#include <sys/types.h>
#include <sys/time.h>
#include "asterisk/compiler.h"
#include "asterisk/endian.h"
#include "asterisk/linkedlists.h"
Include dependency graph for frame.h:
This graph shows which files directly or indirectly include this file:
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_MP4_VIDEO (1 << 22) |
#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_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_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 } |
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_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. | |
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. | |
static void force_inline | ast_frfree (struct ast_frame *fr) |
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 230 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 226 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_generate(), zt_new(), zt_read(), and zt_write().
#define AST_FORMAT_AUDIO_MASK ((1 << 16)-1) |
Maximum audio mask
Definition at line 248 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(), begin_dial(), func_channel_read(), 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 244 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_samples(), convertcap(), and g722tolin_sample().
#define AST_FORMAT_G723_1 (1 << 0) |
G.723.1 compression
Definition at line 220 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_samples(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), g723_read(), g723_write(), load_module(), phone_request(), phone_setup(), and phone_write().
#define AST_FORMAT_G726 (1 << 11) |
ADPCM (G.726, 32kbps, RFC3551 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_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 228 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 236 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(), and g729_write().
#define AST_FORMAT_GSM (1 << 1) |
GSM compression
Definition at line 222 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 254 of file frame.h.
Referenced by codec_ast2skinny(), and codec_skinny2ast().
#define AST_FORMAT_H263 (1 << 19) |
H.263 Video
Definition at line 256 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 240 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 250 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 234 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 246 of file frame.h.
Referenced by add_sdp(), ast_closestream(), ast_filehelper(), ast_openvstream(), ast_playstream(), ast_rtp_raw_write(), 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 264 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 232 of file frame.h.
Referenced by __ast_play_and_record(), __ast_register_translator(), action_originate(), agent_new(), alsa_new(), alsa_read(), alsa_request(), ast_best_codec(), ast_channel_make_compatible(), ast_channel_spy_add(), ast_channel_start_silence_generator(), ast_channel_whisper_stop(), 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(), ast_write(), attempt_reconnect(), attempt_thread(), background_detect_exec(), build_conf(), channel_spy(), chanspy_exec(), conf_run(), connect_link(), dictate_exec(), do_waiting(), eagi_exec(), extenspy_exec(), fast_originate(), handle_recordfile(), iax_frame_wrap(), ices_exec(), isAnsweringMachine(), launch_monitor_thread(), linear_alloc(), linear_generator(), lintoadpcm_sample(), lintoalaw_sample(), lintog722_sample(), lintog726_sample(), lintogsm_sample(), lintoilbc_sample(), lintolpc10_sample(), lintospeex_sample(), lintoulaw_sample(), load_module(), load_moh_classes(), measurenoise(), misdn_set_opt_exec(), moh_class_malloc(), mp3_exec(), mp3_open(), mp3_read(), mwanalyze_exec(), nbs_request(), nbs_xwrite(), NBScat_exec(), nv_background_detect_exec(), nv_detectfax_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(), queue_frame_to_spies(), read_config(), rpt(), rpt_call(), rpt_tele_thread(), rxfax_exec(), send_waveform_to_channel(), silence_generator_generate(), slinear_read(), slinear_write(), sms_generate(), socket_process(), speech_background(), speech_create(), tonepair_alloc(), txfax_exec(), wav_read(), wav_write(), zt_new(), zt_read(), and zt_write().
#define AST_FORMAT_SPEEX (1 << 9) |
SpeeX Free Compression
Definition at line 238 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 224 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_exec(), milliwatt_generate(), oh323_rtp_read(), 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 265 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 124 of file frame.h.
Referenced by __adsi_transmit_messages(), __ast_play_and_record(), action_zapdialoffhook(), agent_ack_sleep(), app_exec(), ast_bridge_call(), ast_dsp_process(), ast_feature_request_and_dial(), ast_jb_put(), background_detect_exec(), bridge_native_loop(), bridge_p2p_loop(), 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(), mgcp_rtp_read(), misdn_bridge(), mp3_exec(), NBScat_exec(), nv_background_detect_exec(), nv_detectfax_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(), 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 169 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_FRIENDLY_OFFSET 64 |
Definition at line 180 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(), mp3_read(), mwanalyze_exec(), NBScat_exec(), newpvt(), ogg_vorbis_read(), oss_read(), pcm_read(), phone_read(), process_rfc3389(), rxfax_exec(), send_tone_burst(), send_waveform_to_channel(), slinear_read(), sms_generate(), txfax_exec(), 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 208 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 210 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 200 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 188 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 186 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 190 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 194 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 310 of file frame.h.
Referenced by zt_hangup(), and zt_setoption().
#define AST_OPTION_ECHOCAN 8 |
Explicitly enable or disable echo cancelation for the given channel
Definition at line 332 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 307 of file frame.h.
Referenced by 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 326 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 304 of file frame.h.
Referenced by handle_tddmode(), 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 301 of file frame.h.
Referenced by conf_run(), rpt(), 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 318 of file frame.h.
Referenced by common_exec(), func_channel_write(), iax2_setoption(), reset_volumes(), set_listen_volume(), set_volume(), and zt_setoption().
#define AST_SMOOTHER_FLAG_BE (1 << 1) |
#define AST_SMOOTHER_FLAG_G729 (1 << 0) |
Definition at line 288 of file frame.h.
Referenced by __ast_smoother_feed(), and ast_smoother_read().
Definition at line 267 of file frame.h.
