#include <sys/time.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 |
AST_Smoother | |
#define | ast_smoother_feed(s, f) __ast_smoother_feed(s, f, 0) |
#define | ast_smoother_feed_be(s, f) __ast_smoother_feed(s, f, 0) |
#define | ast_smoother_feed_le(s, f) __ast_smoother_feed(s, f, 1) |
int | __ast_smoother_feed (struct ast_smoother *s, struct ast_frame *f, int swap) |
void | ast_smoother_free (struct ast_smoother *s) |
int | ast_smoother_get_flags (struct ast_smoother *smoother) |
ast_smoother * | ast_smoother_new (int bytes) |
ast_frame * | ast_smoother_read (struct ast_smoother *s) |
void | ast_smoother_reconfigure (struct ast_smoother *s, int bytes) |
Reconfigure an existing smoother to output a different number of bytes per frame. | |
void | ast_smoother_reset (struct ast_smoother *s, int bytes) |
void | ast_smoother_set_flags (struct ast_smoother *smoother, int flags) |
int | ast_smoother_test_flag (struct ast_smoother *s, int flag) |
Defines | |
#define | AST_FORMAT_ADPCM (1 << 5) |
#define | AST_FORMAT_ALAW (1 << 3) |
#define | AST_FORMAT_AUDIO_MASK ((1 << 16)-1) |
#define | AST_FORMAT_AUDIO_UNDEFINED ((1 << 13) | (1 << 14)) |
#define | AST_FORMAT_G722 (1 << 12) |
#define | AST_FORMAT_G723_1 (1 << 0) |
#define | AST_FORMAT_G726 (1 << 11) |
#define | AST_FORMAT_G726_AAL2 (1 << 4) |
#define | AST_FORMAT_G729A (1 << 8) |
#define | AST_FORMAT_GSM (1 << 1) |
#define | AST_FORMAT_H261 (1 << 18) |
#define | AST_FORMAT_H263 (1 << 19) |
#define | AST_FORMAT_H263_PLUS (1 << 20) |
#define | AST_FORMAT_H264 (1 << 21) |
#define | AST_FORMAT_ILBC (1 << 10) |
#define | AST_FORMAT_JPEG (1 << 16) |
#define | AST_FORMAT_LPC10 (1 << 7) |
#define | AST_FORMAT_MAX_TEXT (1 << 28)) |
#define | AST_FORMAT_MP4_VIDEO (1 << 22) |
#define | AST_FORMAT_PNG (1 << 17) |
#define | AST_FORMAT_SLINEAR (1 << 6) |
#define | AST_FORMAT_SLINEAR16 (1 << 15) |
#define | AST_FORMAT_SPEEX (1 << 9) |
#define | AST_FORMAT_T140 (1 << 27) |
#define | AST_FORMAT_T140RED (1 << 26) |
#define | AST_FORMAT_TEXT_MASK (((1 << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK)) |
#define | AST_FORMAT_ULAW (1 << 2) |
#define | AST_FORMAT_VIDEO_MASK (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) |
#define | ast_frame_byteswap_be(fr) do { ; } while(0) |
#define | ast_frame_byteswap_le(fr) do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data.ptr, __f->data.ptr, __f->samples); } while(0) |
#define | AST_FRAME_DTMF AST_FRAME_DTMF_END |
#define | AST_FRAME_SET_BUFFER(fr, _base, _ofs, _datalen) |
#define | ast_frfree(fr) ast_frame_free(fr, 1) |
#define | AST_FRIENDLY_OFFSET 64 |
Offset into a frame's data buffer. | |
#define | AST_HTML_BEGIN 4 |
#define | AST_HTML_DATA 2 |
#define | AST_HTML_END 8 |
#define | AST_HTML_LDCOMPLETE 16 |
#define | AST_HTML_LINKREJECT 20 |
#define | AST_HTML_LINKURL 18 |
#define | AST_HTML_NOSUPPORT 17 |
#define | AST_HTML_UNLINK 19 |
#define | AST_HTML_URL 1 |
#define | AST_MALLOCD_DATA (1 << 1) |
#define | AST_MALLOCD_HDR (1 << 0) |
#define | AST_MALLOCD_SRC (1 << 2) |
#define | AST_MIN_OFFSET 32 |
#define | AST_MODEM_T38 1 |
#define | AST_MODEM_V150 2 |
#define | AST_OPTION_AUDIO_MODE 4 |
#define | AST_OPTION_ECHOCAN 8 |
#define | AST_OPTION_FLAG_ACCEPT 1 |
#define | AST_OPTION_FLAG_ANSWER 5 |
#define | AST_OPTION_FLAG_QUERY 4 |
#define | AST_OPTION_FLAG_REJECT 2 |
#define | AST_OPTION_FLAG_REQUEST 0 |
#define | AST_OPTION_FLAG_WTF 6 |
#define | AST_OPTION_OPRMODE 7 |
#define | AST_OPTION_RELAXDTMF 3 |
#define | AST_OPTION_RXGAIN 6 |
#define | AST_OPTION_T38_STATE 10 |
#define | AST_OPTION_TDD 2 |
#define | AST_OPTION_TONE_VERIFY 1 |
