#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), AST_FRFLAG_FROM_FILESTREAM = (1 << 3) } |
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_ATXFERCMD = 19, 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. | |
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) |
ast_frame * | ast_frame_enqueue (struct ast_frame *head, struct ast_frame *f, int maxlen, int dupe) |
Appends a frame to the end of a list of frames, truncating the maximum length of the list. | |
void | ast_frame_free (struct ast_frame *fr, int cache) |
Requests a frame to be allocated Frees a frame. | |
int | ast_frame_slinear_sum (struct ast_frame *f1, struct ast_frame *f2) |
Sums two frames of audio samples. | |
ast_frame * | ast_frdup (const struct ast_frame *fr) |
Copies a frame. | |
ast_frame * | ast_frisolate (struct ast_frame *fr) |
Makes a frame independent of any static storage. | |
ast_format_list * | ast_get_format_list (size_t *size) |
ast_format_list * | ast_get_format_list_index (int index) |
int | ast_getformatbyname (const char *name) |
Gets a format from a name. | |
char * | ast_getformatname (int format) |
Get the name of a format. | |
char * | ast_getformatname_multiple (char *buf, size_t size, int format) |
Get the names of a set of formats. | |
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) |
ast_smoother * | ast_smoother_new (int bytes) |
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 | |
ast_frame | ast_null_frame |
Definition in file frame.h.
#define AST_FORMAT_ADPCM (1 << 5) |
ADPCM (IMA)
Definition at line 248 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 244 of file frame.h.
Referenced by alawtolin_sample(), alawtoulaw_sample(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), cb_events(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_new(), dahdi_read(), dahdi_write(), misdn_read(), misdn_set_opt_exec(), oh323_rtp_read(), pcm_seek(), pcm_write(), read_config(), and sms_generate().
#define AST_FORMAT_AUDIO_MASK ((1 << 16)-1) |
Maximum audio mask
Definition at line 266 of file frame.h.
Referenced by add_sdp(), ast_best_codec(), ast_codec_choose(), ast_openstream_full(), ast_parse_allow_disallow(), ast_request(), 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 262 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(), g722tolin_sample(), and pcm_read().
#define AST_FORMAT_G723_1 (1 << 0) |
G.723.1 compression
Definition at line 238 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 260 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 246 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 254 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(), and zap_translate().
#define AST_FORMAT_GSM (1 << 1) |
GSM compression
Definition at line 240 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 272 of file frame.h.
Referenced by codec_ast2skinny(), and codec_skinny2ast().
#define AST_FORMAT_H263 (1 << 19) |
H.263 Video
Definition at line 274 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 258 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_codec_interp_len(), convertcap(), ilbc_read(), ilbc_write(), and ilbctolin_sample().
#define AST_FORMAT_JPEG (1 << 16) |
JPEG Images
Definition at line 268 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 252 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 264 of file frame.h.
Referenced by add_sdp(), ast_filehelper(), ast_openvstream(), ast_playstream(), ast_rtp_read(), ast_translate_available_formats(), ast_writestream(), filestream_destructor(), 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 280 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 250 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_slinfactory_feed(), attempt_reconnect(), audio_audiohook_write_list(), audiohook_read_frame_both(), audiohook_read_frame_single(), background_detect_exec(), build_conf(), chanspy_exec(), conf_run(), connect_link(), dahdi_new(), dahdi_read(), dahdi_write(), dictate_exec(), do_waiting(), eagi_exec(), extenspy_exec(), handle_recordfile(), iax_frame_wrap(), ices_exec(), init_outgoing(), isAnsweringMachine(), linear_alloc(), linear_generator(), lintoadpcm_sample(), lintoalaw_sample(), lintog722_sample(), lintog726_sample(), lintogsm_sample(), lintoilbc_sample(), lintolpc10_sample(), lintospeex_sample(), lintoulaw_sample(), load_module(), 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(), read_config(), rpt(), rpt_call(), rpt_tele_thread(), send_waveform_to_channel(), silence_generator_generate(), slinear_read(), slinear_write(), sms_generate(), socket_process(), speech_background(), speech_create(), spy_generate(), tonepair_alloc(), wav_read(), and wav_write().
#define AST_FORMAT_SPEEX (1 << 9) |
SpeeX Free Compression
Definition at line 256 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 242 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(), dahdi_new(), dahdi_read(), dahdi_write(), disa_exec(), load_module(), milliwatt_generate(), oh323_rtp_read(), old_milliwatt_exec(), phone_request(), phone_setup(), phone_write(), pri_dchannel(), send_tone_burst(), ulawtoalaw_sample(), and ulawtolin_sample().
#define AST_FORMAT_VIDEO_MASK (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) |
Definition at line 281 of file frame.h.
Referenced by add_sdp(), ast_request(), 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 __action_dialoffhook(), __adsi_transmit_messages(), __ast_play_and_record(), agent_ack_sleep(), app_exec(), ast_audiohook_write_list(), ast_bridge_call(), ast_dsp_process(), ast_feature_request_and_dial(), ast_jb_put(), background_detect_exec(), cb_events(), channel_spy(), conf_exec(), conf_run(), console_dial(), console_dial_deprecated(), dahdi_bridge(), dahdi_read(), dictate_exec(), disa_exec(), do_immediate_setup(), echo_exec(), gtalk_handle_dtmf(), handle_recordfile(), handle_request(), handle_request_info(), mgcp_rtp_read(), misdn_bridge(), mp3_exec(), NBScat_exec(), oh323_rtp_read(), phone_exception(), process_ast_dsp(), receive_dtmf_digits(), rpt(), rpt_call(), send_waveform_to_channel(), sip_rtp_read(), speech_background(), ss_thread(), wait_for_answer(), and wait_for_winner().
#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 187 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 408 of file frame.h.
Referenced by __adsi_transmit_messages(), __ast_play_and_record(), __ast_queue_frame(), __ast_read(), __ast_request_and_dial(), 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_jb_destroy(), ast_jb_put(), 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(), calc_cost(), channel_spy(), check_goto_on_transfer(), conf_exec(), conf_flush(), conf_free(), conf_run(), create_jb(), dahdi_bridge(), dictate_exec(), disa_exec(), do_atxfer(), do_idle_thread(), do_parking_thread(), do_waiting(), echo_exec(), find_cache(), gen_generate(), handle_invite_replaces(), handle_recordfile(), iax_park_thread(), ices_exec(), isAnsweringMachine(), jb_empty_and_reset_adaptive(), jb_empty_and_reset_fixed(), jb_get_and_deliver(), masq_park_call(), measurenoise(), moh_files_generator(), monitor_dial(), mp3_exec(), NBScat_exec(), receive_dtmf_digits(), recordthread(), rpt(), run_agi(), send_tone_burst(), send_waveform_to_channel(), sendurl_exec(), speech_background(), spy_generate(), ss_thread(), wait_for_answer(), wait_for_hangup(), wait_for_winner(), waitforring_exec(), and waitstream_core().