00267 { 00268 AST_CONTROL_HANGUP = 1, /*!< Other end has hungup */ 00269 AST_CONTROL_RING = 2, /*!< Local ring */ 00270 AST_CONTROL_RINGING = 3, /*!< Remote end is ringing */ 00271 AST_CONTROL_ANSWER = 4, /*!< Remote end has answered */ 00272 AST_CONTROL_BUSY = 5, /*!< Remote end is busy */ 00273 AST_CONTROL_TAKEOFFHOOK = 6, /*!< Make it go off hook */ 00274 AST_CONTROL_OFFHOOK = 7, /*!< Line is off hook */ 00275 AST_CONTROL_CONGESTION = 8, /*!< Congestion (circuits busy) */ 00276 AST_CONTROL_FLASH = 9, /*!< Flash hook */ 00277 AST_CONTROL_WINK = 10, /*!< Wink */ 00278 AST_CONTROL_OPTION = 11, /*!< Set a low-level option */ 00279 AST_CONTROL_RADIO_KEY = 12, /*!< Key Radio */ 00280 AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */ 00281 AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */ 00282 AST_CONTROL_PROCEEDING = 15, /*!< Indicate CALL PROCEEDING */ 00283 AST_CONTROL_HOLD = 16, /*!< Indicate call is placed on hold */ 00284 AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold */ 00285 AST_CONTROL_VIDUPDATE = 18, /*!< Indicate video frame update */ 00286 };
enum ast_frame_type |
Frame types.
Definition at line 97 of file frame.h.
00097 { 00098 /*! DTMF end event, subclass is the digit */ 00099 AST_FRAME_DTMF_END = 1, 00100 /*! Voice data, subclass is AST_FORMAT_* */ 00101 AST_FRAME_VOICE, 00102 /*! Video frame, maybe?? :) */ 00103 AST_FRAME_VIDEO, 00104 /*! A control frame, subclass is AST_CONTROL_* */ 00105 AST_FRAME_CONTROL, 00106 /*! An empty, useless frame */ 00107 AST_FRAME_NULL, 00108 /*! Inter Asterisk Exchange private frame type */ 00109 AST_FRAME_IAX, 00110 /*! Text messages */ 00111 AST_FRAME_TEXT, 00112 /*! Image Frames */ 00113 AST_FRAME_IMAGE, 00114 /*! HTML Frame */ 00115 AST_FRAME_HTML, 00116 /*! Comfort Noise frame (subclass is level of CNG in -dBov), 00117 body may include zero or more 8-bit quantization coefficients */ 00118 AST_FRAME_CNG, 00119 /*! Modem-over-IP data streams */ 00120 AST_FRAME_MODEM, 00121 /*! DTMF begin event, subclass is the digit */ 00122 AST_FRAME_DTMF_BEGIN, 00123 };
int __ast_smoother_feed | ( | struct ast_smoother * | s, | |
struct ast_frame * | f, | |||
int | swap | |||
) |
Definition at line 171 of file frame.c.
References AST_FRAME_VOICE, ast_log(), AST_MIN_OFFSET, AST_SMOOTHER_FLAG_G729, ast_swapcopy_samples(), f, LOG_NOTICE, LOG_WARNING, s, and SMOOTHER_SIZE.
00172 { 00173 if (f->frametype != AST_FRAME_VOICE) { 00174 ast_log(LOG_WARNING, "Huh? Can't smooth a non-voice frame!\n"); 00175 return -1; 00176 } 00177 if (!s->format) { 00178 s->format = f->subclass; 00179 s->samplesperbyte = (float)f->samples / (float)f->datalen; 00180 } else if (s->format != f->subclass) { 00181 ast_log(LOG_WARNING, "Smoother was working on %d format frames, now trying to feed %d?\n", s->format, f->subclass); 00182 return -1; 00183 } 00184 if (s->len + f->datalen > SMOOTHER_SIZE) { 00185 ast_log(LOG_WARNING, "Out of smoother space\n"); 00186 return -1; 00187 } 00188 if (((f->datalen == s->size) || ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) 00189 && !s->opt && (f->offset >= AST_MIN_OFFSET)) { 00190 if (!s->len) { 00191 /* Optimize by sending the frame we just got 00192 on the next read, thus eliminating the douple 00193 copy */ 00194 if (swap) 00195 ast_swapcopy_samples(f->data, f->data, f->samples); 00196 s->opt = f; 00197 return 0; 00198 } else { 00199 s->optimizablestream++; 00200 if (s->optimizablestream > 10) { 00201 /* For the past 10 rounds, we have input and output 00202 frames of the correct size for this smoother, yet 00203 we were unable to optimize because there was still 00204 some cruft left over. Lets just drop the cruft so 00205 we can move to a fully optimized path */ 00206 if (swap) 00207 ast_swapcopy_samples(f->data, f->data, f->samples); 00208 s->len = 0; 00209 s->opt = f; 00210 return 0; 00211 } 00212 } 00213 } else 00214 s->optimizablestream = 0; 00215 if (s->flags & AST_SMOOTHER_FLAG_G729) { 00216 if (s->len % 10) { 00217 ast_log(LOG_NOTICE, "Dropping extra frame of G.729 since we already have a VAD frame at the end\n"); 00218 return 0; 00219 } 00220 } 00221 if (swap) 00222 ast_swapcopy_samples(s->data+s->len, f->data, f->samples); 00223 else 00224 memcpy(s->data + s->len, f->data, f->datalen); 00225 /* If either side is empty, reset the delivery time */ 00226 if (!s->len || ast_tvzero(f->delivery) || ast_tvzero(s->delivery)) /* XXX really ? */ 00227 s->delivery = f->delivery; 00228 s->len += f->datalen; 00229 return 0; 00230 }
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 609 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().
00610 { 00611 int x; 00612 char *ret = "unknown"; 00613 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00614 if(AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == codec) { 00615 ret = AST_FORMAT_LIST[x].desc; 00616 break; 00617 } 00618 } 00619 return ret; 00620 }
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 1245 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().