#define | AST_OPTION_TXGAIN 5 |
#define | AST_SMOOTHER_FLAG_BE (1 << 1) |
#define | AST_SMOOTHER_FLAG_G729 (1 << 0) |
Enumerations | |
enum | { AST_FRFLAG_HAS_TIMING_INFO = (1 << 0), AST_FRFLAG_FROM_TRANSLATOR = (1 << 1), AST_FRFLAG_FROM_DSP = (1 << 2), 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_T38 = 19, AST_CONTROL_SRCUPDATE = 20 } |
enum | ast_control_t38 { AST_T38_REQUEST_NEGOTIATE = 1, AST_T38_REQUEST_TERMINATE, AST_T38_NEGOTIATED, AST_T38_TERMINATED, AST_T38_REFUSED } |
enum | ast_frame_type { AST_FRAME_DTMF_END = 1, AST_FRAME_VOICE, AST_FRAME_VIDEO, AST_FRAME_CONTROL, AST_FRAME_NULL, AST_FRAME_IAX, AST_FRAME_TEXT, AST_FRAME_IMAGE, AST_FRAME_HTML, AST_FRAME_CNG, AST_FRAME_MODEM, AST_FRAME_DTMF_BEGIN } |
Frame types. More... | |
Functions | |
char * | ast_codec2str (int codec) |
Get a name from a format Gets a name from a format. | |
int | ast_codec_choose (struct ast_codec_pref *pref, int formats, int find_best) |
Select the best audio format according to preference list from supplied options. If "find_best" is non-zero then if nothing is found, the "Best" format of the format list is selected, otherwise 0 is returned. | |
int | ast_codec_get_len (int format, int samples) |
Returns the number of bytes for the number of samples of the given format. | |
int | ast_codec_get_samples (struct ast_frame *f) |
Returns the number of samples contained in the frame. | |
static int | ast_codec_interp_len (int format) |
Gets duration in ms of interpolation frame for a format. | |
int | ast_codec_pref_append (struct ast_codec_pref *pref, int format) |
Append a audio codec to a preference list, removing it first if it was already there. | |
void | ast_codec_pref_convert (struct ast_codec_pref *pref, char *buf, size_t size, int right) |
Shift an audio codec preference list up or down 65 bytes so that it becomes an ASCII string. | |
ast_format_list | ast_codec_pref_getsize (struct ast_codec_pref *pref, int format) |
Get packet size for codec. | |
int | ast_codec_pref_index (struct ast_codec_pref *pref, int index) |
Codec located at a particular place in the preference index. | |
void | ast_codec_pref_init (struct ast_codec_pref *pref) |
Initialize an audio codec preference to "no preference". | |
void | ast_codec_pref_prepend (struct ast_codec_pref *pref, int format, int only_if_existing) |
Prepend an audio codec to a preference list, removing it first if it was already there. | |
void | ast_codec_pref_remove (struct ast_codec_pref *pref, int format) |
Remove audio a codec from a preference list. | |
int | ast_codec_pref_setsize (struct ast_codec_pref *pref, int format, int framems) |
Set packet size for codec. | |
int | ast_codec_pref_string (struct ast_codec_pref *pref, char *buf, size_t size) |
Dump audio codec preference list into a string. | |
static force_inline int | ast_format_rate (int format) |
Get the sample rate for a given format. | |
int | ast_frame_adjust_volume (struct ast_frame *f, int adjustment) |
Adjusts the volume of the audio samples contained in a frame. | |
void | ast_frame_dump (const char *name, struct ast_frame *f, char *prefix) |
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. | |
int | ast_parse_allow_disallow (struct ast_codec_pref *pref, int *mask, const char *list, int allowing) |
Parse an "allow" or "deny" line in a channel or device configuration and update the capabilities mask and pref if provided. Video codecs are not added to codec preference lists, since we can not transcode. | |
void | ast_swapcopy_samples (void *dst, const void *src, int samples) |
Variables | |
ast_frame | ast_null_frame |
Definition in file frame.h.