#define AST_FRIENDLY_OFFSET 64 |
Definition at line 198 of file frame.h.
Referenced by __get_from_jb(), alsa_read(), ast_frdup(), ast_frisolate(), ast_prod(), ast_rtcp_read(), ast_rtp_read(), ast_smoother_read(), ast_trans_frameout(), ast_udptl_read(), conf_run(), dahdi_read(), g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), iax_frame_wrap(), ilbc_read(), jb_get_and_deliver(), linear_generator(), milliwatt_generate(), moh_generate(), mohalloc(), mp3_exec(), NBScat_exec(), newpvt(), ogg_vorbis_read(), oss_read(), pcm_read(), phone_read(), process_rfc3389(), send_tone_burst(), send_waveform_to_channel(), slinear_read(), sms_generate(), vox_read(), wav_read(), and zap_frameout().
#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 226 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 228 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 218 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 206 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 204 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 208 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 212 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 328 of file frame.h.
Referenced by dahdi_hangup(), and dahdi_setoption().
#define AST_OPTION_ECHOCAN 8 |
Explicitly enable or disable echo cancelation for the given channel
Definition at line 350 of file frame.h.
Referenced by dahdi_setoption().
#define AST_OPTION_FLAG_REQUEST 0 |
#define AST_OPTION_OPRMODE 7 |
#define AST_OPTION_RELAXDTMF 3 |
Relax the parameters for DTMF reception (mainly for radio use)
Definition at line 325 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 344 of file frame.h.
Referenced by dahdi_setoption(), func_channel_write(), iax2_setoption(), play_record_review(), reset_volumes(), set_talk_volume(), and vm_forwardoptions().
#define AST_OPTION_TDD 2 |
Put a compatible channel into TDD (TTY for the hearing-impared) mode
Definition at line 322 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 319 of file frame.h.
Referenced by conf_run(), dahdi_hangup(), dahdi_setoption(), rpt(), 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 336 of file frame.h.
Referenced by common_exec(), dahdi_setoption(), func_channel_write(), iax2_setoption(), reset_volumes(), and set_listen_volume().
#define AST_SMOOTHER_FLAG_BE (1 << 1) |
#define AST_SMOOTHER_FLAG_G729 (1 << 0) |
Definition at line 306 of file frame.h.
Referenced by __ast_smoother_feed(), and ast_smoother_read().
anonymous enum |
Definition at line 127 of file frame.h.
00127 { 00128 /*! This frame contains valid timing information */ 00129 AST_FRFLAG_HAS_TIMING_INFO = (1 << 0), 00130 /*! 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 /*! This frame came from a filestream and is still the original frame. 00139 * The filestream cannot be free'd if the frame inside of it still has 00140 * this flag set. */ 00141 AST_FRFLAG_FROM_FILESTREAM = (1 << 3), 00142 };
Definition at line 283 of file frame.h.
00283 { 00284 AST_CONTROL_HANGUP = 1, /*!< Other end has hungup */ 00285 AST_CONTROL_RING = 2, /*!< Local ring */ 00286 AST_CONTROL_RINGING = 3, /*!< Remote end is ringing */ 00287 AST_CONTROL_ANSWER = 4, /*!< Remote end has answered */ 00288 AST_CONTROL_BUSY = 5, /*!< Remote end is busy */ 00289 AST_CONTROL_TAKEOFFHOOK = 6, /*!< Make it go off hook */ 00290 AST_CONTROL_OFFHOOK = 7, /*!< Line is off hook */ 00291 AST_CONTROL_CONGESTION = 8, /*!< Congestion (circuits busy) */ 00292 AST_CONTROL_FLASH = 9, /*!< Flash hook */ 00293 AST_CONTROL_WINK = 10, /*!< Wink */ 00294 AST_CONTROL_OPTION = 11, /*!< Set a low-level option */ 00295 AST_CONTROL_RADIO_KEY = 12, /*!< Key Radio */ 00296 AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */ 00297 AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */ 00298 AST_CONTROL_PROCEEDING = 15, /*!< Indicate CALL PROCEEDING */ 00299 AST_CONTROL_HOLD = 16, /*!< Indicate call is placed on hold */ 00300 AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold */ 00301 AST_CONTROL_VIDUPDATE = 18, /*!< Indicate video frame update */ 00302 AST_CONTROL_ATXFERCMD = 19, /*!< AMI triggered attended transfer */ 00303 AST_CONTROL_SRCUPDATE = 20, /*!< Indicate source of media has changed */ 00304 };
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 173 of file frame.c.
References AST_FRAME_VOICE, ast_log(), AST_MIN_OFFSET, AST_SMOOTHER_FLAG_G729, ast_swapcopy_samples(), ast_tvzero(), f, LOG_NOTICE, LOG_WARNING, s, and SMOOTHER_SIZE.
00174 { 00175 if (f->frametype != AST_FRAME_VOICE) { 00176 ast_log(LOG_WARNING, "Huh? Can't smooth a non-voice frame!\n"); 00177 return -1; 00178 } 00179 if (!s->format) { 00180 s->format = f->subclass; 00181 s->samplesperbyte = (float)f->samples / (float)f->datalen; 00182 } else if (s->format != f->subclass) { 00183 ast_log(LOG_WARNING, "Smoother was working on %d format frames, now trying to feed %d?\n", s->format, f->subclass); 00184 return -1; 00185 } 00186 if (s->len + f->datalen > SMOOTHER_SIZE) { 00187 ast_log(LOG_WARNING, "Out of smoother space\n"); 00188 return -1; 00189 } 00190 if (((f->datalen == s->size) || ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) 00191 && !s->opt && (f->offset >= AST_MIN_OFFSET)) { 00192 if (!s->len) { 00193 /* Optimize by sending the frame we just got 00194 on the next read, thus eliminating the douple 00195 copy */ 00196 if (swap) 00197 ast_swapcopy_samples(f->data, f->data, f->samples); 00198 s->opt = f; 00199 return 0; 00200 } 00201 } 00202 if (s->flags & AST_SMOOTHER_FLAG_G729) { 00203 if (s->len % 10) { 00204 ast_log(LOG_NOTICE, "Dropping extra frame of G.729 since we already have a VAD frame at the end\n"); 00205 return 0; 00206 } 00207 } 00208 if (swap) 00209 ast_swapcopy_samples(s->data+s->len, f->data, f->samples); 00210 else 00211 memcpy(s->data + s->len, f->data, f->datalen); 00212 /* If either side is empty, reset the delivery time */ 00213 if (!s->len || ast_tvzero(f->delivery) || ast_tvzero(s->delivery)) /* XXX really ? */ 00214 s->delivery = f->delivery; 00215 s->len += f->datalen; 00216 return 0; 00217 }
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 607 of file frame.c.