01246 { 01247 int x, ret = 0, slot; 01248 01249 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01250 slot = pref->order[x]; 01251 01252 if (!slot) 01253 break; 01254 if (formats & AST_FORMAT_LIST[slot-1].bits) { 01255 ret = AST_FORMAT_LIST[slot-1].bits; 01256 break; 01257 } 01258 } 01259 if(ret & AST_FORMAT_AUDIO_MASK) 01260 return ret; 01261 01262 if (option_debug > 3) 01263 ast_log(LOG_DEBUG, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec"); 01264 01265 return find_best ? ast_best_codec(formats) : 0; 01266 }
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 1504 of file frame.c.
References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, 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 ast_channel_spy_read_frame(), copy_data_from_queue(), moh_generate(), and monmp3thread().
01505 { 01506 int len = 0; 01507 01508 /* XXX Still need speex, g723, and lpc10 XXX */ 01509 switch(format) { 01510 case AST_FORMAT_ILBC: 01511 len = (samples / 240) * 50; 01512 break; 01513 case AST_FORMAT_GSM: 01514 len = (samples / 160) * 33; 01515 break; 01516 case AST_FORMAT_G729A: 01517 len = samples / 8; 01518 break; 01519 case AST_FORMAT_SLINEAR: 01520 len = samples * 2; 01521 break; 01522 case AST_FORMAT_ULAW: 01523 case AST_FORMAT_ALAW: 01524 len = samples; 01525 break; 01526 case AST_FORMAT_ADPCM: 01527 case AST_FORMAT_G726: 01528 case AST_FORMAT_G726_AAL2: 01529 len = samples / 2; 01530 break; 01531 default: 01532 ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format)); 01533 } 01534 01535 return len; 01536 }
int ast_codec_get_samples | ( | struct ast_frame * | f | ) |
Returns the number of samples contained in the frame.
Definition at line 1461 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().
01462 { 01463 int samples=0; 01464 switch(f->subclass) { 01465 case AST_FORMAT_SPEEX: 01466 samples = speex_samples(f->data, f->datalen); 01467 break; 01468 case AST_FORMAT_G723_1: 01469 samples = g723_samples(f->data, f->datalen); 01470 break; 01471 case AST_FORMAT_ILBC: 01472 samples = 240 * (f->datalen / 50); 01473 break; 01474 case AST_FORMAT_GSM: 01475 samples = 160 * (f->datalen / 33); 01476 break; 01477 case AST_FORMAT_G729A: 01478 samples = f->datalen * 8; 01479 break; 01480 case AST_FORMAT_SLINEAR: 01481 samples = f->datalen / 2; 01482 break; 01483 case AST_FORMAT_LPC10: 01484 /* assumes that the RTP packet contains one LPC10 frame */ 01485 samples = 22 * 8; 01486 samples += (((char *)(f->data))[7] & 0x1) * 8; 01487 break; 01488 case AST_FORMAT_ULAW: 01489 case AST_FORMAT_ALAW: 01490 case AST_FORMAT_G722: 01491 samples = f->datalen; 01492 break; 01493 case AST_FORMAT_ADPCM: 01494 case AST_FORMAT_G726: 01495 case AST_FORMAT_G726_AAL2: 01496 samples = f->datalen * 2; 01497 break; 01498 default: 01499 ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass)); 01500 } 01501 return samples; 01502 }
static int ast_codec_interp_len | ( | int | format | ) | [inline, static] |
Gets duration in ms of interpolation frame for a format.
Definition at line 553 of file frame.h.
References AST_FORMAT_ILBC.
Referenced by __get_from_jb(), and jb_get_and_deliver().
00554 { 00555 return (format == AST_FORMAT_ILBC) ? 30 : 20; 00556 }
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 1140 of file frame.c.
References ast_codec_pref_remove(), AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow().
01141 { 01142 int x, newindex = -1; 01143 01144 ast_codec_pref_remove(pref, format); 01145 01146 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01147 if(AST_FORMAT_LIST[x].bits == format) { 01148 newindex = x + 1; 01149 break; 01150 } 01151 } 01152 01153 if(newindex) { 01154 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01155 if(!pref->order[x]) { 01156 pref->order[x] = newindex; 01157 break; 01158 } 01159 } 01160 } 01161 01162 return x; 01163 }
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 1042 of file frame.c.
References ast_codec_pref::order.
Referenced by check_access(), create_addr(), dump_prefs(), and socket_process().
01043 { 01044 int x, differential = (int) 'A', mem; 01045 char *from, *to; 01046 01047 if(right) { 01048 from = pref->order; 01049 to = buf; 01050 mem = size; 01051 } else { 01052 to = pref->order; 01053 from = buf; 01054 mem = 32; 01055 } 01056 01057 memset(to, 0, mem); 01058 for (x = 0; x < 32 ; x++) { 01059 if(!from[x]) 01060 break; 01061 to[x] = right ? (from[x] + differential) : (from[x] - differential); 01062 } 01063 }
struct ast_format_list ast_codec_pref_getsize | ( | struct ast_codec_pref * | pref, | |
int | format | |||
) |
Get packet size for codec.
Definition at line 1206 of file frame.c.
References AST_FORMAT_LIST, ast_format_list::bits, fmt, and format.
Referenced by add_codec_to_sdp(), ast_rtp_bridge(), ast_rtp_write(), handle_open_receive_channel_ack_message(), and transmit_connect().
01207 { 01208 int x, index = -1, framems = 0; 01209 struct ast_format_list fmt = {0}; 01210 01211 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01212 if(AST_FORMAT_LIST[x].bits == format) { 01213 fmt = AST_FORMAT_LIST[x]; 01214 index = x; 01215 break; 01216 } 01217 } 01218 01219 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01220 if(pref->order[x] == (index + 1)) { 01221 framems = pref->framing[x]; 01222 break; 01223 } 01224 } 01225 01226 /* size validation */ 01227 if(!framems) 01228 framems = AST_FORMAT_LIST[index].def_ms; 01229 01230 if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */ 01231 framems -= framems % AST_FORMAT_LIST[index].inc_ms; 01232 01233 if(framems < AST_FORMAT_LIST[index].min_ms) 01234 framems = AST_FORMAT_LIST[index].min_ms; 01235 01236 if(framems > AST_FORMAT_LIST[index].max_ms) 01237 framems = AST_FORMAT_LIST[index].max_ms; 01238 01239 fmt.cur_ms = framems; 01240 01241 return fmt; 01242 }
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 1100 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().