#define AST_FORMAT_ADPCM (1 << 5) |
ADPCM (IMA)
Definition at line 255 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 251 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(), find_transcoders(), is_encoder(), misdn_read(), misdn_set_opt_exec(), oh323_rtp_read(), pcm_seek(), pcm_write(), read_config(), and start_rtp().
#define AST_FORMAT_AUDIO_MASK ((1 << 16)-1) |
Maximum audio mask
Definition at line 275 of file frame.h.
Referenced by add_sdp(), ast_best_codec(), ast_channel_make_compatible_helper(), ast_codec_choose(), ast_filehelper(), ast_openstream_full(), ast_openvstream(), ast_parse_allow_disallow(), ast_playstream(), ast_request(), ast_rtp_read(), ast_translate_available_formats(), ast_translator_best_choice(), ast_writestream(), begin_dial_channel(), filestream_destructor(), func_channel_read(), generator_force(), gtalk_rtp_read(), jingle_rtp_read(), oh323_request(), phone_read(), process_sdp(), set_format(), sip_call(), sip_request_call(), sip_rtp_read(), sip_write(), skinny_request(), transmit_connect_with_sdp(), and transmit_modify_with_sdp().
#define AST_FORMAT_AUDIO_UNDEFINED ((1 << 13) | (1 << 14)) |
#define AST_FORMAT_G722 (1 << 12) |
G.722
Definition at line 269 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_write(), ast_slinfactory_feed(), au_seek(), convertcap(), g722tolin16_sample(), g722tolin_sample(), and pcm_read().
#define AST_FORMAT_G723_1 (1 << 0) |
G.723.1 compression
Definition at line 245 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_write(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_destroy(), dahdi_translate(), g723_read(), g723_write(), load_module(), phone_request(), phone_setup(), phone_write(), register_translator(), and start_rtp().
#define AST_FORMAT_G726 (1 << 11) |
ADPCM (G.726, 32kbps, RFC3551 codeword packing)
Definition at line 267 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 253 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(), codec_skinny2ast(), and setup_rtp_connection().
#define AST_FORMAT_G729A (1 << 8) |
G.729A audio
Definition at line 261 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_destroy(), dahdi_translate(), g729_read(), g729_write(), load_module(), phone_request(), phone_setup(), phone_write(), and start_rtp().
#define AST_FORMAT_GSM (1 << 1) |
GSM compression
Definition at line 247 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 281 of file frame.h.
Referenced by codec_ast2skinny(), codec_skinny2ast(), and h261_encap().
#define AST_FORMAT_H263 (1 << 19) |
H.263 Video
Definition at line 283 of file frame.h.
Referenced by codec_ast2skinny(), codec_skinny2ast(), h263_encap(), h263_read(), and h263_write().
