#include <sys/types.h>
#include <sys/time.h>
#include "asterisk/compiler.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_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_AUDIO (1 << 15) |
#define | AST_FORMAT_MAX_VIDEO (1 << 24) |
#define | AST_FORMAT_PNG (1 << 17) |
#define | AST_FORMAT_SLINEAR (1 << 6) |
#define | AST_FORMAT_SPEEX (1 << 9) |
#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 { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data, __f->data, __f->samples); } while(0) |
#define | ast_frame_byteswap_le(fr) do { ; } 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 |
#define | AST_HTML_BEGIN 4 |
#define | AST_HTML_DATA 2 |
#define | AST_HTML_END 8 |
#define | AST_HTML_LDCOMPLETE 16 |
#define | AST_HTML_LINKREJECT 20 |
#define | AST_HTML_LINKURL 18 |
#define | AST_HTML_NOSUPPORT 17 |
#define | AST_HTML_UNLINK 19 |
#define | AST_HTML_URL 1 |
#define | AST_MALLOCD_DATA (1 << 1) |
#define | AST_MALLOCD_HDR (1 << 0) |
#define | AST_MALLOCD_SRC (1 << 2) |
#define | AST_MIN_OFFSET 32 |
#define | AST_MODEM_T38 1 |
#define | AST_MODEM_V150 2 |
#define | AST_OPTION_AUDIO_MODE 4 |
#define | AST_OPTION_ECHOCAN 8 |
#define | AST_OPTION_FLAG_ACCEPT 1 |
#define | AST_OPTION_FLAG_ANSWER 5 |
#define | AST_OPTION_FLAG_QUERY 4 |
#define | AST_OPTION_FLAG_REJECT 2 |
#define | AST_OPTION_FLAG_REQUEST 0 |
#define | AST_OPTION_FLAG_WTF 6 |
#define | AST_OPTION_OPRMODE 7 |
#define | AST_OPTION_RELAXDTMF 3 |
#define | AST_OPTION_RXGAIN 6 |
#define | AST_OPTION_TDD 2 |
#define | AST_OPTION_TONE_VERIFY 1 |
#define | AST_OPTION_TXGAIN 5 |
#define | ast_smoother_feed(s, f) __ast_smoother_feed(s, f, 0) |
#define | ast_smoother_feed_be(s, f) __ast_smoother_feed(s, f, 1) |
#define | ast_smoother_feed_le(s, f) __ast_smoother_feed(s, f, 0) |
#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, AST_CONTROL_SRCUPDATE = 20 } |
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 | |
int | __ast_smoother_feed (struct ast_smoother *s, struct ast_frame *f, int swap) |
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 See Audio Codec Preferences. | |
void | ast_codec_pref_init (struct ast_codec_pref *pref) |
Initialize an audio codec preference to "no preference" See Audio Codec Preferences. | |
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. | |
void | 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_smoother_free (struct ast_smoother *s) |
int | ast_smoother_get_flags (struct ast_smoother *smoother) |
struct ast_smoother * | ast_smoother_new (int bytes) |
struct ast_frame * | ast_smoother_read (struct ast_smoother *s) |
void | ast_smoother_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) |
void | ast_swapcopy_samples (void *dst, const void *src, int samples) |
Variables | |
struct ast_frame | ast_null_frame |
Definition in file frame.h.
#define AST_FORMAT_ADPCM (1 << 5) |
ADPCM (IMA)
Definition at line 244 of file frame.h.
Referenced by adpcmtolin_sample(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), convertcap(), vox_read(), and vox_write().
#define AST_FORMAT_ALAW (1 << 3) |
Raw A-law data (G.711)
Definition at line 240 of file frame.h.
Referenced by alawtolin_sample(), alawtoulaw_sample(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), cb_events(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), misdn_read(), misdn_set_opt_exec(), oh323_rtp_read(), pcm_seek(), pcm_write(), read_config(), sms_exec(), sms_generate(), zt_new(), zt_read(), and zt_write().
#define AST_FORMAT_AUDIO_MASK ((1 << 16)-1) |
Maximum audio mask
Definition at line 262 of file frame.h.
Referenced by add_sdp(), ast_best_codec(), ast_codec_choose(), ast_openstream_full(), ast_parse_allow_disallow(), ast_request_with_uniqueid(), ast_translate_available_formats(), ast_translator_best_choice(), begin_dial(), func_channel_read(), generator_force(), gtalk_rtp_read(), process_sdp(), set_format(), sip_call(), sip_rtp_read(), and sip_write().
#define AST_FORMAT_G722 (1 << 12) |
G.722
Definition at line 258 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_write(), au_seek(), convertcap(), and pcm_read().
#define AST_FORMAT_G723_1 (1 << 0) |
G.723.1 compression
Definition at line 234 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_samples(), ast_rtp_write(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), g723_read(), g723_write(), load_module(), phone_request(), phone_setup(), phone_write(), zap_destroy(), and zap_translate().
#define AST_FORMAT_G726 (1 << 11) |
ADPCM (G.726, 32kbps, RFC3551 codeword packing)
Definition at line 256 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_set_rtpmap_type(), g726_read(), g726_write(), and g726tolin_sample().
#define AST_FORMAT_G726_AAL2 (1 << 4) |
ADPCM (G.726, 32kbps, AAL2 codeword packing)
Definition at line 242 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_lookup_mime_subtype(), ast_rtp_set_rtpmap_type(), codec_ast2skinny(), and codec_skinny2ast().
#define AST_FORMAT_G729A (1 << 8) |
G.729A audio
Definition at line 250 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), g729_read(), g729_write(), zap_destroy(), zap_framein(), and zap_translate().
#define AST_FORMAT_GSM (1 << 1) |
GSM compression
Definition at line 236 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), convertcap(), gsm_read(), gsm_write(), gsmtolin_sample(), wav_read(), and wav_write().
#define AST_FORMAT_H261 (1 << 18) |
H.261 Video
Definition at line 268 of file frame.h.
Referenced by codec_ast2skinny(), and codec_skinny2ast().
#define AST_FORMAT_H263 (1 << 19) |
H.263 Video
Definition at line 270 of file frame.h.
Referenced by codec_ast2skinny(), codec_skinny2ast(), h263_read(), and h263_write().
#define AST_FORMAT_H264 (1 << 21) |
#define AST_FORMAT_ILBC (1 << 10) |
iLBC Free Compression
Definition at line 254 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_codec_interp_len(), convertcap(), ilbc_read(), and ilbc_write().
#define AST_FORMAT_JPEG (1 << 16) |
JPEG Images
Definition at line 264 of file frame.h.
Referenced by jpeg_read_image(), and jpeg_write_image().
#define AST_FORMAT_LPC10 (1 << 7) |
LPC10, 180 samples/frame
Definition at line 248 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_samples(), and lpc10tolin_sample().
#define AST_FORMAT_MAX_AUDIO (1 << 15) |
Maximum audio format
Definition at line 260 of file frame.h.
Referenced by add_sdp(), ast_closestream(), ast_filehelper(), ast_openvstream(), ast_playstream(), ast_rtp_read(), ast_translate_available_formats(), ast_writestream(), oh323_request(), phone_read(), sip_request_call(), skinny_request(), transmit_connect_with_sdp(), and transmit_modify_with_sdp().
#define AST_FORMAT_MAX_VIDEO (1 << 24) |
Maximum video format
Definition at line 276 of file frame.h.
Referenced by add_sdp(), ast_openvstream(), and ast_translate_available_formats().
#define AST_FORMAT_PNG (1 << 17) |
#define AST_FORMAT_SLINEAR (1 << 6) |
Raw 16-bit Signed Linear (8000 Hz) PCM
Definition at line 246 of file frame.h.