References AST_FORMAT_LIST, and desc.
Referenced by moh_alloc(), show_codec_n(), show_codec_n_deprecated(), show_codecs(), and show_codecs_deprecated().
00608 { 00609 int x; 00610 char *ret = "unknown"; 00611 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00612 if(AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == codec) { 00613 ret = AST_FORMAT_LIST[x].desc; 00614 break; 00615 } 00616 } 00617 return ret; 00618 }
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 1282 of file frame.c.
References ast_best_codec(), AST_FORMAT_AUDIO_MASK, AST_FORMAT_LIST, 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().
01283 { 01284 int x, ret = 0, slot; 01285 01286 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01287 slot = pref->order[x]; 01288 01289 if (!slot) 01290 break; 01291 if (formats & AST_FORMAT_LIST[slot-1].bits) { 01292 ret = AST_FORMAT_LIST[slot-1].bits; 01293 break; 01294 } 01295 } 01296 if(ret & AST_FORMAT_AUDIO_MASK) 01297 return ret; 01298 01299 if (option_debug > 3) 01300 ast_log(LOG_DEBUG, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec"); 01301 01302 return find_best ? ast_best_codec(formats) : 0; 01303 }
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 1541 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().
01542 { 01543 int len = 0; 01544 01545 /* XXX Still need speex, g723, and lpc10 XXX */ 01546 switch(format) { 01547 case AST_FORMAT_ILBC: 01548 len = (samples / 240) * 50; 01549 break; 01550 case AST_FORMAT_GSM: 01551 len = (samples / 160) * 33; 01552 break; 01553 case AST_FORMAT_G729A: 01554 len = samples / 8; 01555 break; 01556 case AST_FORMAT_SLINEAR: 01557 len = samples * 2; 01558 break; 01559 case AST_FORMAT_ULAW: 01560 case AST_FORMAT_ALAW: 01561 len = samples; 01562 break; 01563 case AST_FORMAT_G722: 01564 case AST_FORMAT_ADPCM: 01565 case AST_FORMAT_G726: 01566 case AST_FORMAT_G726_AAL2: 01567 len = samples / 2; 01568 break; 01569 default: 01570 ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format)); 01571 } 01572 01573 return len; 01574 }
int ast_codec_get_samples | ( | struct ast_frame * | f | ) |
Returns the number of samples contained in the frame.
Definition at line 1498 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(), f, g723_samples(), LOG_WARNING, and speex_samples().
Referenced by ast_rtp_read(), isAnsweringMachine(), moh_generate(), schedule_delivery(), and socket_process().
01499 { 01500 int samples=0; 01501 switch(f->subclass) { 01502 case AST_FORMAT_SPEEX: 01503 samples = speex_samples(f->data, f->datalen); 01504 break; 01505 case AST_FORMAT_G723_1: 01506 samples = g723_samples(f->data, f->datalen); 01507 break; 01508 case AST_FORMAT_ILBC: 01509 samples = 240 * (f->datalen / 50); 01510 break; 01511 case AST_FORMAT_GSM: 01512 samples = 160 * (f->datalen / 33); 01513 break; 01514 case AST_FORMAT_G729A: 01515 samples = f->datalen * 8; 01516 break; 01517 case AST_FORMAT_SLINEAR: 01518 samples = f->datalen / 2; 01519 break; 01520 case AST_FORMAT_LPC10: 01521 /* assumes that the RTP packet contains one LPC10 frame */ 01522 samples = 22 * 8; 01523 samples += (((char *)(f->data))[7] & 0x1) * 8; 01524 break; 01525 case AST_FORMAT_ULAW: 01526 case AST_FORMAT_ALAW: 01527 samples = f->datalen; 01528 break; 01529 case AST_FORMAT_G722: 01530 case AST_FORMAT_ADPCM: 01531 case AST_FORMAT_G726: 01532 case AST_FORMAT_G726_AAL2: 01533 samples = f->datalen * 2; 01534 break; 01535 default: 01536 ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass)); 01537 } 01538 return samples; 01539 }
static int ast_codec_interp_len | ( | int | format | ) | [inline, static] |
Gets duration in ms of interpolation frame for a format.
Definition at line 559 of file frame.h.
References AST_FORMAT_ILBC.
Referenced by __get_from_jb(), and jb_get_and_deliver().
00560 { 00561 return (format == AST_FORMAT_ILBC) ? 30 : 20; 00562 }
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 1141 of file frame.c.
References ast_codec_pref_remove(), AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow().
01142 { 01143 int x, newindex = 0; 01144 01145 ast_codec_pref_remove(pref, format); 01146 01147 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01148 if(AST_FORMAT_LIST[x].bits == format) { 01149 newindex = x + 1; 01150 break; 01151 } 01152 } 01153 01154 if(newindex) { 01155 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01156 if(!pref->order[x]) { 01157 pref->order[x] = newindex; 01158 break; 01159 } 01160 } 01161 } 01162 01163 return x; 01164 }
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 1043 of file frame.c.
References ast_codec_pref::order.
Referenced by check_access(), create_addr(), dump_prefs(), and socket_process().
01044 { 01045 int x, differential = (int) 'A', mem; 01046 char *from, *to; 01047 01048 if(right) { 01049 from = pref->order; 01050 to = buf; 01051 mem = size; 01052 } else { 01053 to = pref->order; 01054 from = buf; 01055 mem = 32; 01056 } 01057 01058 memset(to, 0, mem); 01059 for (x = 0; x < 32 ; x++) { 01060 if(!from[x]) 01061 break; 01062 to[x] = right ? (from[x] + differential) : (from[x] - differential); 01063 } 01064 }
struct ast_format_list ast_codec_pref_getsize | ( | struct ast_codec_pref * | pref, | |
int | format | |||
) |
Get packet size for codec.