#define AST_FORMAT_H263_PLUS (1 << 20) |
#define AST_FORMAT_H264 (1 << 21) |
H.264 Video
Definition at line 287 of file frame.h.
Referenced by h264_encap(), h264_read(), and h264_write().
#define AST_FORMAT_ILBC (1 << 10) |
iLBC Free Compression
Definition at line 265 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 277 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 259 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_samples(), and lpc10tolin_sample().
#define AST_FORMAT_MP4_VIDEO (1 << 22) |
#define AST_FORMAT_PNG (1 << 17) |
#define AST_FORMAT_SLINEAR (1 << 6) |
Raw 16-bit Signed Linear (8000 Hz) PCM
Definition at line 257 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_helper(), ast_channel_start_silence_generator(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_call_progress(), ast_dsp_noise(), ast_dsp_process(), ast_dsp_silence(), ast_frame_adjust_volume(), ast_frame_slinear_sum(), ast_rtp_read(), ast_slinfactory_feed(), ast_speech_new(), 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_read(), dahdi_translate(), dahdi_write(), dictate_exec(), do_waiting(), eagi_exec(), extenspy_exec(), fax_generator_generate(), find_transcoders(), handle_jack_audio(), handle_recordfile(), handle_speechcreate(), handle_speechrecognize(), iax_frame_wrap(), ices_exec(), init_outgoing(), is_encoder(), isAnsweringMachine(), jack_hook_callback(), 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(), local_ast_moh_start(), 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_exec(), rpt_tele_thread(), send_waveform_to_channel(), silence_generator_generate(), slin8_to_slin16_sample(), slinear_read(), slinear_write(), socket_process(), speech_background(), spy_generate(), tonepair_alloc(), transmit_audio(), usbradio_new(), usbradio_read(), usbradio_request(), wav_read(), and wav_write().
#define AST_FORMAT_SLINEAR16 (1 << 15) |
Raw 16-bit Signed Linear (16000 Hz) PCM
Definition at line 273 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_slinfactory_feed(), console_new(), lin16tog722_sample(), slin16_to_slin8_sample(), slinear_read(), slinear_write(), and stream_monitor().
#define AST_FORMAT_SPEEX (1 << 9) |
SpeeX Free Compression
Definition at line 263 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_samples(), ast_rtp_write(), convertcap(), and speextolin_sample().
#define AST_FORMAT_T140 (1 << 27) |
T.140 Text format - ITU T.140, RFC 4103
Definition at line 294 of file frame.h.
Referenced by add_tcodec_to_sdp(), ast_rtp_read(), and ast_write().
#define AST_FORMAT_T140RED (1 << 26) |
T.140 RED Text format RFC 4103
Definition at line 292 of file frame.h.
Referenced by add_tcodec_to_sdp(), ast_rtp_read(), process_sdp(), and rtp_red_init().
#define AST_FORMAT_TEXT_MASK (((1 << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK)) |
Definition at line 297 of file frame.h.
Referenced by add_sdp(), ast_request(), check_peer_ok(), sip_new(), and sip_rtp_read().
#define AST_FORMAT_ULAW (1 << 2) |
Raw mu-law data (G.711)
Definition at line 249 of file frame.h.
Referenced by __adsi_transmit_messages(), _ast_adsi_transmit_message_full(), adsi_careful_send(), alarmreceiver_exec(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), calc_energy(), codec_ast2skinny(), codec_skinny2ast(), conf_run(), convertcap(), dahdi_new(), dahdi_read(), dahdi_translate(), dahdi_write(), find_transcoders(), is_encoder(), load_module(), milliwatt_generate(), oh323_rtp_read(), old_milliwatt_exec(), phone_request(), phone_setup(), phone_write(), pri_dchannel(), send_tone_burst(), start_rtp(), ulawtoalaw_sample(), and ulawtolin_sample().
#define AST_FORMAT_VIDEO_MASK (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) |
Definition at line 290 of file frame.h.