Referenced by __ast_play_and_record(), __ast_register_translator(), action_originate(), agent_new(), alsa_new(), alsa_read(), alsa_request(), ast_audiohook_read_frame(), ast_best_codec(), ast_channel_make_compatible(), ast_channel_start_silence_generator(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_call_progress(), ast_dsp_digitdetect(), ast_dsp_process(), ast_dsp_silence(), ast_frame_adjust_volume(), ast_frame_slinear_sum(), ast_rtp_read(), ast_send_message(), ast_slinfactory_feed(), attempt_reconnect(), attempt_thread(), audio_audiohook_write_list(), audiohook_read_frame_both(), audiohook_read_frame_single(), background_detect_exec(), build_conf(), chanspy_exec(), chanspychan_exec(), conf_run(), connect_link(), dictate_exec(), do_waiting(), eagi_exec(), extenspy_exec(), fast_originate(), handle_recordfile(), iax_frame_wrap(), ices_exec(), isAnsweringMachine(), linear_alloc(), linear_generator(), lintoadpcm_sample(), lintoalaw_sample(), lintog726_sample(), lintogsm_sample(), lintolpc10_sample(), lintospeex_sample(), lintoulaw_sample(), load_module(), load_moh_classes(), measurenoise(), misdn_set_opt_exec(), mixmonitor_thread(), moh_class_malloc(), mp3_exec(), nbs_request(), nbs_xwrite(), NBScat_exec(), ogg_vorbis_read(), ogg_vorbis_write(), oh323_rtp_read(), orig_app(), orig_exten(), oss_new(), oss_read(), oss_request(), parkandannounce_exec(), phone_new(), phone_read(), phone_request(), phone_setup(), phone_write(), playtones_alloc(), playtones_generator(), read_config(), record_exec(), rpt(), rpt_call(), rpt_exec(), rpt_tele_thread(), run_agi(), send_waveform_to_channel(), silence_generator_generate(), slinear_read(), slinear_write(), sms_exec(), sms_generate(), socket_process(), speech_background(), speech_create(), spy_generate(), tonepair_alloc(), tonepair_generator(), wav_read(), wav_write(), xagi_exec(), zt_new(), zt_read(), and zt_write().
#define AST_FORMAT_SPEEX (1 << 9) |
SpeeX Free Compression
Definition at line 252 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_samples(), ast_rtp_write(), convertcap(), and speextolin_sample().
#define AST_FORMAT_ULAW (1 << 2) |
Raw mu-law data (G.711)
Definition at line 238 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(), codec_ast2skinny(), codec_skinny2ast(), conf_run(), convertcap(), disa_exec(), load_module(), milliwatt_generate(), oh323_rtp_read(), old_milliwatt_exec(), phone_request(), phone_setup(), phone_write(), send_tone_burst(), ulawtoalaw_sample(), ulawtolin_sample(), zt_new(), zt_read(), and zt_write().
#define AST_FORMAT_VIDEO_MASK (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) |
Definition at line 277 of file frame.h.
Referenced by add_sdp(), ast_request_with_uniqueid(), ast_translate_available_formats(), check_user_full(), create_addr_from_peer(), func_channel_read(), gtalk_new(), gtalk_rtp_read(), sip_new(), and sip_rtp_read().
#define ast_frame_byteswap_be | ( | fr | ) | do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data, __f->data, __f->samples); } while(0) |
#define ast_frame_byteswap_le | ( | fr | ) | do { ; } while(0) |
#define AST_FRAME_DTMF AST_FRAME_DTMF_END |
Definition at line 125 of file frame.h.
Referenced by __adsi_transmit_messages(), __ast_play_and_record(), action_zapdialoffhook(), agent_ack_sleep(), app_exec(), ast_audiohook_write_list(), ast_bridge_call(), ast_dsp_process(), ast_feature_request_and_dial(), ast_generic_bridge(), ast_jb_put(), background_detect_exec(), cb_events(), channel_spy(), conf_exec(), conf_run(), console_dial(), console_dial_deprecated(), dictate_exec(), disa_exec(), do_immediate_setup(), echo_exec(), gtalk_handle_dtmf(), handle_recordfile(), handle_request(), handle_request_info(), iax2_bridge(), 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(), wait_for_answer(), wait_for_winner(), zt_bridge(), and zt_read().
#define AST_FRAME_SET_BUFFER | ( | fr, | |||
_base, | |||||
_ofs, | |||||
_datalen | ) |
Value:
Set the various field of a frame to point to a buffer. Typically you set the base address of the buffer, the offset as AST_FRIENDLY_OFFSET, and the datalen as the amount of bytes queued. The remaining things (to be done manually) is set the number of samples, which cannot be derived from the datalen unless you know the number of bits per sample.Definition at line 183 of file frame.h.
Referenced by g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), ilbc_read(), ogg_vorbis_read(), pcm_read(), slinear_read(), vox_read(), and wav_read().
#define ast_frfree | ( | fr | ) | ast_frame_free(fr, 1) |
Definition at line 403 of file frame.h.
Referenced by __adsi_transmit_messages(), __ast_play_and_record(), __ast_read(), __ast_request_and_dial_uniqueid(), adsi_careful_send(), agent_ack_sleep(), agent_read(), app_exec(), ast_audiohook_read_frame(), ast_autoservice_stop(), ast_bridge_call(), ast_channel_free(), ast_dsp_process(), ast_feature_request_and_dial(), ast_generic_bridge(), ast_jb_destroy(), ast_jb_put(), ast_masq_autoanswer_login(), ast_masq_hold_call(), ast_masq_park_call(), ast_queue_frame(), ast_recvtext(), ast_rtp_write(), ast_safe_sleep_conditional(), ast_send_image(), ast_slinfactory_destroy(), ast_slinfactory_feed(), ast_slinfactory_flush(), ast_slinfactory_read(), ast_tonepair(), ast_translate(), ast_udptl_bridge(), ast_waitfordigit_full(), ast_write(), ast_writestream(), async_wait(), audio_audiohook_write_list(), autoservice_run(), background_detect_exec(), bridge_native_loop(), bridge_p2p_loop(), builtin_atxfer(), calc_cost(), channel_spy(), check_goto_on_transfer(), conf_exec(), conf_flush(), conf_free(), conf_run(), create_jb(), dictate_exec(), disa_exec(), do_autoanswer_thread(), do_holding_thread(), do_parking_thread(), do_waiting(), echo_exec(), find_cache(), gen_generate(), handle_invite_replaces(), handle_recordfile(), iax2_bridge(), iax_park_thread(), ices_exec(), isAnsweringMachine(), jb_get_and_deliver(), measurenoise(), moh_files_generator(), monitor_dial(), mp3_exec(), NBScat_exec(), receive_dtmf_digits(), record_exec(), recordthread(), rpt(), rpt_exec(), run_agi(), send_tone_burst(), send_waveform_to_channel(), sendurl_exec(), sms_exec(), speech_background(), spy_generate(), ss_thread(), wait_for_answer(), wait_for_hangup(), wait_for_winner(), waitforring_exec(), waitstream_core(), and zt_bridge().
#define AST_FRIENDLY_OFFSET 64 |
Definition at line 194 of file frame.h.
Referenced by __get_from_jb(), alsa_read(), ast_frdup(), ast_frisolate(), ast_prod(), ast_rtcp_read(), ast_rtp_read(), ast_smoother_read(), ast_trans_frameout(), ast_udptl_read(), conf_run(), g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), iax_frame_wrap(), ilbc_read(), jb_get_and_deliver(), linear_generator(), milliwatt_generate(), moh_generate(), mohalloc(), mp3_exec(), NBScat_exec(), newpvt(), ogg_vorbis_read(), oss_read(), pcm_read(), phone_read(), playtones_generator(), process_rfc3389(), send_tone_burst(), send_waveform_to_channel(), slinear_read(), sms_generate(), tonepair_generator(), vox_read(), wav_read(), zap_frameout(), and zt_read().
#define AST_HTML_BEGIN 4 |
#define AST_HTML_DATA 2 |
#define AST_HTML_END 8 |
#define AST_HTML_LDCOMPLETE 16 |
Load is complete
Definition at line 222 of file frame.h.
Referenced by ast_frame_dump(), and sendurl_exec().
#define AST_HTML_LINKREJECT 20 |
#define AST_HTML_LINKURL 18 |
#define AST_HTML_NOSUPPORT 17 |
Peer is unable to support HTML
Definition at line 224 of file frame.h.
Referenced by ast_frame_dump(), and sendurl_exec().
#define AST_HTML_UNLINK 19 |
#define AST_HTML_URL 1 |
Sending a URL
Definition at line 214 of file frame.h.
Referenced by ast_channel_sendurl(), and ast_frame_dump().
#define AST_MALLOCD_DATA (1 << 1) |
Need the data be free'd?
Definition at line 202 of file frame.h.
Referenced by ast_frame_free(), and ast_frisolate().
#define AST_MALLOCD_HDR (1 << 0) |
Need the header be free'd?
