_setpref().
{ /* if there is no change, then nothing to do */ if (s->size == bytes) { return; } /* set the new desired output size */ s->size = bytes; /* if there is no 'optimized' frame in the smoother, * then there is nothing left to do */ if (!s->opt) { return; } /* there is an 'optimized' frame here at the old size, * but it must now be put into the buffer so the data * can be extracted at the new size */ smoother_frame_feed(s, s->opt, s->opt_needs_swap); s->opt = NULL; }
void ast_smoother_reset | ( | struct ast_smoother * | s, |
int | bytes | ||
) |
Definition at line 148 of file frame.c.
References ast_smoother::size.
Referenced by ast_smoother_new().
{ memset(s, 0, sizeof(*s)); s->size = bytes; }
void ast_smoother_set_flags | ( | struct ast_smoother * | smoother, |
int | flags | ||
) |
Definition at line 191 of file frame.c.
References ast_smoother::flags.
Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().
int ast_smoother_test_flag | ( | struct ast_smoother * | s, |
int | flag | ||
) |
Definition at line 196 of file frame.c.
References ast_smoother::flags.
Referenced by ast_rtp_write().
{ return (s->flags & flag); }
void ast_swapcopy_samples | ( | void * | dst, |
const void * | src, | ||
int | samples | ||
) |
Definition at line 551 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 124 of file frame.c.
Referenced by __ast_read(), __oh323_rtp_create(), __oh323_update_info(), agent_new(), agent_read(), ast_channel_masquerade(), ast_channel_setwhentohangup_tv(), ast_do_masquerade(), ast_rtcp_read(), ast_rtp_read(), ast_softhangup_nolock(), ast_udptl_read(), 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(), oh323_read(), oh323_rtp_read(), process_sdp(), sip_read(), sip_rtp_read(), skinny_rtp_read(), unistim_rtp_read(), and wakeup_sub().