Referenced by add_sdp(), ast_filehelper(), ast_openvstream(), ast_request(), ast_rtp_read(), ast_translate_available_formats(), check_peer_ok(), create_addr_from_peer(), func_channel_read(), gtalk_new(), gtalk_rtp_read(), jingle_new(), jingle_rtp_read(), sip_new(), and sip_rtp_read().
#define ast_frame_byteswap_be | ( | fr | ) | do { ; } while(0) |
#define ast_frame_byteswap_le | ( | fr | ) | do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data.ptr, __f->data.ptr, __f->samples); } while(0) |
#define AST_FRAME_DTMF AST_FRAME_DTMF_END |
Definition at line 124 of file frame.h.
Referenced by __adsi_transmit_messages(), __ast_play_and_record(), action_atxfer(), action_dahdidialoffhook(), agent_ack_sleep(), ast_audiohook_write_list(), ast_bridge_call(), ast_dsp_process(), ast_feature_request_and_dial(), ast_jb_put(), background_detect_exec(), cb_events(), channel_spy(), cli_console_dial(), conf_exec(), conf_run(), console_dial(), dahdi_bridge(), dahdi_read(), dictate_exec(), disa_exec(), do_immediate_setup(), echo_exec(), eivr_comm(), gtalk_handle_dtmf(), handle_recordfile(), handle_request(), handle_request_info(), handle_speechrecognize(), jingle_handle_dtmf(), keypad_digit(), mgcp_rtp_read(), misdn_bridge(), mp3_exec(), NBScat_exec(), oh323_rtp_read(), phone_exception(), process_ast_dsp(), receive_dtmf_digits(), rpt(), rpt_call(), send_waveform_to_channel(), sip_rtp_read(), speech_background(), ss_thread(), transmit_audio(), unistim_do_senddigit(), unistim_senddigit_end(), volume_callback(), and wait_for_winner().
#define AST_FRAME_SET_BUFFER | ( | fr, | |||
_base, | |||||
_ofs, | |||||
_datalen | ) |
Value:
{ \ (fr)->data.ptr = (char *)_base + (_ofs); \ (fr)->offset = (_ofs); \ (fr)->datalen = (_datalen); \ }
Definition at line 186 of file frame.h.
Referenced by fax_generator_generate(), g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), ilbc_read(), ogg_vorbis_read(), pcm_read(), slinear_read(), t38_tx_packet_handler(), vox_read(), and wav_read().
#define ast_frfree | ( | fr | ) | ast_frame_free(fr, 1) |
Definition at line 438 of file frame.h.
Referenced by __adsi_transmit_messages(), __ast_answer(), __ast_play_and_record(), __ast_queue_frame(), __ast_read(), __ast_request_and_dial(), adsi_careful_send(), agent_ack_sleep(), agent_read(), ast_audiohook_read_frame(), ast_autoservice_stop(), ast_bridge_call(), ast_channel_free(), ast_dsp_process(), ast_feature_request_and_dial(), ast_jb_destroy(), ast_jb_put(), ast_readaudio_callback(), ast_readvideo_callback(), ast_recvtext(), ast_rtp_write(), ast_safe_sleep_conditional(), ast_send_image(), ast_slinfactory_destroy(), ast_slinfactory_feed(), ast_slinfactory_flush(), ast_slinfactory_read(), ast_tonepair(), ast_translate(), ast_udptl_bridge(), ast_waitfordigit_full(), ast_write(), ast_writestream(), async_wait(), audio_audiohook_write_list(), autoservice_run(), background_detect_exec(), bridge_native_loop(), bridge_p2p_loop(), builtin_atxfer(), calc_cost(), channel_spy(), check_goto_on_transfer(), conf_exec(), conf_flush(), conf_free(), conf_run(), create_jb(), dahdi_bridge(), dictate_exec(), disa_exec(), do_idle_thread(), do_waiting(), echo_exec(), eivr_comm(), find_cache(), gen_generate(), handle_invite_replaces(), handle_recordfile(), handle_speechrecognize(), iax_park_thread(), ices_exec(), isAnsweringMachine(), jb_empty_and_reset_adaptive(), jb_empty_and_reset_fixed(), jb_get_and_deliver(), launch_asyncagi(), manage_parkinglot(), masq_park_call(), measurenoise(), moh_files_generator(), monitor_dial(), mp3_exec(), NBScat_exec(), receive_dtmf_digits(), recordthread(), rpt(), run_agi(), send_tone_burst(), send_waveform_to_channel(), sendurl_exec(), speech_background(), spy_generate(), ss_thread(), transmit_audio(), transmit_t38(), wait_for_answer(), wait_for_hangup(), wait_for_winner(), waitforring_exec(), and waitstream_core().