Definition at line 1243 of file frame.c.
References AST_FORMAT_LIST, ast_format_list::bits, and format.
Referenced by add_codec_to_sdp(), ast_rtp_bridge(), ast_rtp_write(), handle_open_receive_channel_ack_message(), and transmit_connect().
01244 { 01245 int x, index = -1, framems = 0; 01246 struct ast_format_list fmt = {0}; 01247 01248 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01249 if(AST_FORMAT_LIST[x].bits == format) { 01250 fmt = AST_FORMAT_LIST[x]; 01251 index = x; 01252 break; 01253 } 01254 } 01255 01256 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01257 if(pref->order[x] == (index + 1)) { 01258 framems = pref->framing[x]; 01259 break; 01260 } 01261 } 01262 01263 /* size validation */ 01264 if(!framems) 01265 framems = AST_FORMAT_LIST[index].def_ms; 01266 01267 if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */ 01268 framems -= framems % AST_FORMAT_LIST[index].inc_ms; 01269 01270 if(framems < AST_FORMAT_LIST[index].min_ms) 01271 framems = AST_FORMAT_LIST[index].min_ms; 01272 01273 if(framems > AST_FORMAT_LIST[index].max_ms) 01274 framems = AST_FORMAT_LIST[index].max_ms; 01275 01276 fmt.cur_ms = framems; 01277 01278 return fmt; 01279 }
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 1101 of file frame.c.
References AST_FORMAT_LIST, ast_format_list::bits, and ast_codec_pref::order.
Referenced by _sip_show_peer(), add_sdp(), ast_codec_pref_string(), function_iaxpeer(), function_sippeer(), gtalk_invite(), iax2_show_peer(), print_codec_to_cli(), and socket_process().
01102 { 01103 int slot = 0; 01104 01105 01106 if((index >= 0) && (index < sizeof(pref->order))) { 01107 slot = pref->order[index]; 01108 } 01109 01110 return slot ? AST_FORMAT_LIST[slot-1].bits : 0; 01111 }
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 1167 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_codec_pref::framing, and ast_codec_pref::order.
Referenced by create_addr().
01168 { 01169 int x, newindex = 0; 01170 01171 /* First step is to get the codecs "index number" */ 01172 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01173 if (AST_FORMAT_LIST[x].bits == format) { 01174 newindex = x + 1; 01175 break; 01176 } 01177 } 01178 /* Done if its unknown */ 01179 if (!newindex) 01180 return; 01181 01182 /* Now find any existing occurrence, or the end */ 01183 for (x = 0; x < 32; x++) { 01184 if (!pref->order[x] || pref->order[x] == newindex) 01185 break; 01186 } 01187 01188 if (only_if_existing && !pref->order[x]) 01189 return; 01190 01191 /* Move down to make space to insert - either all the way to the end, 01192 or as far as the existing location (which will be overwritten) */ 01193 for (; x > 0; x--) { 01194 pref->order[x] = pref->order[x - 1]; 01195 pref->framing[x] = pref->framing[x - 1]; 01196 } 01197 01198 /* And insert the new entry */ 01199 pref->order[0] = newindex; 01200 pref->framing[0] = 0; /* ? */ 01201 }
void ast_codec_pref_remove | ( | struct ast_codec_pref * | pref, | |
int | format | |||
) |
Remove audio a codec from a preference list.
Definition at line 1114 of file frame.c.
References AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_codec_pref_append(), and ast_parse_allow_disallow().
01115 { 01116 struct ast_codec_pref oldorder; 01117 int x, y = 0; 01118 int slot; 01119 int size; 01120 01121 if(!pref->order[0]) 01122 return; 01123 01124 memcpy(&oldorder, pref, sizeof(oldorder)); 01125 memset(pref, 0, sizeof(*pref)); 01126 01127 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01128 slot = oldorder.order[x]; 01129 size = oldorder.framing[x]; 01130 if(! slot) 01131 break; 01132 if(AST_FORMAT_LIST[slot-1].bits != format) { 01133 pref->order[y] = slot; 01134 pref->framing[y++] = size; 01135 } 01136 } 01137 01138 }
int ast_codec_pref_setsize | ( | struct ast_codec_pref * | pref, | |
int | format, | |||
int | framems | |||
) |
Set packet size for codec.
Definition at line 1204 of file frame.c.
References AST_FORMAT_LIST, ast_codec_pref::framing, and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow(), and process_sdp().
01205 { 01206 int x, index = -1; 01207 01208 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01209 if(AST_FORMAT_LIST[x].bits == format) { 01210 index = x; 01211 break; 01212 } 01213 } 01214 01215 if(index < 0) 01216 return -1; 01217 01218 /* size validation */ 01219 if(!framems) 01220 framems = AST_FORMAT_LIST[index].def_ms; 01221 01222 if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */ 01223 framems -= framems % AST_FORMAT_LIST[index].inc_ms; 01224 01225 if(framems < AST_FORMAT_LIST[index].min_ms) 01226 framems = AST_FORMAT_LIST[index].min_ms; 01227 01228 if(framems > AST_FORMAT_LIST[index].max_ms) 01229 framems = AST_FORMAT_LIST[index].max_ms; 01230 01231 01232 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01233 if(pref->order[x] == (index + 1)) { 01234 pref->framing[x] = framems; 01235 break; 01236 } 01237 } 01238 01239 return x; 01240 }
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 1066 of file frame.c.
References ast_codec_pref_index(), and ast_getformatname().
Referenced by dump_prefs(), and socket_process().
01067 { 01068 int x, codec; 01069 size_t total_len, slen; 01070 char *formatname; 01071 01072 memset(buf,0,size); 01073 total_len = size; 01074 buf[0] = '('; 01075 total_len--; 01076 for(x = 0; x < 32 ; x++) { 01077 if(total_len <= 0) 01078 break; 01079 if(!(codec = ast_codec_pref_index(pref,x))) 01080 break; 01081 if((formatname = ast_getformatname(codec))) { 01082 slen = strlen(formatname); 01083 if(slen > total_len) 01084 break; 01085 strncat(buf, formatname, total_len - 1); /* safe */ 01086 total_len -= slen; 01087 } 01088 if(total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) { 01089 strncat(buf, "|", total_len - 1); /* safe */ 01090 total_len--; 01091 } 01092 } 01093 if(total_len) { 01094 strncat(buf, ")", total_len - 1); /* safe */ 01095 total_len--; 01096 } 01097 01098 return size - total_len; 01099 }
static force_inline int ast_format_rate | ( | int | format | ) | [static] |
Get the sample rate for a given format.