01101 { 01102 int slot = 0; 01103 01104 01105 if((index >= 0) && (index < sizeof(pref->order))) { 01106 slot = pref->order[index]; 01107 } 01108 01109 return slot ? AST_FORMAT_LIST[slot-1].bits : 0; 01110 }
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_remove | ( | struct ast_codec_pref * | pref, | |
int | format | |||
) |
Remove audio a codec from a preference list.
Definition at line 1113 of file frame.c.
References AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_codec_pref_append(), and ast_parse_allow_disallow().
01114 { 01115 struct ast_codec_pref oldorder; 01116 int x, y = 0; 01117 int slot; 01118 int size; 01119 01120 if(!pref->order[0]) 01121 return; 01122 01123 memcpy(&oldorder, pref, sizeof(oldorder)); 01124 memset(pref, 0, sizeof(*pref)); 01125 01126 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01127 slot = oldorder.order[x]; 01128 size = oldorder.framing[x]; 01129 if(! slot) 01130 break; 01131 if(AST_FORMAT_LIST[slot-1].bits != format) { 01132 pref->order[y] = slot; 01133 pref->framing[y++] = size; 01134 } 01135 } 01136 01137 }
int ast_codec_pref_setsize | ( | struct ast_codec_pref * | pref, | |
int | format, | |||
int | framems | |||
) |
Set packet size for codec.
Definition at line 1167 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().
01168 { 01169 int x, index = -1; 01170 01171 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01172 if(AST_FORMAT_LIST[x].bits == format) { 01173 index = x; 01174 break; 01175 } 01176 } 01177 01178 if(index < 0) 01179 return -1; 01180 01181 /* size validation */ 01182 if(!framems) 01183 framems = AST_FORMAT_LIST[index].def_ms; 01184 01185 if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */ 01186 framems -= framems % AST_FORMAT_LIST[index].inc_ms; 01187 01188 if(framems < AST_FORMAT_LIST[index].min_ms) 01189 framems = AST_FORMAT_LIST[index].min_ms; 01190 01191 if(framems > AST_FORMAT_LIST[index].max_ms) 01192 framems = AST_FORMAT_LIST[index].max_ms; 01193 01194 01195 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01196 if(pref->order[x] == (index + 1)) { 01197 pref->framing[x] = framems; 01198 break; 01199 } 01200 } 01201 01202 return x; 01203 }
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 1065 of file frame.c.
References ast_codec_pref_index(), and ast_getformatname().
Referenced by dump_prefs(), and socket_process().
01066 { 01067 int x, codec; 01068 size_t total_len, slen; 01069 char *formatname; 01070 01071 memset(buf,0,size); 01072 total_len = size; 01073 buf[0] = '('; 01074 total_len--; 01075 for(x = 0; x < 32 ; x++) { 01076 if(total_len <= 0) 01077 break; 01078 if(!(codec = ast_codec_pref_index(pref,x))) 01079 break; 01080 if((formatname = ast_getformatname(codec))) { 01081 slen = strlen(formatname); 01082 if(slen > total_len) 01083 break; 01084 strncat(buf,formatname,total_len); 01085 total_len -= slen; 01086 } 01087 if(total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) { 01088 strncat(buf,"|",total_len); 01089 total_len--; 01090 } 01091 } 01092 if(total_len) { 01093 strncat(buf,")",total_len); 01094 total_len--; 01095 } 01096 01097 return size - total_len; 01098 }
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 1538 of file frame.c.
References AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_slinear_saturated_divide(), ast_slinear_saturated_multiply(), and f.
Referenced by ast_channel_spy_read_frame(), ast_write(), and conf_run().
01539 { 01540 int count; 01541 short *fdata = f->data; 01542 short adjust_value = abs(adjustment); 01543 01544 if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR)) 01545 return -1; 01546 01547 if (!adjustment) 01548 return 0; 01549 01550 for (count = 0; count < f->samples; count++) { 01551 if (adjustment > 0) { 01552 ast_slinear_saturated_multiply(&fdata[count], &adjust_value); 01553 } else if (adjustment < 0) { 01554 ast_slinear_saturated_divide(&fdata[count], &adjust_value); 01555 } 01556 } 01557 01558 return 0; 01559 }
void ast_frame_dump | ( | const char * | name, | |
struct ast_frame * | f, | |||
char * | prefix | |||
) |
Dump a frame for debugging purposes
Definition at line 763 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_log(), AST_MODEM_T38, AST_MODEM_V150, ast_strlen_zero(), f, and LOG_DEBUG.
Referenced by __ast_read(), ast_write(), and sip_write().