#define AST_FRIENDLY_OFFSET 64 |
Offset into a frame's data buffer.
By providing some "empty" space prior to the actual data of an ast_frame, this gives any consumer of the frame ample space to prepend other necessary information without having to create a new buffer.
As an example, RTP can use the data from an ast_frame and simply prepend the RTP header information into the space provided by AST_FRIENDLY_OFFSET instead of having to create a new buffer with the necessary space allocated.
Definition at line 207 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_decoder_frameout(), dahdi_encoder_frameout(), dahdi_read(), fax_generator_generate(), g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), iax_frame_wrap(), ilbc_read(), jb_get_and_deliver(), linear_generator(), milliwatt_generate(), moh_generate(), mohalloc(), mp3_exec(), NBScat_exec(), newpvt(), ogg_vorbis_read(), oss_read(), pcm_read(), phone_read(), process_rfc3389(), send_tone_burst(), send_waveform_to_channel(), slinear_read(), sms_generate(), usbradio_read(), vox_read(), and wav_read().
#define AST_HTML_BEGIN 4 |
#define AST_HTML_DATA 2 |
#define AST_HTML_END 8 |
#define AST_HTML_LDCOMPLETE 16 |
Load is complete
Definition at line 233 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 235 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 225 of file frame.h.
Referenced by ast_channel_sendurl(), ast_frame_dump(), and sip_sendhtml().
#define AST_MALLOCD_DATA (1 << 1) |
Need the data be free'd?
Definition at line 213 of file frame.h.
Referenced by ast_frame_free(), ast_frisolate(), and create_video_frame().
#define AST_MALLOCD_HDR (1 << 0) |
Need the header be free'd?
Definition at line 211 of file frame.h.
Referenced by ast_frame_free(), ast_frame_header_new(), ast_frdup(), ast_frisolate(), and create_video_frame().
#define AST_MALLOCD_SRC (1 << 2) |
Need the source be free'd? (haha!)
Definition at line 215 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 219 of file frame.h.
Referenced by ast_frame_dump(), t38_tx_packet_handler(), transmit_t38(), and udptl_rx_packet().
#define AST_MODEM_V150 2 |
#define AST_OPTION_AUDIO_MODE 4 |
Set (or clear) Audio (Not-Clear) Mode
Definition at line 352 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 374 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 349 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 368 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_T38_STATE 10 |
Definition at line 380 of file frame.h.
Referenced by ast_channel_get_t38_state(), and sip_queryoption().
#define AST_OPTION_TDD 2 |
Put a compatible channel into TDD (TTY for the hearing-impared) mode
Definition at line 346 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 343 of file frame.h.
Referenced by conf_run(), dahdi_hangup(), dahdi_setoption(), rpt(), rpt_exec(), and try_calling().
#define AST_OPTION_TXGAIN 5 |
Set channel transmit gain Option data is a single signed char representing number of decibels (dB) to set gain to (on top of any gain specified in channel driver)
Definition at line 360 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 330 of file frame.h.
Referenced by __ast_smoother_feed(), ast_smoother_read(), and smoother_frame_feed().
anonymous enum |
Definition at line 126 of file frame.h.