Definition at line 586 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().
00587 { 00588 if (format == AST_FORMAT_G722) 00589 return 16000; 00590 00591 return 8000; 00592 }
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 1576 of file frame.c.
References AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_slinear_saturated_divide(), ast_slinear_saturated_multiply(), and f.
Referenced by audiohook_read_frame_single(), and conf_run().
01577 { 01578 int count; 01579 short *fdata = f->data; 01580 short adjust_value = abs(adjustment); 01581 01582 if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR)) 01583 return -1; 01584 01585 if (!adjustment) 01586 return 0; 01587 01588 for (count = 0; count < f->samples; count++) { 01589 if (adjustment > 0) { 01590 ast_slinear_saturated_multiply(&fdata[count], &adjust_value); 01591 } else if (adjustment < 0) { 01592 ast_slinear_saturated_divide(&fdata[count], &adjust_value); 01593 } 01594 } 01595 01596 return 0; 01597 }
void ast_frame_dump | ( | const char * | name, | |
struct ast_frame * | f, | |||
char * | prefix | |||
) |
Dump a frame for debugging purposes
Definition at line 761 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_PROCEEDING, AST_CONTROL_PROGRESS, AST_CONTROL_RADIO_KEY, AST_CONTROL_RADIO_UNKEY, AST_CONTROL_RING, AST_CONTROL_RINGING, 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_verbose(), COLOR_BLACK, COLOR_BRCYAN, COLOR_BRGREEN, COLOR_BRMAGENTA, COLOR_BRRED, COLOR_YELLOW, f, and term_color().
Referenced by __ast_read(), and ast_write().
00762 { 00763 const char noname[] = "unknown"; 00764 char ftype[40] = "Unknown Frametype"; 00765 char cft[80]; 00766 char subclass[40] = "Unknown Subclass"; 00767 char csub[80]; 00768 char moreinfo[40] = ""; 00769 char cn[60]; 00770 char cp[40]; 00771 char cmn[40]; 00772 00773 if (!name) 00774 name = noname; 00775 00776 00777 if (!f) { 00778 ast_verbose("%s [ %s (NULL) ] [%s]\n", 00779 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00780 term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00781 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00782 return; 00783 } 00784 /* XXX We should probably print one each of voice and video when the format changes XXX */ 00785 if (f->frametype == AST_FRAME_VOICE) 00786 return; 00787 if (f->frametype == AST_FRAME_VIDEO) 00788 return; 00789 switch(f->frametype) { 00790 case AST_FRAME_DTMF_BEGIN: 00791 strcpy(ftype, "DTMF Begin"); 00792 subclass[0] = f->subclass; 00793 subclass[1] = '\0'; 00794 break; 00795 case AST_FRAME_DTMF_END: 00796 strcpy(ftype, "DTMF End"); 00797 subclass[0] = f->subclass; 00798 subclass[1] = '\0'; 00799 break; 00800 case AST_FRAME_CONTROL: 00801 strcpy(ftype, "Control"); 00802 switch(f->subclass) { 00803 case AST_CONTROL_HANGUP: 00804 strcpy(subclass, "Hangup"); 00805 break; 00806 case AST_CONTROL_RING: 00807 strcpy(subclass, "Ring"); 00808 break; 00809 case AST_CONTROL_RINGING: 00810 strcpy(subclass, "Ringing"); 00811 break; 00812 case AST_CONTROL_ANSWER: 00813 strcpy(subclass, "Answer"); 00814 break; 00815 case AST_CONTROL_BUSY: 00816 strcpy(subclass, "Busy"); 00817 break; 00818 case AST_CONTROL_TAKEOFFHOOK: 00819 strcpy(subclass, "Take Off Hook"); 00820 break; 00821 case AST_CONTROL_OFFHOOK: 00822 strcpy(subclass, "Line Off Hook"); 00823 break; 00824 case AST_CONTROL_CONGESTION: 00825 strcpy(subclass, "Congestion"); 00826 break; 00827 case AST_CONTROL_FLASH: 00828 strcpy(subclass, "Flash"); 00829 break; 00830 case AST_CONTROL_WINK: 00831 strcpy(subclass, "Wink"); 00832 break; 00833 case AST_CONTROL_OPTION: 00834 strcpy(subclass, "Option"); 00835 break; 00836 case AST_CONTROL_RADIO_KEY: 00837 strcpy(subclass, "Key Radio"); 00838 break; 00839 case AST_CONTROL_RADIO_UNKEY: 00840 strcpy(subclass, "Unkey Radio"); 00841 break; 00842 case AST_CONTROL_PROGRESS: 00843 strcpy(subclass, "Call Progress"); 00844 break; 00845 case AST_CONTROL_PROCEEDING: 00846 strcpy(subclass, "Proceeding"); 00847 break; 00848 case AST_CONTROL_HOLD: 00849 strcpy(subclass, "Hold"); 00850 break; 00851 case AST_CONTROL_UNHOLD: 00852 strcpy(subclass, "UnHold"); 00853 break; 00854 case -1: 00855 strcpy(subclass, "Stop generators"); 00856 break; 00857 default: 00858 snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass); 00859 } 00860 break; 00861 case AST_FRAME_NULL: 00862 strcpy(ftype, "Null Frame"); 00863 strcpy(subclass, "N/A"); 00864 break; 00865 case AST_FRAME_IAX: 00866 /* Should never happen */ 00867 strcpy(ftype, "IAX Specific"); 00868 snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass); 00869 break; 00870 case AST_FRAME_TEXT: 00871 strcpy(ftype, "Text"); 00872 strcpy(subclass, "N/A"); 00873 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00874 break; 00875 case AST_FRAME_IMAGE: 00876 strcpy(ftype, "Image"); 00877 snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass)); 00878 break; 00879 case AST_FRAME_HTML: 00880 strcpy(ftype, "HTML"); 00881 switch(f->subclass) { 00882 case AST_HTML_URL: 00883 strcpy(subclass, "URL"); 00884 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00885 break; 00886 case AST_HTML_DATA: 00887 strcpy(subclass, "Data"); 00888 break; 00889 case AST_HTML_BEGIN: 00890 strcpy(subclass, "Begin"); 00891 break; 00892 case AST_HTML_END: 00893 strcpy(subclass, "End"); 00894 break; 00895 case AST_HTML_LDCOMPLETE: 00896 strcpy(subclass, "Load Complete"); 00897 break; 00898 case