00764 { 00765 const char noname[] = "unknown"; 00766 char ftype[40] = "Unknown Frametype"; 00767 char subclass[40] = "Unknown Subclass"; 00768 char moreinfo[40] = ""; 00769 00770 if (!name) 00771 name = noname; 00772 00773 if (!f) { 00774 ast_log( LOG_DEBUG, "%s [ %s (NULL) ] [%s]\n", 00775 prefix, "HANGUP", name); 00776 return; 00777 } 00778 /* XXX We should probably print one each of voice and video when the format changes XXX */ 00779 if (f->frametype == AST_FRAME_VOICE) 00780 return; 00781 if (f->frametype == AST_FRAME_VIDEO) 00782 return; 00783 switch(f->frametype) { 00784 case AST_FRAME_DTMF_BEGIN: 00785 strcpy(ftype, "DTMF Begin"); 00786 subclass[0] = f->subclass; 00787 subclass[1] = '\0'; 00788 break; 00789 case AST_FRAME_DTMF_END: 00790 strcpy(ftype, "DTMF End"); 00791 subclass[0] = f->subclass; 00792 subclass[1] = '\0'; 00793 break; 00794 case AST_FRAME_CONTROL: 00795 strcpy(ftype, "Control"); 00796 switch(f->subclass) { 00797 case AST_CONTROL_HANGUP: 00798 strcpy(subclass, "Hangup"); 00799 break; 00800 case AST_CONTROL_RING: 00801 strcpy(subclass, "Ring"); 00802 break; 00803 case AST_CONTROL_RINGING: 00804 strcpy(subclass, "Ringing"); 00805 break; 00806 case AST_CONTROL_ANSWER: 00807 strcpy(subclass, "Answer"); 00808 break; 00809 case AST_CONTROL_BUSY: 00810 strcpy(subclass, "Busy"); 00811 break; 00812 case AST_CONTROL_TAKEOFFHOOK: 00813 strcpy(subclass, "Take Off Hook"); 00814 break; 00815 case AST_CONTROL_OFFHOOK: 00816 strcpy(subclass, "Line Off Hook"); 00817 break; 00818 case AST_CONTROL_CONGESTION: 00819 strcpy(subclass, "Congestion"); 00820 break; 00821 case AST_CONTROL_FLASH: 00822 strcpy(subclass, "Flash"); 00823 break; 00824 case AST_CONTROL_WINK: 00825 strcpy(subclass, "Wink"); 00826 break; 00827 case AST_CONTROL_OPTION: 00828 strcpy(subclass, "Option"); 00829 break; 00830 case AST_CONTROL_RADIO_KEY: 00831 strcpy(subclass, "Key Radio"); 00832 break; 00833 case AST_CONTROL_RADIO_UNKEY: 00834 strcpy(subclass, "Unkey Radio"); 00835 break; 00836 case -1: 00837 strcpy(subclass, "Stop generators"); 00838 break; 00839 default: 00840 snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass); 00841 } 00842 break; 00843 case AST_FRAME_NULL: 00844 strcpy(ftype, "Null Frame"); 00845 strcpy(subclass, "N/A"); 00846 break; 00847 case AST_FRAME_IAX: 00848 /* Should never happen */ 00849 strcpy(ftype, "IAX Specific"); 00850 snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass); 00851 break; 00852 case AST_FRAME_TEXT: 00853 strcpy(ftype, "Text"); 00854 strcpy(subclass, "N/A"); 00855 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00856 break; 00857 case AST_FRAME_IMAGE: 00858 strcpy(ftype, "Image"); 00859 snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass)); 00860 break; 00861 case AST_FRAME_HTML: 00862 strcpy(ftype, "HTML"); 00863 switch(f->subclass) { 00864 case AST_HTML_URL: 00865 strcpy(subclass, "URL"); 00866 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00867 break; 00868 case AST_HTML_DATA: 00869 strcpy(subclass, "Data"); 00870 break; 00871 case AST_HTML_BEGIN: 00872 strcpy(subclass, "Begin"); 00873 break; 00874 case AST_HTML_END: 00875 strcpy(subclass, "End"); 00876 break; 00877 case AST_HTML_LDCOMPLETE: 00878 strcpy(subclass, "Load Complete"); 00879 break; 00880 case AST_HTML_NOSUPPORT: 00881 strcpy(subclass, "No Support"); 00882 break; 00883 case AST_HTML_LINKURL: 00884 strcpy(subclass, "Link URL"); 00885 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00886 break; 00887 case AST_HTML_UNLINK: 00888 strcpy(subclass, "Unlink"); 00889 break; 00890 case AST_HTML_LINKREJECT: 00891 strcpy(subclass, "Link Reject"); 00892 break; 00893 default: 00894 snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass); 00895 break; 00896 } 00897 break; 00898 case AST_FRAME_MODEM: 00899 strcpy(ftype, "Modem"); 00900 switch (f->subclass) { 00901 case AST_MODEM_T38: 00902 strcpy(subclass, "T.38"); 00903 break; 00904 case AST_MODEM_V150: 00905 strcpy(subclass, "V.150"); 00906 break; 00907 default: 00908 snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass); 00909 break; 00910 } 00911 break; 00912 default: 00913 snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype); 00914 } 00915 if (!ast_strlen_zero(moreinfo)) 00916 ast_log( LOG_DEBUG, "%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n", 00917 prefix, 00918 ftype, 00919 f->frametype, 00920 subclass, 00921 f->subclass, 00922 moreinfo, 00923 name); 00924 else 00925 ast_log( LOG_DEBUG, "%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n", 00926 prefix, 00927 ftype, 00928 f->frametype, 00929 subclass, 00930 f->subclass, 00931 name); 00932 }
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 335 of file frame.c.
References AST_LIST_INSERT_HEAD, AST_LIST_LOCK, AST_LIST_REMOVE, AST_LIST_UNLOCK, AST_MALLOCD_DATA, AST_MALLOCD_HDR, AST_MALLOCD_SRC, ast_frame::data, FRAME_CACHE_MAX_SIZE, frames, free, ast_frame::mallocd, ast_frame::offset, and ast_frame::src.
Referenced by ast_frfree(), and mixmonitor_thread().