Definition at line 200 of file frame.h.
Referenced by ast_frame_free(), ast_frame_header_new(), ast_frdup(), and ast_frisolate().
#define AST_MALLOCD_SRC (1 << 2) |
Need the source be free'd? (haha!)
Definition at line 204 of file frame.h.
Referenced by ast_frame_free(), and ast_frisolate().
#define AST_MIN_OFFSET 32 |
#define AST_MODEM_T38 1 |
T.38 Fax-over-IP
Definition at line 208 of file frame.h.
Referenced by ast_frame_dump(), 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 323 of file frame.h.
Referenced by ast_bridge_call(), iax2_setoption(), zt_hangup(), and zt_setoption().
#define AST_OPTION_ECHOCAN 8 |
Explicitly enable or disable echo cancelation for the given channel
Definition at line 345 of file frame.h.
Referenced by zt_setoption().
#define AST_OPTION_FLAG_REQUEST 0 |
#define AST_OPTION_OPRMODE 7 |
#define AST_OPTION_RELAXDTMF 3 |
Relax the parameters for DTMF reception (mainly for radio use)
Definition at line 320 of file frame.h.
Referenced by ast_bridge_call(), iax2_setoption(), rpt(), and zt_setoption().
#define AST_OPTION_RXGAIN 6 |
Set channel receive gain Option data is a single signed char representing number of decibels (dB) to set gain to (on top of any gain specified in channel driver)
Definition at line 339 of file frame.h.
Referenced by func_channel_write(), iax2_setoption(), play_record_review(), reset_volumes(), set_talk_volume(), vm_forwardoptions(), and zt_setoption().
#define AST_OPTION_TDD 2 |
Put a compatible channel into TDD (TTY for the hearing-impared) mode
Definition at line 317 of file frame.h.
Referenced by ast_bridge_call(), handle_tddmode(), iax2_setoption(), zt_hangup(), and zt_setoption().
#define AST_OPTION_TONE_VERIFY 1 |
Verify touchtones by muting audio transmission (and reception) and verify the tone is still present
Definition at line 314 of file frame.h.
Referenced by ast_bridge_call(), conf_run(), iax2_setoption(), rpt(), rpt_exec(), try_calling(), zt_hangup(), and zt_setoption().
#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 331 of file frame.h.
Referenced by common_exec(), func_channel_write(), iax2_setoption(), reset_volumes(), set_listen_volume(), and zt_setoption().
#define AST_SMOOTHER_FLAG_BE (1 << 1) |
#define AST_SMOOTHER_FLAG_G729 (1 << 0) |
Definition at line 301 of file frame.h.
Referenced by __ast_smoother_feed(), and ast_smoother_read().
anonymous enum |
AST_FRFLAG_HAS_TIMING_INFO | This frame contains valid timing information |
AST_FRFLAG_FROM_TRANSLATOR | 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_DSP | 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. |
Definition at line 127 of file frame.h.
00127 { 00128 /*! This frame contains valid timing information */ 00129 AST_FRFLAG_HAS_TIMING_INFO = (1 << 0), 00130 /*! This frame came from a translator and is still the original frame. 00131 * The translator can not be free'd if the frame inside of it still has 00132 * this flag set. */ 00133 AST_FRFLAG_FROM_TRANSLATOR = (1 << 1), 00134 /*! This frame came from a dsp and is still the original frame. 00135 * The dsp cannot be free'd if the frame inside of it still has 00136 * this flag set. */ 00137 AST_FRFLAG_FROM_DSP = (1 << 2), 00138 };
Definition at line 279 of file frame.h.
00279 { 00280 AST_CONTROL_HANGUP = 1, /*!< Other end has hungup */ 00281 AST_CONTROL_RING = 2, /*!< Local ring */ 00282 AST_CONTROL_RINGING = 3, /*!< Remote end is ringing */ 00283 AST_CONTROL_ANSWER = 4, /*!< Remote end has answered */ 00284 AST_CONTROL_BUSY = 5, /*!< Remote end is busy */ 00285 AST_CONTROL_TAKEOFFHOOK = 6, /*!< Make it go off hook */ 00286 AST_CONTROL_OFFHOOK = 7, /*!< Line is off hook */ 00287 AST_CONTROL_CONGESTION = 8, /*!< Congestion (circuits busy) */ 00288 AST_CONTROL_FLASH = 9, /*!< Flash hook */ 00289 AST_CONTROL_WINK = 10, /*!< Wink */ 00290 AST_CONTROL_OPTION = 11, /*!< Set a low-level option */ 00291 AST_CONTROL_RADIO_KEY = 12, /*!< Key Radio */ 00292 AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */ 00293 AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */ 00294 AST_CONTROL_PROCEEDING = 15, /*!< Indicate CALL PROCEEDING */ 00295 AST_CONTROL_HOLD = 16, /*!< Indicate call is placed on hold */ 00296 AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold */ 00297 AST_CONTROL_VIDUPDATE = 18, /*!< Indicate video frame update */ 00298 AST_CONTROL_SRCUPDATE = 20, /*!< Indicate source of media has changed */ 00299 };
enum ast_frame_type |
Frame types.
Definition at line 98 of file frame.h.
00098 { 00099 /*! DTMF end event, subclass is the digit */ 00100 AST_FRAME_DTMF_END = 1, 00101 /*! Voice data, subclass is AST_FORMAT_* */ 00102 AST_FRAME_VOICE, 00103 /*! Video frame, maybe?? :) */ 00104 AST_FRAME_VIDEO, 00105 /*! A control frame, subclass is AST_CONTROL_* */ 00106 AST_FRAME_CONTROL, 00107 /*! An empty, useless frame */ 00108 AST_FRAME_NULL, 00109 /*! Inter Asterisk Exchange private frame type */ 00110 AST_FRAME_IAX, 00111 /*! Text messages */ 00112 AST_FRAME_TEXT, 00113 /*! Image Frames */ 00114 AST_FRAME_IMAGE, 00115 /*! HTML Frame */ 00116 AST_FRAME_HTML, 00117 /*! Comfort Noise frame (subclass is level of CNG in -dBov), 00118 body may include zero or more 8-bit quantization coefficients */ 00119 AST_FRAME_CNG, 00120 /*! Modem-over-IP data streams */ 00121 AST_FRAME_MODEM, 00122 /*! DTMF begin event, subclass is the digit */ 00123 AST_FRAME_DTMF_BEGIN, 00124 };
int __ast_smoother_feed | ( | struct ast_smoother * | s, | |
struct ast_frame * | f, | |||
int | swap | |||
) |
Definition at line 172 of file frame.c.
References AST_FRAME_VOICE, ast_log(), AST_MIN_OFFSET, AST_SMOOTHER_FLAG_G729, ast_swapcopy_samples(), ast_smoother::data, ast_frame::data, ast_frame::datalen, ast_smoother::delivery, ast_frame::delivery, ast_smoother::flags, ast_smoother::format, ast_frame::frametype, ast_smoother::len, LOG_NOTICE, LOG_WARNING, ast_frame::offset, ast_smoother::opt, ast_frame::samples, ast_smoother::samplesperbyte, ast_smoother::size, SMOOTHER_SIZE, and ast_frame::subclass.
00173 { 00174 if (f->frametype != AST_FRAME_VOICE) { 00175 ast_log(LOG_WARNING, "Huh? Can't smooth a non-voice frame!\n"); 00176 return -1; 00177 } 00178 if (!s->format) { 00179 s->format = f->subclass; 00180 s->samplesperbyte = (float)f->samples / (float)f->datalen; 00181 } else if (s->format != f->subclass) { 00182 ast_log(LOG_WARNING, "Smoother was working on %d format frames, now trying to feed %d?\n", s->format, f->subclass); 00183 return -1; 00184 } 00185 if (s->len + f->datalen > SMOOTHER_SIZE) { 00186 ast_log(LOG_WARNING, "Out of smoother space\n"); 00187 return -1; 00188 } 00189 if (((f->datalen == s->size) || ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) 00190 && !s->opt && (f->offset >= AST_MIN_OFFSET)) { 00191 if (!s->len) { 00192 /* Optimize by sending the frame we just got 00193 on the next read, thus eliminating the douple 00194 copy */ 00195 if (swap) 00196 ast_swapcopy_samples(f->data, f->data, f->samples); 00197 s->opt = f; 00198 return 0; 00199 } 00200 } 00201 if (s->flags & AST_SMOOTHER_FLAG_G729) { 00202 if (s->len % 10) { 00203 ast_log(LOG_NOTICE, "Dropping extra frame of G.729 since we already have a VAD frame at the end\n"); 00204 return 0; 00205 } 00206 } 00207 if (swap) 00208 ast_swapcopy_samples(s->data+s->len, f->data, f->samples); 00209 else 00210 memcpy(s->data + s->len, f->data, f->datalen); 00211 /* If either side is empty, reset the delivery time */ 00212 if (!s->len || ast_tvzero(f->delivery) || ast_tvzero(s->delivery)) /* XXX really ? */ 00213 s->delivery = f->delivery; 00214 s->len += f->datalen; 00215 return 0; 00216 }
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 600 of file frame.c.