AST_HTML_NOSUPPORT: 00899 strcpy(subclass, "No Support"); 00900 break; 00901 case AST_HTML_LINKURL: 00902 strcpy(subclass, "Link URL"); 00903 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00904 break; 00905 case AST_HTML_UNLINK: 00906 strcpy(subclass, "Unlink"); 00907 break; 00908 case AST_HTML_LINKREJECT: 00909 strcpy(subclass, "Link Reject"); 00910 break; 00911 default: 00912 snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass); 00913 break; 00914 } 00915 break; 00916 case AST_FRAME_MODEM: 00917 strcpy(ftype, "Modem"); 00918 switch (f->subclass) { 00919 case AST_MODEM_T38: 00920 strcpy(subclass, "T.38"); 00921 break; 00922 case AST_MODEM_V150: 00923 strcpy(subclass, "V.150"); 00924 break; 00925 default: 00926 snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass); 00927 break; 00928 } 00929 break; 00930 default: 00931 snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype); 00932 } 00933 if (!ast_strlen_zero(moreinfo)) 00934 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n", 00935 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00936 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00937 f->frametype, 00938 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00939 f->subclass, 00940 term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)), 00941 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00942 else 00943 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n", 00944 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00945 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00946 f->frametype, 00947 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00948 f->subclass, 00949 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00950 }
struct ast_frame* ast_frame_enqueue | ( | struct ast_frame * | head, | |
struct ast_frame * | f, | |||
int | maxlen, | |||
int | dupe | |||
) |
Appends a frame to the end of a list of frames, truncating the maximum length of the list.
void ast_frame_free | ( | struct ast_frame * | fr, | |
int | cache | |||
) |
Requests a frame to be allocated Frees a frame.
fr | Frame to free | |
cache | Whether to consider this frame for frame caching |
Definition at line 322 of file frame.c.
References ast_dsp_frame_freed(), ast_filestream_frame_freed(), AST_FRFLAG_FROM_DSP, AST_FRFLAG_FROM_FILESTREAM, 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_threadstorage_get(), ast_translate_frame_freed(), ast_frame::data, frame_cache, FRAME_CACHE_MAX_SIZE, frames, free, ast_frame::mallocd, ast_frame::offset, and ast_frame::src.
Referenced by mixmonitor_thread().
00323 { 00324 if (ast_test_flag(fr, AST_FRFLAG_FROM_TRANSLATOR)) { 00325 ast_translate_frame_freed(fr); 00326 } else if (ast_test_flag(fr, AST_FRFLAG_FROM_DSP)) { 00327 ast_dsp_frame_freed(fr); 00328 } else if (ast_test_flag(fr, AST_FRFLAG_FROM_FILESTREAM)) { 00329 ast_filestream_frame_freed(fr); 00330 } 00331 00332 if (!fr->mallocd) 00333 return; 00334 00335 #if !defined(LOW_MEMORY) 00336 if (cache && fr->mallocd == AST_MALLOCD_HDR) { 00337 /* Cool, only the header is malloc'd, let's just cache those for now 00338 * to keep things simple... */ 00339 struct ast_frame_cache *frames; 00340 00341 if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames))) 00342 && frames->size < FRAME_CACHE_MAX_SIZE) { 00343 AST_LIST_INSERT_HEAD(&frames->list, fr, frame_list); 00344 frames->size++; 00345 return; 00346 } 00347 } 00348 #endif 00349 00350 if (fr->mallocd & AST_MALLOCD_DATA) { 00351 if (fr->data) 00352 free(fr->data - fr->offset); 00353 } 00354 if (fr->mallocd & AST_MALLOCD_SRC) { 00355 if (fr->src) 00356 free((char *)fr->src); 00357 } 00358 if (fr->mallocd & AST_MALLOCD_HDR) { 00359 #ifdef TRACE_FRAMES 00360 AST_LIST_LOCK(&headerlist); 00361 headers--; 00362 AST_LIST_REMOVE(&headerlist, fr, frame_list); 00363 AST_LIST_UNLOCK(&headerlist); 00364 #endif 00365 free(fr); 00366 } 00367 }
Sums two frames of audio samples.
f1 | The first frame (which will contain the result) | |
f2 | The second frame |
Definition at line 1599 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.
01600 { 01601 int count; 01602 short *data1, *data2; 01603 01604 if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR)) 01605 return -1; 01606 01607 if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR)) 01608 return -1; 01609 01610 if (f1->samples != f2->samples) 01611 return -1; 01612 01613 for (count = 0, data1 = f1->data, data2 = f2->data; 01614 count < f1->samples; 01615 count++, data1++, data2++) 01616 ast_slinear_saturated_add(data1, data2); 01617 01618 return 0; 01619 }
Copies a frame.
fr | frame to copy Duplicates a frame -- should only rarely be used, typically frisolate is good enough |
Definition at line 433 of file frame.c.
References ast_calloc_cache, ast_copy_flags, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, AST_LIST_REMOVE_CURRENT, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, AST_MALLOCD_HDR, ast_threadstorage_get(), ast_frame::data, ast_frame::datalen, ast_frame::delivery, f, frame_cache, frames, ast_frame::frametype, ast_frame::len, len(), ast_frame::mallocd, ast_frame::mallocd_hdr_len, ast_frame::offset, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, and ast_frame::ts.
Referenced by __ast_queue_frame(), ast_jb_put(), ast_rtp_write(), ast_slinfactory_feed(), audiohook_read_frame_single(), autoservice_run(), recordthread(), and rpt().