00336 { 00337 if( !fr ) 00338 return; 00339 00340 if (!fr->mallocd) 00341 return; 00342 00343 #if !defined(LOW_MEMORY) 00344 if (cache && fr->mallocd == AST_MALLOCD_HDR) { 00345 /* Cool, only the header is malloc'd, let's just cache those for now 00346 * to keep things simple... */ 00347 struct ast_frame_cache *frames; 00348 00349 if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames))) 00350 && frames->size < FRAME_CACHE_MAX_SIZE) { 00351 AST_LIST_INSERT_HEAD(&frames->list, fr, frame_list); 00352 frames->size++; 00353 return; 00354 } 00355 } 00356 #endif 00357 00358 if (fr->mallocd & AST_MALLOCD_DATA) { 00359 if (fr->data) 00360 free(fr->data - fr->offset); 00361 } 00362 if (fr->mallocd & AST_MALLOCD_SRC) { 00363 if (fr->src) 00364 free((char *)fr->src); 00365 } 00366 if (fr->mallocd & AST_MALLOCD_HDR) { 00367 #ifdef TRACE_FRAMES 00368 AST_LIST_LOCK(&headerlist); 00369 headers--; 00370 AST_LIST_REMOVE(&headerlist, fr, frame_list); 00371 AST_LIST_UNLOCK(&headerlist); 00372 #endif 00373 free(fr); 00374 } 00375 }
Sums two frames of audio samples.
f1 | The first frame (which will contain the result) | |
f2 | The second frame |
Definition at line 1561 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.
Referenced by ast_channel_spy_read_frame(), and ast_write().
01562 { 01563 int count; 01564 short *data1, *data2; 01565 01566 if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR)) 01567 return -1; 01568 01569 if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR)) 01570 return -1; 01571 01572 if (f1->samples != f2->samples) 01573 return -1; 01574 01575 for (count = 0, data1 = f1->data, data2 = f2->data; 01576 count < f1->samples; 01577 count++, data1++, data2++) 01578 ast_slinear_saturated_add(data1, data2); 01579 01580 return 0; 01581 }
Copies a frame.
fr | frame to copy Duplicates a frame -- should only rarely be used, typically frisolate is good enough |
Definition at line 438 of file frame.c.
References ast_calloc_cache, 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, f, frames, ast_frame::frametype, ast_frame::has_timing_info, 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_channel_spy_read_frame(), ast_jb_put(), ast_queue_frame(), ast_rtp_write(), ast_slinfactory_feed(), queue_frame_to_spies(), recordthread(), and rpt().
00439 { 00440 struct ast_frame *out = NULL; 00441 int len, srclen = 0; 00442 void *buf = NULL; 00443 00444 #if !defined(LOW_MEMORY) 00445 struct ast_frame_cache *frames; 00446 #endif 00447 00448 /* Start with standard stuff */ 00449 len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00450 /* If we have a source, add space for it */ 00451 /* 00452 * XXX Watch out here - if we receive a src which is not terminated 00453 * properly, we can be easily attacked. Should limit the size we deal with. 00454 */ 00455 if (f->src) 00456 srclen = strlen(f->src); 00457 if (srclen > 0) 00458 len += srclen + 1; 00459 00460 #if !defined(LOW_MEMORY) 00461 if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) { 00462 AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) { 00463 if (out->mallocd_hdr_len >= len) { 00464 size_t mallocd_len = out->mallocd_hdr_len; 00465 AST_LIST_REMOVE_CURRENT(&frames->list, frame_list); 00466 memset(out, 0, sizeof(*out)); 00467 out->mallocd_hdr_len = mallocd_len; 00468 buf = out; 00469 frames->size--; 00470 break; 00471 } 00472 } 00473 AST_LIST_TRAVERSE_SAFE_END 00474 } 00475 #endif 00476 00477 if (!buf) { 00478 if (!(buf = ast_calloc_cache(1, len))) 00479 return NULL; 00480 out = buf; 00481 out->mallocd_hdr_len = len; 00482 } 00483 00484 out->frametype = f->frametype; 00485 out->subclass = f->subclass; 00486 out->datalen = f->datalen; 00487 out->samples = f->samples; 00488 out->delivery = f->delivery; 00489 /* Set us as having malloc'd header only, so it will eventually 00490 get freed. */ 00491 out->mallocd = AST_MALLOCD_HDR; 00492 out->offset = AST_FRIENDLY_OFFSET; 00493 if (out->datalen) { 00494 out->data = buf + sizeof(*out) + AST_FRIENDLY_OFFSET; 00495 memcpy(out->data, f->data, out->datalen); 00496 } 00497 if (srclen > 0) { 00498 out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00499 /* Must have space since we allocated for it */ 00500 strcpy((char *)out->src, f->src); 00501 } 00502 out->has_timing_info = f->has_timing_info; 00503 out->ts = f->ts; 00504 out->len = f->len; 00505 out->seqno = f->seqno; 00506 return out; 00507 }
static void force_inline ast_frfree | ( | struct ast_frame * | fr | ) | [static] |
Definition at line 390 of file frame.h.
References ast_frame_free().
Referenced by __adsi_transmit_messages(), __ast_play_and_record(), __ast_read(), __ast_request_and_dial(), adsi_careful_send(), agent_ack_sleep(), agent_read(), app_exec(), ast_bridge_call(), ast_channel_free(), ast_channel_spy_free(), ast_channel_spy_read_frame(), ast_dsp_process(), ast_feature_request_and_dial(), ast_jb_destroy(), ast_jb_put(), ast_masq_park_call(), ast_queue_frame(), ast_recvtext(), ast_rtp_write(), ast_safe_sleep_conditional(), ast_send_image(), ast_slinfactory_destroy(), ast_slinfactory_read(), ast_tonepair(), ast_translate(), ast_udptl_bridge(), ast_waitfordigit_full(), ast_write(), async_wait(), 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(), copy_data_from_queue(), create_jb(), dictate_exec(), disa_exec(), do_parking_thread(), do_waiting(), echo_exec(), find_cache(), gen_generate(), handle_invite_replaces(), handle_recordfile(), iax_park_thread(), ices_exec(), isAnsweringMachine(), jb_get_and_deliver(), measurenoise(), moh_files_generator(), monitor_dial(), mp3_exec(), mwanalyze_exec(), NBScat_exec(), nv_background_detect_exec(), nv_detectfax_exec(), queue_frame_to_spies(), receive_dtmf_digits(), recordthread(), rpt(), run_agi(), rxfax_exec(), send_tone_burst(), send_waveform_to_channel(), sendurl_exec(), speech_background(), spy_generate(), ss_thread(), txfax_exec(), wait_for_answer(), wait_for_hangup(), wait_for_winner(), waitforring_exec(), waitstream_core(), and zt_bridge().