References ast_format_list::desc.
Referenced by moh_alloc(), show_codec_n(), show_codec_n_deprecated(), show_codecs(), and show_codecs_deprecated().
00601 { 00602 int x; 00603 char *ret = "unknown"; 00604 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00605 if(AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == codec) { 00606 ret = AST_FORMAT_LIST[x].desc; 00607 break; 00608 } 00609 } 00610 return ret; 00611 }
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 1280 of file frame.c.
References ast_best_codec(), AST_FORMAT_AUDIO_MASK, ast_log(), ast_format_list::bits, LOG_DEBUG, option_debug, and ast_codec_pref::order.
Referenced by __oh323_new(), gtalk_new(), process_sdp(), sip_new(), and socket_process().
01281 { 01282 int x, ret = 0, slot; 01283 01284 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01285 slot = pref->order[x]; 01286 01287 if (!slot) 01288 break; 01289 if (formats & AST_FORMAT_LIST[slot-1].bits) { 01290 ret = AST_FORMAT_LIST[slot-1].bits; 01291 break; 01292 } 01293 } 01294 if(ret & AST_FORMAT_AUDIO_MASK) 01295 return ret; 01296 01297 if (option_debug > 3) 01298 ast_log(LOG_DEBUG, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec"); 01299 01300 return find_best ? ast_best_codec(formats) : 0; 01301 }
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 1539 of file frame.c.
References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G722, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_ILBC, AST_FORMAT_SLINEAR, AST_FORMAT_ULAW, ast_getformatname(), ast_log(), len, and LOG_WARNING.
Referenced by moh_generate(), and monmp3thread().
01540 { 01541 int len = 0; 01542 01543 /* XXX Still need speex, g723, and lpc10 XXX */ 01544 switch(format) { 01545 case AST_FORMAT_ILBC: 01546 len = (samples / 240) * 50; 01547 break; 01548 case AST_FORMAT_GSM: 01549 len = (samples / 160) * 33; 01550 break; 01551 case AST_FORMAT_G729A: 01552 len = samples / 8; 01553 break; 01554 case AST_FORMAT_SLINEAR: 01555 len = samples * 2; 01556 break; 01557 case AST_FORMAT_ULAW: 01558 case AST_FORMAT_ALAW: 01559 len = samples; 01560 break; 01561 case AST_FORMAT_G722: 01562 case AST_FORMAT_ADPCM: 01563 case AST_FORMAT_G726: 01564 case AST_FORMAT_G726_AAL2: 01565 len = samples / 2; 01566 break; 01567 default: 01568 ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format)); 01569 } 01570 01571 return len; 01572 }
int ast_codec_get_samples | ( | struct ast_frame * | f | ) |
Returns the number of samples contained in the frame.
Definition at line 1496 of file frame.c.
References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G722, AST_FORMAT_G723_1, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_ILBC, AST_FORMAT_LPC10, AST_FORMAT_SLINEAR, AST_FORMAT_SPEEX, AST_FORMAT_ULAW, ast_getformatname(), ast_log(), ast_frame::data, ast_frame::datalen, g723_samples(), LOG_WARNING, speex_samples(), and ast_frame::subclass.
Referenced by ast_rtp_read(), isAnsweringMachine(), moh_generate(), schedule_delivery(), and socket_process().
01497 { 01498 int samples=0; 01499 switch(f->subclass) { 01500 case AST_FORMAT_SPEEX: 01501 samples = speex_samples(f->data, f->datalen); 01502 break; 01503 case AST_FORMAT_G723_1: 01504 samples = g723_samples(f->data, f->datalen); 01505 break; 01506 case AST_FORMAT_ILBC: 01507 samples = 240 * (f->datalen / 50); 01508 break; 01509 case AST_FORMAT_GSM: 01510 samples = 160 * (f->datalen / 33); 01511 break; 01512 case AST_FORMAT_G729A: 01513 samples = f->datalen * 8; 01514 break; 01515 case AST_FORMAT_SLINEAR: 01516 samples = f->datalen / 2; 01517 break; 01518 case AST_FORMAT_LPC10: 01519 /* assumes that the RTP packet contains one LPC10 frame */ 01520 samples = 22 * 8; 01521 samples += (((char *)(f->data))[7] & 0x1) * 8; 01522 break; 01523 case AST_FORMAT_ULAW: 01524 case AST_FORMAT_ALAW: 01525 samples = f->datalen; 01526 break; 01527 case AST_FORMAT_G722: 01528 case AST_FORMAT_ADPCM: 01529 case AST_FORMAT_G726: 01530 case AST_FORMAT_G726_AAL2: 01531 samples = f->datalen * 2; 01532 break; 01533 default: 01534 ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass)); 01535 } 01536 return samples; 01537 }
static int ast_codec_interp_len | ( | int | format | ) | [inline, static] |
Gets duration in ms of interpolation frame for a format.
Definition at line 554 of file frame.h.
References AST_FORMAT_ILBC.
Referenced by __get_from_jb(), and jb_get_and_deliver().
00555 { 00556 return (format == AST_FORMAT_ILBC) ? 30 : 20; 00557 }
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 1139 of file frame.c.
References ast_codec_pref_remove(), and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow().
01140 { 01141 int x, newindex = 0; 01142 01143 ast_codec_pref_remove(pref, format); 01144 01145 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01146 if(AST_FORMAT_LIST[x].bits == format) { 01147 newindex = x + 1; 01148 break; 01149 } 01150 } 01151 01152 if(newindex) { 01153 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01154 if(!pref->order[x]) { 01155 pref->order[x] = newindex; 01156 break; 01157 } 01158 } 01159 } 01160 01161 return x; 01162 }
void ast_codec_pref_convert | ( | struct ast_codec_pref * | pref, | |
char * | buf, | |||
size_t | size, | |||
int | right | |||
) |
Shift an audio codec preference list up or down 65 bytes so that it becomes an ASCII string.
Definition at line 1041 of file frame.c.
References ast_codec_pref::order.
Referenced by check_access(), create_addr(), dump_prefs(), and socket_process().
01042 { 01043 int x, differential = (int) 'A', mem; 01044 char *from, *to; 01045 01046 if(right) { 01047 from = pref->order; 01048 to = buf; 01049 mem = size; 01050 } else { 01051 to = pref->order; 01052 from = buf; 01053 mem = 32; 01054 } 01055 01056 memset(to, 0, mem); 01057 for (x = 0; x < 32 ; x++) { 01058 if(!from[x]) 01059 break; 01060 to[x] = right ? (from[x] + differential) : (from[x] - differential); 01061 } 01062 }
struct ast_format_list ast_codec_pref_getsize | ( | struct ast_codec_pref * | pref, | |
int | format | |||
) | [read] |
Get packet size for codec.
Definition at line 1241 of file frame.c.
References 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_write(), handle_open_receive_channel_ack_message(), and transmit_connect().
01242 { 01243 int x, index = -1, framems = 0; 01244 struct ast_format_list fmt = {0}; 01245 01246 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01247 if(AST_FORMAT_LIST[x].bits == format) { 01248 fmt = AST_FORMAT_LIST[x]; 01249 index = x; 01250 break; 01251 } 01252 } 01253 01254 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01255 if(pref->order[x] == (index + 1)) { 01256 framems = pref->framing[x]; 01257 break; 01258 } 01259 } 01260 01261 /* size validation */ 01262 if(!framems) 01263 framems = AST_FORMAT_LIST[index].def_ms; 01264 01265 if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */ 01266 framems -= framems % AST_FORMAT_LIST[index].inc_ms; 01267 01268 if(framems < AST_FORMAT_LIST[index].min_ms) 01269 framems = AST_FORMAT_LIST[index].min_ms; 01270 01271 if(framems > AST_FORMAT_LIST[index].max_ms) 01272 framems = AST_FORMAT_LIST[index].max_ms; 01273 01274 fmt.cur_ms = framems; 01275 01276 return fmt; 01277 }
int ast_codec_pref_index | ( | struct ast_codec_pref * | pref, | |
int | index | |||
) |
Codec located at a particular place in the preference index See Audio Codec Preferences.