00434 { 00435 struct ast_frame *out = NULL; 00436 int len, srclen = 0; 00437 void *buf = NULL; 00438 00439 #if !defined(LOW_MEMORY) 00440 struct ast_frame_cache *frames; 00441 #endif 00442 00443 /* Start with standard stuff */ 00444 len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00445 /* If we have a source, add space for it */ 00446 /* 00447 * XXX Watch out here - if we receive a src which is not terminated 00448 * properly, we can be easily attacked. Should limit the size we deal with. 00449 */ 00450 if (f->src) 00451 srclen = strlen(f->src); 00452 if (srclen > 0) 00453 len += srclen + 1; 00454 00455 #if !defined(LOW_MEMORY) 00456 if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) { 00457 AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) { 00458 if (out->mallocd_hdr_len >= len) { 00459 size_t mallocd_len = out->mallocd_hdr_len; 00460 AST_LIST_REMOVE_CURRENT(&frames->list, frame_list); 00461 memset(out, 0, sizeof(*out)); 00462 out->mallocd_hdr_len = mallocd_len; 00463 buf = out; 00464 frames->size--; 00465 break; 00466 } 00467 } 00468 AST_LIST_TRAVERSE_SAFE_END 00469 } 00470 #endif 00471 00472 if (!buf) { 00473 if (!(buf = ast_calloc_cache(1, len))) 00474 return NULL; 00475 out = buf; 00476 out->mallocd_hdr_len = len; 00477 } 00478 00479 out->frametype = f->frametype; 00480 out->subclass = f->subclass; 00481 out->datalen = f->datalen; 00482 out->samples = f->samples; 00483 out->delivery = f->delivery; 00484 /* Set us as having malloc'd header only, so it will eventually 00485 get freed. */ 00486 out->mallocd = AST_MALLOCD_HDR; 00487 out->offset = AST_FRIENDLY_OFFSET; 00488 if (out->datalen) { 00489 out->data = buf + sizeof(*out) + AST_FRIENDLY_OFFSET; 00490 memcpy(out->data, f->data, out->datalen); 00491 } 00492 if (srclen > 0) { 00493 /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */ 00494 char *src; 00495 out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00496 src = (char *) out->src; 00497 /* Must have space since we allocated for it */ 00498 strcpy(src, f->src); 00499 } 00500 ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO); 00501 out->ts = f->ts; 00502 out->len = f->len; 00503 out->seqno = f->seqno; 00504 return out; 00505 }
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 374 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().
00375 { 00376 struct ast_frame *out; 00377 void *newdata; 00378 00379 ast_clear_flag(fr, AST_FRFLAG_FROM_TRANSLATOR); 00380 ast_clear_flag(fr, AST_FRFLAG_FROM_DSP); 00381 00382 if (!(fr->mallocd & AST_MALLOCD_HDR)) { 00383 /* Allocate a new header if needed */ 00384 if (!(out = ast_frame_header_new())) 00385 return NULL; 00386 out->frametype = fr->frametype; 00387 out->subclass = fr->subclass; 00388 out->datalen = fr->datalen; 00389 out->samples = fr->samples; 00390 out->offset = fr->offset; 00391 out->data = fr->data; 00392 /* Copy the timing data */ 00393 ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO); 00394 if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) { 00395 out->ts = fr->ts; 00396 out->len = fr->len; 00397 out->seqno = fr->seqno; 00398 } 00399 } else 00400 out = fr; 00401 00402 if (!(fr->mallocd & AST_MALLOCD_SRC)) { 00403 if (fr->src) { 00404 if (!(out->src = ast_strdup(fr->src))) { 00405 if (out != fr) 00406 free(out); 00407 return NULL; 00408 } 00409 } 00410 } else 00411 out->src = fr->src; 00412 00413 if (!(fr->mallocd & AST_MALLOCD_DATA)) { 00414 if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) { 00415 if (out->src != fr->src) 00416 free((void *) out->src); 00417 if (out != fr) 00418 free(out); 00419 return NULL; 00420 } 00421 newdata += AST_FRIENDLY_OFFSET; 00422 out->offset = AST_FRIENDLY_OFFSET; 00423 out->datalen = fr->datalen; 00424 memcpy(newdata, fr->data, fr->datalen); 00425 out->data = newdata; 00426 } 00427 00428 out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA; 00429 00430 return out; 00431 }
struct ast_format_list* ast_get_format_list | ( | size_t * | size | ) |
Definition at line 523 of file frame.c.
References AST_FORMAT_LIST.
00524 { 00525 *size = (sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0])); 00526 return AST_FORMAT_LIST; 00527 }
struct ast_format_list* ast_get_format_list_index | ( | int | index | ) |
Definition at line 518 of file frame.c.
References AST_FORMAT_LIST.
00519 { 00520 return &AST_FORMAT_LIST[index]; 00521 }
int ast_getformatbyname | ( | const char * | name | ) |
Gets a format from a name.
name | string of format |
Definition at line 589 of file frame.c.
References ast_expand_codec_alias(), AST_FORMAT_LIST, and format.
Referenced by ast_parse_allow_disallow(), iax_template_parse(), reload_config(), and try_suggested_sip_codec().
00590 { 00591 int x, all, format = 0; 00592 00593 all = strcasecmp(name, "all") ? 0 : 1; 00594 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00595 if(AST_FORMAT_LIST[x].visible && (all || 00596 !strcasecmp(AST_FORMAT_LIST[x].name,name) || 00597 !strcasecmp(AST_FORMAT_LIST[x].name,ast_expand_codec_alias(name)))) { 00598 format |= AST_FORMAT_LIST[x].bits; 00599 if(!all) 00600 break; 00601 } 00602 } 00603 00604 return format; 00605 }
char* ast_getformatname | ( | int | format | ) |
Get the name of a format.
format | id of format |
Definition at line 529 of file frame.c.
References AST_FORMAT_LIST, ast_format_list::bits, 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(), dahdi_read(), 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(), and socket_process().
00530 { 00531 int x; 00532 char *ret = "unknown"; 00533 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00534 if(AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == format) { 00535 ret = AST_FORMAT_LIST[x].name; 00536 break; 00537 } 00538 } 00539 return ret; 00540 }
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 542 of file frame.c.
References AST_FORMAT_LIST, ast_format_list::bits, len(), name, and ast_format_list::visible.
Referenced by __ast_read(), __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_show_peer(), process_sdp(), serialize_showchan(), set_format(), sip_new(), sip_request_call(), sip_show_channel(), sip_show_settings(), and sip_write().
00543 { 00544 int x; 00545 unsigned len; 00546 char *start, *end = buf; 00547 00548 if (!size) 00549 return buf; 00550 snprintf(end, size, "0x%x (", format); 00551 len = strlen(end); 00552 end += len; 00553 size -= len; 00554 start = end; 00555 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00556 if (AST_FORMAT_LIST[x].visible && (AST_FORMAT_LIST[x].bits & format)) { 00557 snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name); 00558 len = strlen(end); 00559 end += len; 00560 size -= len; 00561 } 00562 } 00563 if (start == end) 00564 snprintf(start, size, "nothing)"); 00565 else if (size > 1) 00566 *(end -1) = ')'; 00567 return buf; 00568 }
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 1305 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, and parse().