00391 { 00392 ast_frame_free(fr, 1); 00393 }
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 382 of file frame.c.
References ast_frame_header_new(), AST_FRIENDLY_OFFSET, ast_malloc, AST_MALLOCD_DATA, AST_MALLOCD_HDR, AST_MALLOCD_SRC, ast_strdup, ast_frame::data, ast_frame::datalen, ast_frame::frametype, free, ast_frame::has_timing_info, 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().
00383 { 00384 struct ast_frame *out; 00385 void *newdata; 00386 00387 if (!(fr->mallocd & AST_MALLOCD_HDR)) { 00388 /* Allocate a new header if needed */ 00389 if (!(out = ast_frame_header_new())) 00390 return NULL; 00391 out->frametype = fr->frametype; 00392 out->subclass = fr->subclass; 00393 out->datalen = fr->datalen; 00394 out->samples = fr->samples; 00395 out->offset = fr->offset; 00396 out->data = fr->data; 00397 /* Copy the timing data */ 00398 out->has_timing_info = fr->has_timing_info; 00399 if (fr->has_timing_info) { 00400 out->ts = fr->ts; 00401 out->len = fr->len; 00402 out->seqno = fr->seqno; 00403 } 00404 } else 00405 out = fr; 00406 00407 if (!(fr->mallocd & AST_MALLOCD_SRC)) { 00408 if (fr->src) { 00409 if (!(out->src = ast_strdup(fr->src))) { 00410 if (out != fr) 00411 free(out); 00412 return NULL; 00413 } 00414 } 00415 } else 00416 out->src = fr->src; 00417 00418 if (!(fr->mallocd & AST_MALLOCD_DATA)) { 00419 if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) { 00420 if (out->src != fr->src) 00421 free((void *) out->src); 00422 if (out != fr) 00423 free(out); 00424 return NULL; 00425 } 00426 newdata += AST_FRIENDLY_OFFSET; 00427 out->offset = AST_FRIENDLY_OFFSET; 00428 out->datalen = fr->datalen; 00429 memcpy(newdata, fr->data, fr->datalen); 00430 out->data = newdata; 00431 } 00432 00433 out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA; 00434 00435 return out; 00436 }
struct ast_format_list* ast_get_format_list | ( | size_t * | size | ) |
Definition at line 525 of file frame.c.
References AST_FORMAT_LIST.
00526 { 00527 *size = (sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0])); 00528 return AST_FORMAT_LIST; 00529 }
struct ast_format_list* ast_get_format_list_index | ( | int | index | ) |
Definition at line 520 of file frame.c.
References AST_FORMAT_LIST.
00521 { 00522 return &AST_FORMAT_LIST[index]; 00523 }
int ast_getformatbyname | ( | const char * | name | ) |
Gets a format from a name.
name | string of format |
Definition at line 591 of file frame.c.
References ast_expand_codec_alias(), AST_FORMAT_LIST, and format.
Referenced by ast_parse_allow_disallow(), iax_template_parse(), load_moh_classes(), reload_config(), and try_suggested_sip_codec().
00592 { 00593 int x, all, format = 0; 00594 00595 all = strcasecmp(name, "all") ? 0 : 1; 00596 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00597 if(AST_FORMAT_LIST[x].visible && (all || 00598 !strcasecmp(AST_FORMAT_LIST[x].name,name) || 00599 !strcasecmp(AST_FORMAT_LIST[x].name,ast_expand_codec_alias(name)))) { 00600 format |= AST_FORMAT_LIST[x].bits; 00601 if(!all) 00602 break; 00603 } 00604 } 00605 00606 return format; 00607 }
char* ast_getformatname | ( | int | format | ) |
Get the name of a format.
format | id of format |
Definition at line 531 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_channels(), _sip_show_peer(), add_codec_to_answer(), add_codec_to_sdp(), agent_call(), ast_channel_getformatname(), ast_channel_spy_add(), 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(), iax2_request(), iax2_show_channels(), iax2_show_peer(), iax_show_provisioning(), moh_classes_show(), moh_release(), nv_background_detect_exec(), nv_detectfax_exec(), oh323_rtp_read(), phone_setup(), print_codec_to_cli(), queue_frame_to_spies(), 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(), and zt_read().
00532 { 00533 int x; 00534 char *ret = "unknown"; 00535 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00536 if(AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == format) { 00537 ret = AST_FORMAT_LIST[x].name; 00538 break; 00539 } 00540 } 00541 return ret; 00542 }
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 544 of file frame.c.
References AST_FORMAT_LIST, ast_format_list::bits, len, name, and ast_format_list::visible.
Referenced by _sip_show_peer(), add_sdp(), ast_channel_getformatname_multiple(), ast_streamfile(), function_iaxpeer(), function_sippeer(), handle_showchan(), handle_showchan_deprecated(), iax2_show_peer(), process_sdp(), serialize_showchan(), sip_new(), sip_request_call(), sip_show_channel(), sip_show_settings(), and sip_write().
00545 { 00546 int x; 00547 unsigned len; 00548 char *start, *end = buf; 00549 00550 if (!size) 00551 return buf; 00552 snprintf(end, size, "0x%x (", format); 00553 len = strlen(end); 00554 end += len; 00555 size -= len; 00556 start = end; 00557 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00558 if (AST_FORMAT_LIST[x].visible && (AST_FORMAT_LIST[x].bits & format)) { 00559 snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name); 00560 len = strlen(end); 00561 end += len; 00562 size -= len; 00563 } 00564 } 00565 if (start == end) 00566 snprintf(start, size, "nothing)"); 00567 else if (size > 1) 00568 *(end -1) = ')'; 00569 return buf; 00570 }
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 1268 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().