Definition at line 1099 of file frame.c.
References ast_format_list::bits, and ast_codec_pref::order.
Referenced by _sip_show_peer(), add_sdp(), ast_codec_pref_string(), function_iaxpeer(), function_sippeer(), gtalk_invite(), iax2_show_peer(), print_codec_to_cli(), and socket_process().
01100 { 01101 int slot = 0; 01102 01103 01104 if((index >= 0) && (index < sizeof(pref->order))) { 01105 slot = pref->order[index]; 01106 } 01107 01108 return slot ? AST_FORMAT_LIST[slot-1].bits : 0; 01109 }
void ast_codec_pref_init | ( | struct ast_codec_pref * | pref | ) |
Initialize an audio codec preference to "no preference" See Audio Codec Preferences.
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 1165 of file frame.c.
References ARRAY_LEN, ast_codec_pref::framing, and ast_codec_pref::order.
Referenced by create_addr().
01166 { 01167 int x, newindex = 0; 01168 01169 /* First step is to get the codecs "index number" */ 01170 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01171 if (AST_FORMAT_LIST[x].bits == format) { 01172 newindex = x + 1; 01173 break; 01174 } 01175 } 01176 /* Done if its unknown */ 01177 if (!newindex) 01178 return; 01179 01180 /* Now find any existing occurrence, or the end */ 01181 for (x = 0; x < 32; x++) { 01182 if (!pref->order[x] || pref->order[x] == newindex) 01183 break; 01184 } 01185 01186 if (only_if_existing && !pref->order[x]) 01187 return; 01188 01189 /* Move down to make space to insert - either all the way to the end, 01190 or as far as the existing location (which will be overwritten) */ 01191 for (; x > 0; x--) { 01192 pref->order[x] = pref->order[x - 1]; 01193 pref->framing[x] = pref->framing[x - 1]; 01194 } 01195 01196 /* And insert the new entry */ 01197 pref->order[0] = newindex; 01198 pref->framing[0] = 0; /* ? */ 01199 }
void ast_codec_pref_remove | ( | struct ast_codec_pref * | pref, | |
int | format | |||
) |
Remove audio a codec from a preference list.
Definition at line 1112 of file frame.c.
References ast_codec_pref::framing, and ast_codec_pref::order.
Referenced by ast_codec_pref_append(), and ast_parse_allow_disallow().
01113 { 01114 struct ast_codec_pref oldorder; 01115 int x, y = 0; 01116 int slot; 01117 int size; 01118 01119 if(!pref->order[0]) 01120 return; 01121 01122 memcpy(&oldorder, pref, sizeof(oldorder)); 01123 memset(pref, 0, sizeof(*pref)); 01124 01125 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01126 slot = oldorder.order[x]; 01127 size = oldorder.framing[x]; 01128 if(! slot) 01129 break; 01130 if(AST_FORMAT_LIST[slot-1].bits != format) { 01131 pref->order[y] = slot; 01132 pref->framing[y++] = size; 01133 } 01134 } 01135 01136 }
int ast_codec_pref_setsize | ( | struct ast_codec_pref * | pref, | |
int | format, | |||
int | framems | |||
) |
Set packet size for codec.
Definition at line 1202 of file frame.c.
References 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().
01203 { 01204 int x, index = -1; 01205 01206 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01207 if(AST_FORMAT_LIST[x].bits == format) { 01208 index = x; 01209 break; 01210 } 01211 } 01212 01213 if(index < 0) 01214 return -1; 01215 01216 /* size validation */ 01217 if(!framems) 01218 framems = AST_FORMAT_LIST[index].def_ms; 01219 01220 if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */ 01221 framems -= framems % AST_FORMAT_LIST[index].inc_ms; 01222 01223 if(framems < AST_FORMAT_LIST[index].min_ms) 01224 framems = AST_FORMAT_LIST[index].min_ms; 01225 01226 if(framems > AST_FORMAT_LIST[index].max_ms) 01227 framems = AST_FORMAT_LIST[index].max_ms; 01228 01229 01230 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01231 if(pref->order[x] == (index + 1)) { 01232 pref->framing[x] = framems; 01233 break; 01234 } 01235 } 01236 01237 return x; 01238 }
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 1064 of file frame.c.
References ast_codec_pref_index(), and ast_getformatname().
Referenced by dump_prefs(), and socket_process().
01065 { 01066 int x, codec; 01067 size_t total_len, slen; 01068 char *formatname; 01069 01070 memset(buf,0,size); 01071 total_len = size; 01072 buf[0] = '('; 01073 total_len--; 01074 for(x = 0; x < 32 ; x++) { 01075 if(total_len <= 0) 01076 break; 01077 if(!(codec = ast_codec_pref_index(pref,x))) 01078 break; 01079 if((formatname = ast_getformatname(codec))) { 01080 slen = strlen(formatname); 01081 if(slen > total_len) 01082 break; 01083 strncat(buf, formatname, total_len - 1); /* safe */ 01084 total_len -= slen; 01085 } 01086 if(total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) { 01087 strncat(buf, "|", total_len - 1); /* safe */ 01088 total_len--; 01089 } 01090 } 01091 if(total_len) { 01092 strncat(buf, ")", total_len - 1); /* safe */ 01093 total_len--; 01094 } 01095 01096 return size - total_len; 01097 }
static force_inline int ast_format_rate | ( | int | format | ) | [static] |
Get the sample rate for a given format.
Definition at line 581 of file frame.h.
References AST_FORMAT_G722.
Referenced by ast_read_generator_actions(), ast_readaudio_callback(), ast_readvideo_callback(), ast_rtp_read(), ast_translate(), calc_cost(), and generator_force().
00582 { 00583 if (format == AST_FORMAT_G722) 00584 return 16000; 00585 00586 return 8000; 00587 }
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 1574 of file frame.c.
References AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_slinear_saturated_divide(), ast_slinear_saturated_multiply(), ast_frame::data, ast_frame::frametype, ast_frame::samples, and ast_frame::subclass.
Referenced by audiohook_read_frame_single(), and conf_run().
01575 { 01576 int count; 01577 short *fdata = f->data; 01578 short adjust_value = abs(adjustment); 01579 01580 if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR)) 01581 return -1; 01582 01583 if (!adjustment) 01584 return 0; 01585 01586 for (count = 0; count < f->samples; count++) { 01587 if (adjustment > 0) { 01588 ast_slinear_saturated_multiply(&fdata[count], &adjust_value); 01589 } else if (adjustment < 0) { 01590 ast_slinear_saturated_divide(&fdata[count], &adjust_value); 01591 } 01592 } 01593 01594 return 0; 01595 }
void ast_frame_dump | ( | const char * | name, | |
struct ast_frame * | f, | |||
char * | prefix | |||
) |
Dump a frame for debugging purposes
Definition at line 754 of file frame.c.
References AST_CONTROL_ANSWER, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_FLASH, AST_CONTROL_HANGUP, AST_CONTROL_OFFHOOK, AST_CONTROL_OPTION, AST_CONTROL_RADIO_KEY, AST_CONTROL_RADIO_UNKEY, AST_CONTROL_RING, AST_CONTROL_RINGING, AST_CONTROL_TAKEOFFHOOK, AST_CONTROL_WINK, 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_verbose(), COLOR_BLACK, COLOR_BRCYAN, COLOR_BRGREEN, COLOR_BRMAGENTA, COLOR_BRRED, COLOR_YELLOW, ast_frame::data, ast_frame::frametype, ast_frame::subclass, and term_color().
Referenced by __ast_read(), and ast_write().