Referenced by action_originate(), apply_outgoing(), build_device(), build_peer(), build_user(), gtalk_create_member(), gtalk_load_config(), reload_config(), set_config(), and update_common_options().
01306 { 01307 char *parse = NULL, *this = NULL, *psize = NULL; 01308 int format = 0, framems = 0; 01309 01310 parse = ast_strdupa(list); 01311 while ((this = strsep(&parse, ","))) { 01312 framems = 0; 01313 if ((psize = strrchr(this, ':'))) { 01314 *psize++ = '\0'; 01315 if (option_debug) 01316 ast_log(LOG_DEBUG,"Packetization for codec: %s is %s\n", this, psize); 01317 framems = atoi(psize); 01318 if (framems < 0) 01319 framems = 0; 01320 } 01321 if (!(format = ast_getformatbyname(this))) { 01322 ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this); 01323 continue; 01324 } 01325 01326 if (mask) { 01327 if (allowing) 01328 *mask |= format; 01329 else 01330 *mask &= ~format; 01331 } 01332 01333 /* Set up a preference list for audio. Do not include video in preferences 01334 since we can not transcode video and have to use whatever is offered 01335 */ 01336 if (pref && (format & AST_FORMAT_AUDIO_MASK)) { 01337 if (strcasecmp(this, "all")) { 01338 if (allowing) { 01339 ast_codec_pref_append(pref, format); 01340 ast_codec_pref_setsize(pref, format, framems); 01341 } 01342 else 01343 ast_codec_pref_remove(pref, format); 01344 } else if (!allowing) { 01345 memset(pref, 0, sizeof(*pref)); 01346 } 01347 } 01348 } 01349 }
void ast_smoother_free | ( | struct ast_smoother * | s | ) |
Definition at line 269 of file frame.c.
Referenced by ast_rtp_codec_setpref(), ast_rtp_destroy(), and ast_rtp_write().
int ast_smoother_get_flags | ( | struct ast_smoother * | smoother | ) |
struct ast_smoother* ast_smoother_new | ( | int | bytes | ) |
Definition at line 148 of file frame.c.
References ast_malloc, ast_smoother_reset(), and s.
Referenced by ast_rtp_write().
00149 { 00150 struct ast_smoother *s; 00151 if (size < 1) 00152 return NULL; 00153 if ((s = ast_malloc(sizeof(*s)))) 00154 ast_smoother_reset(s, size); 00155 return s; 00156 }
struct ast_frame* ast_smoother_read | ( | struct ast_smoother * | s | ) |
Definition at line 219 of file frame.c.
References AST_FRAME_VOICE, AST_FRIENDLY_OFFSET, ast_log(), ast_samp2tv(), AST_SMOOTHER_FLAG_G729, ast_tvadd(), ast_tvzero(), len(), LOG_WARNING, and s.
Referenced by ast_rtp_write().
00220 { 00221 struct ast_frame *opt; 00222 int len; 00223 00224 /* IF we have an optimization frame, send it */ 00225 if (s->opt) { 00226 if (s->opt->offset < AST_FRIENDLY_OFFSET) 00227 ast_log(LOG_WARNING, "Returning a frame of inappropriate offset (%d).\n", 00228 s->opt->offset); 00229 opt = s->opt; 00230 s->opt = NULL; 00231 return opt; 00232 } 00233 00234 /* Make sure we have enough data */ 00235 if (s->len < s->size) { 00236 /* Or, if this is a G.729 frame with VAD on it, send it immediately anyway */ 00237 if (!((s->flags & AST_SMOOTHER_FLAG_G729) && (s->size % 10))) 00238 return NULL; 00239 } 00240 len = s->size; 00241 if (len > s->len) 00242 len = s->len; 00243 /* Make frame */ 00244 s->f.frametype = AST_FRAME_VOICE; 00245 s->f.subclass = s->format; 00246 s->f.data = s->framedata + AST_FRIENDLY_OFFSET; 00247 s->f.offset = AST_FRIENDLY_OFFSET; 00248 s->f.datalen = len; 00249 /* Samples will be improper given VAD, but with VAD the concept really doesn't even exist */ 00250 s->f.samples = len * s->samplesperbyte; /* XXX rounding */ 00251 s->f.delivery = s->delivery; 00252 /* Fill Data */ 00253 memcpy(s->f.data, s->data, len); 00254 s->len -= len; 00255 /* Move remaining data to the front if applicable */ 00256 if (s->len) { 00257 /* In principle this should all be fine because if we are sending 00258 G.729 VAD, the next timestamp will take over anyawy */ 00259 memmove(s->data, s->data + len, s->len); 00260 if (!ast_tvzero(s->delivery)) { 00261 /* If we have delivery time, increment it, otherwise, leave it at 0 */ 00262 s->delivery = ast_tvadd(s->delivery, ast_samp2tv(s->f.samples, 8000)); 00263 } 00264 } 00265 /* Return frame */ 00266 return &s->f; 00267 }
void ast_smoother_reset | ( | struct ast_smoother * | s, | |
int | bytes | |||
) |
Definition at line 142 of file frame.c.
References s.
Referenced by ast_smoother_new().
00143 { 00144 memset(s, 0, sizeof(*s)); 00145 s->size = size; 00146 }
void ast_smoother_set_flags | ( | struct ast_smoother * | smoother, | |
int | flags | |||
) |
int ast_smoother_test_flag | ( | struct ast_smoother * | s, | |
int | flag | |||
) |
Definition at line 168 of file frame.c.
References s.
Referenced by ast_rtp_write().
00169 { 00170 return (s->flags & flag); 00171 }
void ast_swapcopy_samples | ( | void * | dst, | |
const void * | src, | |||
int | samples | |||
) |
Definition at line 507 of file frame.c.
Referenced by __ast_smoother_feed(), iax_frame_wrap(), and phone_write_buf().
00508 { 00509 int i; 00510 unsigned short *dst_s = dst; 00511 const unsigned short *src_s = src; 00512 00513 for (i = 0; i < samples; i++) 00514 dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8); 00515 }
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 140 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(), 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().