01269 { 01270 char *parse = NULL, *this = NULL, *psize = NULL; 01271 int format = 0, framems = 0; 01272 01273 parse = ast_strdupa(list); 01274 while ((this = strsep(&parse, ","))) { 01275 framems = 0; 01276 if ((psize = strrchr(this, ':'))) { 01277 *psize++ = '\0'; 01278 if (option_debug) 01279 ast_log(LOG_DEBUG,"Packetization for codec: %s is %s\n", this, psize); 01280 framems = atoi(psize); 01281 if (framems < 0) 01282 framems = 0; 01283 } 01284 if (!(format = ast_getformatbyname(this))) { 01285 ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this); 01286 continue; 01287 } 01288 01289 if (mask) { 01290 if (allowing) 01291 *mask |= format; 01292 else 01293 *mask &= ~format; 01294 } 01295 01296 /* Set up a preference list for audio. Do not include video in preferences 01297 since we can not transcode video and have to use whatever is offered 01298 */ 01299 if (pref && (format & AST_FORMAT_AUDIO_MASK)) { 01300 if (strcasecmp(this, "all")) { 01301 if (allowing) { 01302 ast_codec_pref_append(pref, format); 01303 ast_codec_pref_setsize(pref, format, framems); 01304 } 01305 else 01306 ast_codec_pref_remove(pref, format); 01307 } else if (!allowing) { 01308 memset(pref, 0, sizeof(*pref)); 01309 } 01310 } 01311 } 01312 }
void ast_smoother_free | ( | struct ast_smoother * | s | ) |
Definition at line 282 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 146 of file frame.c.
References ast_malloc, ast_smoother_reset(), and s.
Referenced by ast_rtp_write().
00147 { 00148 struct ast_smoother *s; 00149 if (size < 1) 00150 return NULL; 00151 if ((s = ast_malloc(sizeof(*s)))) 00152 ast_smoother_reset(s, size); 00153 return s; 00154 }
struct ast_frame* ast_smoother_read | ( | struct ast_smoother * | s | ) |
Definition at line 232 of file frame.c.
References AST_FRAME_VOICE, AST_FRIENDLY_OFFSET, ast_log(), AST_SMOOTHER_FLAG_G729, ast_tvadd(), len, LOG_WARNING, and s.
Referenced by ast_rtp_write().
00233 { 00234 struct ast_frame *opt; 00235 int len; 00236 00237 /* IF we have an optimization frame, send it */ 00238 if (s->opt) { 00239 if (s->opt->offset < AST_FRIENDLY_OFFSET) 00240 ast_log(LOG_WARNING, "Returning a frame of inappropriate offset (%d).\n", 00241 s->opt->offset); 00242 opt = s->opt; 00243 s->opt = NULL; 00244 return opt; 00245 } 00246 00247 /* Make sure we have enough data */ 00248 if (s->len < s->size) { 00249 /* Or, if this is a G.729 frame with VAD on it, send it immediately anyway */ 00250 if (!((s->flags & AST_SMOOTHER_FLAG_G729) && (s->size % 10))) 00251 return NULL; 00252 } 00253 len = s->size; 00254 if (len > s->len) 00255 len = s->len; 00256 /* Make frame */ 00257 s->f.frametype = AST_FRAME_VOICE; 00258 s->f.subclass = s->format; 00259 s->f.data = s->framedata + AST_FRIENDLY_OFFSET; 00260 s->f.offset = AST_FRIENDLY_OFFSET; 00261 s->f.datalen = len; 00262 /* Samples will be improper given VAD, but with VAD the concept really doesn't even exist */ 00263 s->f.samples = len * s->samplesperbyte; /* XXX rounding */ 00264 s->f.delivery = s->delivery; 00265 /* Fill Data */ 00266 memcpy(s->f.data, s->data, len); 00267 s->len -= len; 00268 /* Move remaining data to the front if applicable */ 00269 if (s->len) { 00270 /* In principle this should all be fine because if we are sending 00271 G.729 VAD, the next timestamp will take over anyawy */ 00272 memmove(s->data, s->data + len, s->len); 00273 if (!ast_tvzero(s->delivery)) { 00274 /* If we have delivery time, increment it, otherwise, leave it at 0 */ 00275 s->delivery = ast_tvadd(s->delivery, ast_samp2tv(s->f.samples, 8000)); 00276 } 00277 } 00278 /* Return frame */ 00279 return &s->f; 00280 }
void ast_smoother_reset | ( | struct ast_smoother * | s, | |
int | bytes | |||
) |
Definition at line 140 of file frame.c.
References s.
Referenced by ast_smoother_new().
00141 { 00142 memset(s, 0, sizeof(*s)); 00143 s->size = size; 00144 }
void ast_smoother_set_flags | ( | struct ast_smoother * | smoother, | |
int | flags | |||
) |
Definition at line 161 of file frame.c.
References s.
Referenced by ast_rtp_write().
00162 { 00163 s->flags = flags; 00164 }
int ast_smoother_test_flag | ( | struct ast_smoother * | s, | |
int | flag | |||
) |
Definition at line 166 of file frame.c.
References s.
Referenced by ast_rtp_write().
00167 { 00168 return (s->flags & flag); 00169 }
void ast_swapcopy_samples | ( | void * | dst, | |
const void * | src, | |||
int | samples | |||
) |
Definition at line 509 of file frame.c.
Referenced by __ast_smoother_feed(), iax_frame_wrap(), and phone_write_buf().
00510 { 00511 int i; 00512 unsigned short *dst_s = dst; 00513 const unsigned short *src_s = src; 00514 00515 for (i = 0; i < samples; i++) 00516 dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8); 00517 }
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 138 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_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_sdp(), send_dtmf(), sip_rtp_read(), skinny_rtp_read(), and wakeup_sub().