00755 { 00756 const char noname[] = "unknown"; 00757 char ftype[40] = "Unknown Frametype"; 00758 char cft[80]; 00759 char subclass[40] = "Unknown Subclass"; 00760 char csub[80]; 00761 char moreinfo[40] = ""; 00762 char cn[60]; 00763 char cp[40]; 00764 char cmn[40]; 00765 00766 if (!name) 00767 name = noname; 00768 00769 00770 if (!f) { 00771 ast_verbose("%s [ %s (NULL) ] [%s]\n", 00772 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00773 term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00774 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00775 return; 00776 } 00777 /* XXX We should probably print one each of voice and video when the format changes XXX */ 00778 if (f->frametype == AST_FRAME_VOICE) 00779 return; 00780 if (f->frametype == AST_FRAME_VIDEO) 00781 return; 00782 switch(f->frametype) { 00783 case AST_FRAME_DTMF_BEGIN: 00784 strcpy(ftype, "DTMF Begin"); 00785 subclass[0] = f->subclass; 00786 subclass[1] = '\0'; 00787 break; 00788 case AST_FRAME_DTMF_END: 00789 strcpy(ftype, "DTMF End"); 00790 subclass[0] = f->subclass; 00791 subclass[1] = '\0'; 00792 break; 00793 case AST_FRAME_CONTROL: 00794 strcpy(ftype, "Control"); 00795 switch(f->subclass) { 00796 case AST_CONTROL_HANGUP: 00797 strcpy(subclass, "Hangup"); 00798 break; 00799 case AST_CONTROL_RING: 00800 strcpy(subclass, "Ring"); 00801 break; 00802 case AST_CONTROL_RINGING: 00803 strcpy(subclass, "Ringing"); 00804 break; 00805 case AST_CONTROL_ANSWER: 00806 strcpy(subclass, "Answer"); 00807 break; 00808 case AST_CONTROL_BUSY: 00809 strcpy(subclass, "Busy"); 00810 break; 00811 case AST_CONTROL_TAKEOFFHOOK: 00812 strcpy(subclass, "Take Off Hook"); 00813 break; 00814 case AST_CONTROL_OFFHOOK: 00815 strcpy(subclass, "Line Off Hook"); 00816 break; 00817 case AST_CONTROL_CONGESTION: 00818 strcpy(subclass, "Congestion"); 00819 break; 00820 case AST_CONTROL_FLASH: 00821 strcpy(subclass, "Flash"); 00822 break; 00823 case AST_CONTROL_WINK: 00824 strcpy(subclass, "Wink"); 00825 break; 00826 case AST_CONTROL_OPTION: 00827 strcpy(subclass, "Option"); 00828 break; 00829 case AST_CONTROL_RADIO_KEY: 00830 strcpy(subclass, "Key Radio"); 00831 break; 00832 case AST_CONTROL_RADIO_UNKEY: 00833 strcpy(subclass, "Unkey Radio"); 00834 break; 00835 case -1: 00836 strcpy(subclass, "Stop generators"); 00837 break; 00838 default: 00839 snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass); 00840 } 00841 break; 00842 case AST_FRAME_NULL: 00843 strcpy(ftype, "Null Frame"); 00844 strcpy(subclass, "N/A"); 00845 break; 00846 case AST_FRAME_IAX: 00847 /* Should never happen */ 00848 strcpy(ftype, "IAX Specific"); 00849 snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass); 00850 break; 00851 case AST_FRAME_TEXT: 00852 strcpy(ftype, "Text"); 00853 strcpy(subclass, "N/A"); 00854 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00855 break; 00856 case AST_FRAME_IMAGE: 00857 strcpy(ftype, "Image"); 00858 snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass)); 00859 break; 00860 case AST_FRAME_HTML: 00861 strcpy(ftype, "HTML"); 00862 switch(f->subclass) { 00863 case AST_HTML_URL: 00864 strcpy(subclass, "URL"); 00865 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00866 break; 00867 case AST_HTML_DATA: 00868 strcpy(subclass, "Data"); 00869 break; 00870 case AST_HTML_BEGIN: 00871 strcpy(subclass, "Begin"); 00872 break; 00873 case AST_HTML_END: 00874 strcpy(subclass, "End"); 00875 break; 00876 case AST_HTML_LDCOMPLETE: 00877 strcpy(subclass, "Load Complete"); 00878 break; 00879 case AST_HTML_NOSUPPORT: 00880 strcpy(subclass, "No Support"); 00881 break; 00882 case AST_HTML_LINKURL: 00883 strcpy(subclass, "Link URL"); 00884 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00885 break; 00886 case AST_HTML_UNLINK: 00887 strcpy(subclass, "Unlink"); 00888 break; 00889 case AST_HTML_LINKREJECT: 00890 strcpy(subclass, "Link Reject"); 00891 break; 00892 default: 00893 snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass); 00894 break; 00895 } 00896 break; 00897 case AST_FRAME_MODEM: 00898 strcpy(ftype, "Modem"); 00899 switch (f->subclass) { 00900 case AST_MODEM_T38: 00901 strcpy(subclass, "T.38"); 00902 break; 00903 case AST_MODEM_V150: 00904 strcpy(subclass, "V.150"); 00905 break; 00906 default: 00907 snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass); 00908 break; 00909 } 00910 break; 00911 default: 00912 snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype); 00913 } 00914 if (!ast_strlen_zero(moreinfo)) 00915 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n", 00916 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00917 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00918 f->frametype, 00919 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00920 f->subclass, 00921 term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)), 00922 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00923 else 00924 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n", 00925 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00926 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00927 f->frametype, 00928 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00929 f->subclass, 00930 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00931 }
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 |
fr | Frame to free | |
cache | Whether to consider this frame for frame caching |
Definition at line 321 of file frame.c.
References ast_dsp_frame_freed(), AST_FRFLAG_FROM_DSP, AST_FRFLAG_FROM_TRANSLATOR, AST_LIST_INSERT_HEAD, AST_LIST_LOCK, AST_LIST_REMOVE, AST_LIST_UNLOCK, AST_MALLOCD_DATA, AST_MALLOCD_HDR, AST_MALLOCD_SRC, ast_test_flag, ast_translate_frame_freed(), ast_frame::data, FRAME_CACHE_MAX_SIZE, frames, free, ast_frame_cache::list, ast_frame::mallocd, ast_frame::offset, ast_frame_cache::size, and ast_frame::src.
Referenced by mixmonitor_thread().
00322 { 00323 if (ast_test_flag(fr, AST_FRFLAG_FROM_TRANSLATOR)) 00324 ast_translate_frame_freed(fr); 00325 else if (ast_test_flag(fr, AST_FRFLAG_FROM_DSP)) 00326 ast_dsp_frame_freed(fr); 00327 00328 if (!fr->mallocd) 00329 return; 00330 00331 #if !defined(LOW_MEMORY) 00332 if (cache && fr->mallocd == AST_MALLOCD_HDR) { 00333 /* Cool, only the header is malloc'd, let's just cache those for now 00334 * to keep things simple... */ 00335 struct ast_frame_cache *frames; 00336 00337 if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames))) 00338 && frames->size < FRAME_CACHE_MAX_SIZE) { 00339 AST_LIST_INSERT_HEAD(&frames->list, fr, frame_list); 00340 frames->size++; 00341 return; 00342 } 00343 } 00344 #endif 00345 00346 if (fr->mallocd & AST_MALLOCD_DATA) { 00347 if (fr->data) 00348 free(fr->data - fr->offset); 00349 } 00350 if (fr->mallocd & AST_MALLOCD_SRC) { 00351 if (fr->src) 00352 free((char *)fr->src); 00353 } 00354 if (fr->mallocd & AST_MALLOCD_HDR) { 00355 #ifdef TRACE_FRAMES 00356 AST_LIST_LOCK(&headerlist); 00357 headers--; 00358 AST_LIST_REMOVE(&headerlist, fr, frame_list); 00359 AST_LIST_UNLOCK(&headerlist); 00360 #endif 00361 free(fr); 00362 } 00363 }
Sums two frames of audio samples.
f1 | The first frame (which will contain the result) | |
f2 | The second frame |
Definition at line 1597 of file frame.c.
References AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_slinear_saturated_add(), ast_frame::data, ast_frame::frametype, ast_frame::samples, and ast_frame::subclass.
01598 { 01599 int count; 01600 short *data1, *data2; 01601 01602 if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR)) 01603 return -1; 01604 01605 if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR)) 01606 return -1; 01607 01608 if (f1->samples != f2->samples) 01609 return -1; 01610 01611 for (count = 0, data1 = f1->data, data2 = f2->data; 01612 count < f1->samples; 01613 count++, data1++, data2++) 01614 ast_slinear_saturated_add(data1, data2); 01615 01616 return 0; 01617 }
Copies a frame.
fr | frame to copy Duplicates a frame -- should only rarely be used, typically frisolate is good enough |
Definition at line 429 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_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::samples, ast_frame::seqno, ast_frame_cache::size, ast_frame::src, ast_frame::subclass, and ast_frame::ts.
Referenced by ast_jb_put(), ast_queue_frame(), ast_rtp_write(), ast_slinfactory_feed(), audiohook_read_frame_both(), audiohook_read_frame_single(), autoservice_run(), recordthread(), rpt(), and rpt_exec().
00430 { 00431 struct ast_frame *out = NULL; 00432 int len, srclen = 0; 00433 void *buf = NULL; 00434 00435 #if !defined(LOW_MEMORY) 00436 struct ast_frame_cache *frames; 00437 #endif 00438 00439 /* Start with standard stuff */ 00440 len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00441 /* If we have a source, add space for it */ 00442 /* 00443 * XXX Watch out here - if we receive a src which is not terminated 00444 * properly, we can be easily attacked. Should limit the size we deal with. 00445 */ 00446 if (f->src) 00447 srclen = strlen(f->src); 00448 if (srclen > 0) 00449 len += srclen + 1; 00450 00451 #if !defined(LOW_MEMORY) 00452 if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) { 00453 AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) { 00454 if (out->mallocd_hdr_len >= len) { 00455 size_t mallocd_len = out->mallocd_hdr_len; 00456 AST_LIST_REMOVE_CURRENT(&frames->list, frame_list); 00457 memset(out, 0, sizeof(*out)); 00458 out->mallocd_hdr_len = mallocd_len; 00459 buf = out; 00460 frames->size--; 00461 break; 00462 } 00463 } 00464 AST_LIST_TRAVERSE_SAFE_END 00465 } 00466 #endif 00467 00468 if (!buf) { 00469 if (!(buf = ast_calloc_cache(1, len))) 00470 return NULL; 00471 out = buf; 00472 out->mallocd_hdr_len = len; 00473 } 00474 00475 out->frametype = f->frametype; 00476 out->subclass = f->subclass; 00477 out->datalen = f->datalen; 00478 out->samples = f->samples; 00479 out->delivery = f->delivery; 00480 /* Set us as having malloc'd header only, so it will eventually 00481 get freed. */ 00482 out->mallocd = AST_MALLOCD_HDR; 00483 out->offset = AST_FRIENDLY_OFFSET; 00484 if (out->datalen) { 00485 out->data = buf + sizeof(*out) + AST_FRIENDLY_OFFSET; 00486 memcpy(out->data, f->data, out->datalen); 00487 } 00488 if (srclen > 0) { 00489 out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00490 /* Must have space since we allocated for it */ 00491 strcpy((char *)out->src, f->src); 00492 } 00493 ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO); 00494 out->ts = f->ts; 00495 out->len = f->len; 00496 out->seqno = f->seqno; 00497 return out; 00498 }
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 370 of file frame.c.
References ast_clear_flag, ast_copy_flags, ast_frame_header_new(), 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, free, ast_frame::len, ast_frame::mallocd, ast_frame::offset, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, and ast_frame::ts.
Referenced by jpeg_read_image().
00371 { 00372 struct ast_frame *out; 00373 void *newdata; 00374 00375 ast_clear_flag(fr, AST_FRFLAG_FROM_TRANSLATOR); 00376 ast_clear_flag(fr, AST_FRFLAG_FROM_DSP); 00377 00378 if (!(fr->mallocd & AST_MALLOCD_HDR)) { 00379 /* Allocate a new header if needed */ 00380 if (!(out = ast_frame_header_new())) 00381 return NULL; 00382 out->frametype = fr->frametype; 00383 out->subclass = fr->subclass; 00384 out->datalen = fr->datalen; 00385 out->samples = fr->samples; 00386 out->offset = fr->offset; 00387 out->data = fr->data; 00388 /* Copy the timing data */ 00389 ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO); 00390 if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) { 00391 out->ts = fr->ts; 00392 out->len = fr->len; 00393 out->seqno = fr->seqno; 00394 } 00395 } else 00396 out = fr; 00397 00398 if (!(fr->mallocd & AST_MALLOCD_SRC)) { 00399 if (fr->src) { 00400 if (!(out->src = ast_strdup(fr->src))) { 00401 if (out != fr) 00402 free(out); 00403 return NULL; 00404 } 00405 } 00406 } else 00407 out->src = fr->src; 00408 00409 if (!(fr->mallocd & AST_MALLOCD_DATA)) { 00410 if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) { 00411 if (out->src != fr->src) 00412 free((void *) out->src); 00413 if (out != fr) 00414 free(out); 00415 return NULL; 00416 } 00417 newdata += AST_FRIENDLY_OFFSET; 00418 out->offset = AST_FRIENDLY_OFFSET; 00419 out->datalen = fr->datalen; 00420 memcpy(newdata, fr->data, fr->datalen); 00421 out->data = newdata; 00422 } 00423 00424 out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA; 00425 00426 return out; 00427 }
struct ast_format_list* ast_get_format_list | ( | size_t * | size | ) | [read] |
Definition at line 516 of file frame.c.
00517 { 00518 *size = (sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0])); 00519 return AST_FORMAT_LIST; 00520 }
struct ast_format_list* ast_get_format_list_index | ( | int | index | ) | [read] |
int ast_getformatbyname | ( | const char * | name | ) |
Gets a format from a name.
name | string of format |
Definition at line 582 of file frame.c.
References ast_expand_codec_alias(), ast_format_list::bits, and format.
Referenced by ast_parse_allow_disallow(), iax_template_parse(), load_moh_classes(), reload_config(), and try_suggested_sip_codec().
00583 { 00584 int x, all, format = 0; 00585 00586 all = strcasecmp(name, "all") ? 0 : 1; 00587 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00588 if(AST_FORMAT_LIST[x].visible && (all || 00589 !strcasecmp(AST_FORMAT_LIST[x].name,name) || 00590 !strcasecmp(AST_FORMAT_LIST[x].name,ast_expand_codec_alias(name)))) { 00591 format |= AST_FORMAT_LIST[x].bits; 00592 if(!all) 00593 break; 00594 } 00595 } 00596 00597 return format; 00598 }
char* ast_getformatname | ( | int | format | ) |
Get the name of a format.
format | id of format |
Definition at line 522 of file frame.c.
References ast_format_list::bits, ast_format_list::name, and ast_format_list::visible.
Referenced by __ast_play_and_record(), __ast_read(), __ast_register_translator(), __login_exec(), _sip_show_peer(), add_codec_to_answer(), add_codec_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(), do_waiting(), eagi_exec(), func_channel_read(), function_iaxpeer(), function_sippeer(), gtalk_show_channels(), iax2_request(), iax2_show_channels(), iax2_show_peer(), iax_show_provisioning(), moh_classes_show(), moh_release(), oh323_rtp_read(), phone_setup(), print_codec_to_cli(), rebuild_matrix(), register_translator(), set_format(), set_peer_capabilities(), show_codecs(), show_codecs_deprecated(), show_file_formats(), show_file_formats_deprecated(), show_image_formats(), show_image_formats_deprecated(), show_translation(), show_translation_deprecated(), sip_request_call(), sip_rtp_read(), socket_process(), xagi_exec(), and zt_read().
00523 { 00524 int x; 00525 char *ret = "unknown"; 00526 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00527 if(AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == format) { 00528 ret = AST_FORMAT_LIST[x].name; 00529 break; 00530 } 00531 } 00532 return ret; 00533 }
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 535 of file frame.c.
References ast_format_list::bits, len, name, and ast_format_list::visible.
Referenced by __sip_show_channels(), _sip_show_peer(), add_sdp(), ast_streamfile(), function_iaxpeer(), function_sippeer(), gtalk_is_answered(), gtalk_newcall(), handle_showchan(), handle_showchan_deprecated(), iax2_bridge(), iax2_show_peer(), process_sdp(), serialize_showchan(), sip_new(), sip_request_call(), sip_show_channel(), sip_show_settings(), and sip_write().
00536 { 00537 int x; 00538 unsigned len; 00539 char *start, *end = buf; 00540 00541 if (!size) 00542 return buf; 00543 snprintf(end, size, "0x%x (", format); 00544 len = strlen(end); 00545 end += len; 00546 size -= len; 00547 start = end; 00548 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00549 if (AST_FORMAT_LIST[x].visible && (AST_FORMAT_LIST[x].bits & format)) { 00550 snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name); 00551 len = strlen(end); 00552 end += len; 00553 size -= len; 00554 } 00555 } 00556 if (start == end) 00557 snprintf(start, size, "nothing)"); 00558 else if (size > 1) 00559 *(end -1) = ')'; 00560 return buf; 00561 }
void 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 1303 of file frame.c.
References ast_codec_pref_append(), ast_codec_pref_remove(), ast_codec_pref_setsize(), AST_FORMAT_AUDIO_MASK, ast_getformatbyname(), ast_log(), ast_strdupa, format, LOG_DEBUG, LOG_WARNING, option_debug, parse(), and strsep().
Referenced by build_device(), build_peer(), build_user(), gtalk_create_member(), gtalk_load_config(), reload_config(), set_config(), and update_common_options().
01304 { 01305 char *parse = NULL, *this = NULL, *psize = NULL; 01306 int format = 0, framems = 0; 01307 01308 parse = ast_strdupa(list); 01309 while ((this = strsep(&parse, ","))) { 01310 framems = 0; 01311 if ((psize = strrchr(this, ':'))) { 01312 *psize++ = '\0'; 01313 if (option_debug) 01314 ast_log(LOG_DEBUG,"Packetization for codec: %s is %s\n", this, psize); 01315 framems = atoi(psize); 01316 if (framems < 0) 01317 framems = 0; 01318 } 01319 if (!(format = ast_getformatbyname(this))) { 01320 ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this); 01321 continue; 01322 } 01323 01324 if (mask) { 01325 if (allowing) 01326 *mask |= format; 01327 else 01328 *mask &= ~format; 01329 } 01330 01331 /* Set up a preference list for audio. Do not include video in preferences 01332 since we can not transcode video and have to use whatever is offered 01333 */ 01334 if (pref && (format & AST_FORMAT_AUDIO_MASK)) { 01335 if (strcasecmp(this, "all")) { 01336 if (allowing) { 01337 ast_codec_pref_append(pref, format); 01338 ast_codec_pref_setsize(pref, format, framems); 01339 } 01340 else 01341 ast_codec_pref_remove(pref, format); 01342 } else if (!allowing) { 01343 memset(pref, 0, sizeof(*pref)); 01344 } 01345 } 01346 } 01347 }
void ast_smoother_free | ( | struct ast_smoother * | s | ) |
Definition at line 268 of file frame.c.
References free.
Referenced by ast_rtp_codec_setpref(), ast_rtp_destroy(), and ast_rtp_write().
00269 { 00270 free(s); 00271 }
int ast_smoother_get_flags | ( | struct ast_smoother * | smoother | ) |
Definition at line 157 of file frame.c.
References ast_smoother::flags.
00158 { 00159 return s->flags; 00160 }
struct ast_smoother* ast_smoother_new | ( | int | bytes | ) | [read] |
Definition at line 147 of file frame.c.
References ast_malloc, ast_smoother_reset(), and s.
Referenced by ast_rtp_write().
00148 { 00149 struct ast_smoother *s; 00150 if (size < 1) 00151 return NULL; 00152 if ((s = ast_malloc(sizeof(*s)))) 00153 ast_smoother_reset(s, size); 00154 return s; 00155 }
struct ast_frame* ast_smoother_read | ( | struct ast_smoother * | s | ) | [read] |
Definition at line 218 of file frame.c.
References AST_FRAME_VOICE, AST_FRIENDLY_OFFSET, ast_log(), AST_SMOOTHER_FLAG_G729, ast_tvadd(), ast_smoother::data, ast_frame::data, ast_frame::datalen, ast_smoother::delivery, ast_frame::delivery, ast_smoother::f, ast_smoother::flags, ast_smoother::format, ast_smoother::framedata, ast_frame::frametype, ast_smoother::len, len, LOG_WARNING, ast_frame::offset, ast_smoother::opt, ast_frame::samples, ast_smoother::samplesperbyte, ast_smoother::size, and ast_frame::subclass.
Referenced by ast_rtp_write().
00219 { 00220 struct ast_frame *opt; 00221 int len; 00222 00223 /* IF we have an optimization frame, send it */ 00224 if (s->opt) { 00225 if (s->opt->offset < AST_FRIENDLY_OFFSET) 00226 ast_log(LOG_WARNING, "Returning a frame of inappropriate offset (%d).\n", 00227 s->opt->offset); 00228 opt = s->opt; 00229 s->opt = NULL; 00230 return opt; 00231 } 00232 00233 /* Make sure we have enough data */ 00234 if (s->len < s->size) { 00235 /* Or, if this is a G.729 frame with VAD on it, send it immediately anyway */ 00236 if (!((s->flags & AST_SMOOTHER_FLAG_G729) && (s->size % 10))) 00237 return NULL; 00238 } 00239 len = s->size; 00240 if (len > s->len) 00241 len = s->len; 00242 /* Make frame */ 00243 s->f.frametype = AST_FRAME_VOICE; 00244 s->f.subclass = s->format; 00245 s->f.data = s->framedata + AST_FRIENDLY_OFFSET; 00246 s->f.offset = AST_FRIENDLY_OFFSET; 00247 s->f.datalen = len; 00248 /* Samples will be improper given VAD, but with VAD the concept really doesn't even exist */ 00249 s->f.samples = len * s->samplesperbyte; /* XXX rounding */ 00250 s->f.delivery = s->delivery; 00251 /* Fill Data */ 00252 memcpy(s->f.data, s->data, len); 00253 s->len -= len; 00254 /* Move remaining data to the front if applicable */ 00255 if (s->len) { 00256 /* In principle this should all be fine because if we are sending 00257 G.729 VAD, the next timestamp will take over anyawy */ 00258 memmove(s->data, s->data + len, s->len); 00259 if (!ast_tvzero(s->delivery)) { 00260 /* If we have delivery time, increment it, otherwise, leave it at 0 */ 00261 s->delivery = ast_tvadd(s->delivery, ast_samp2tv(s->f.samples, 8000)); 00262 } 00263 } 00264 /* Return frame */ 00265 return &s->f; 00266 }
void ast_smoother_reset | ( | struct ast_smoother * | s, | |
int | bytes | |||
) |
Definition at line 141 of file frame.c.
References ast_smoother::size.
Referenced by ast_smoother_new().
00142 { 00143 memset(s, 0, sizeof(*s)); 00144 s->size = size; 00145 }
void ast_smoother_set_flags | ( | struct ast_smoother * | smoother, | |
int | flags | |||
) |
Definition at line 162 of file frame.c.
References ast_smoother::flags.
Referenced by ast_rtp_write().
int ast_smoother_test_flag | ( | struct ast_smoother * | s, | |
int | flag | |||
) |
Definition at line 167 of file frame.c.
References ast_smoother::flags.
Referenced by ast_rtp_write().
00168 { 00169 return (s->flags & flag); 00170 }
void ast_swapcopy_samples | ( | void * | dst, | |
const void * | src, | |||
int | samples | |||
) |
Definition at line 500 of file frame.c.
Referenced by __ast_smoother_feed(), iax_frame_wrap(), and phone_write_buf().
00501 { 00502 int i; 00503 unsigned short *dst_s = dst; 00504 const unsigned short *src_s = src; 00505 00506 for (i = 0; i < samples; i++) 00507 dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8); 00508 }
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 139 of file frame.c.
Referenced by __ast_read(), __oh323_rtp_create(), __oh323_update_info(), agent_new(), agent_read(), ast_channel_masquerade(), ast_channel_setwhentohangup(), ast_do_masquerade(), ast_rtcp_read(), ast_rtp_read(), ast_softhangup_nolock(), ast_udptl_read(), conf_run(), features_read(), gtalk_rtp_read(), handle_request_invite(), handle_response_invite(), iax2_read(), local_read(), mgcp_rtp_read(), oh323_read(), oh323_rtp_read(), process_rfc2833(), process_sdp(), send_dtmf(), sip_read(), sip_rtp_read(), skinny_rtp_read(), and wakeup_sub().