00126 { 00127 /*! This frame contains valid timing information */ 00128 AST_FRFLAG_HAS_TIMING_INFO = (1 << 0), 00129 /*! This frame came from a translator and is still the original frame. 00130 * The translator can not be free'd if the frame inside of it still has 00131 * this flag set. */ 00132 AST_FRFLAG_FROM_TRANSLATOR = (1 << 1), 00133 /*! This frame came from a dsp and is still the original frame. 00134 * The dsp cannot be free'd if the frame inside of it still has 00135 * this flag set. */ 00136 AST_FRFLAG_FROM_DSP = (1 << 2), 00137 /*! This frame came from a filestream and is still the original frame. 00138 * The filestream cannot be free'd if the frame inside of it still has 00139 * this flag set. */ 00140 AST_FRFLAG_FROM_FILESTREAM = (1 << 3), 00141 };
Definition at line 299 of file frame.h.
00299 { 00300 AST_CONTROL_HANGUP = 1, /*!< Other end has hungup */ 00301 AST_CONTROL_RING = 2, /*!< Local ring */ 00302 AST_CONTROL_RINGING = 3, /*!< Remote end is ringing */ 00303 AST_CONTROL_ANSWER = 4, /*!< Remote end has answered */ 00304 AST_CONTROL_BUSY = 5, /*!< Remote end is busy */ 00305 AST_CONTROL_TAKEOFFHOOK = 6, /*!< Make it go off hook */ 00306 AST_CONTROL_OFFHOOK = 7, /*!< Line is off hook */ 00307 AST_CONTROL_CONGESTION = 8, /*!< Congestion (circuits busy) */ 00308 AST_CONTROL_FLASH = 9, /*!< Flash hook */ 00309 AST_CONTROL_WINK = 10, /*!< Wink */ 00310 AST_CONTROL_OPTION = 11, /*!< Set a low-level option */ 00311 AST_CONTROL_RADIO_KEY = 12, /*!< Key Radio */ 00312 AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */ 00313 AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */ 00314 AST_CONTROL_PROCEEDING = 15, /*!< Indicate CALL PROCEEDING */ 00315 AST_CONTROL_HOLD = 16, /*!< Indicate call is placed on hold */ 00316 AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold */ 00317 AST_CONTROL_VIDUPDATE = 18, /*!< Indicate video frame update */ 00318 AST_CONTROL_T38 = 19, /*!< T38 state change request/notification */ 00319 AST_CONTROL_SRCUPDATE = 20, /*!< Indicate source of media has changed */ 00320 };
enum ast_control_t38 |
Definition at line 322 of file frame.h.
00322 { 00323 AST_T38_REQUEST_NEGOTIATE = 1, /*!< Request T38 on a channel (voice to fax) */ 00324 AST_T38_REQUEST_TERMINATE, /*!< Terminate T38 on a channel (fax to voice) */ 00325 AST_T38_NEGOTIATED, /*!< T38 negotiated (fax mode) */ 00326 AST_T38_TERMINATED, /*!< T38 terminated (back to voice) */ 00327 AST_T38_REFUSED /*!< T38 refused for some reason (usually rejected by remote end) */ 00328 };
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 204 of file frame.c.
References AST_FRAME_VOICE, ast_log(), AST_MIN_OFFSET, AST_SMOOTHER_FLAG_G729, ast_swapcopy_samples(), f, LOG_WARNING, s, smoother_frame_feed(), and SMOOTHER_SIZE.
00205 { 00206 if (f->frametype != AST_FRAME_VOICE) { 00207 ast_log(LOG_WARNING, "Huh? Can't smooth a non-voice frame!\n"); 00208 return -1; 00209 } 00210 if (!s->format) { 00211 s->format = f->subclass; 00212 s->samplesperbyte = (float)f->samples / (float)f->datalen; 00213 } else if (s->format != f->subclass) { 00214 ast_log(LOG_WARNING, "Smoother was working on %d format frames, now trying to feed %d?\n", s->format, f->subclass); 00215 return -1; 00216 } 00217 if (s->len + f->datalen > SMOOTHER_SIZE) { 00218 ast_log(LOG_WARNING, "Out of smoother space\n"); 00219 return -1; 00220 } 00221 if (((f->datalen == s->size) || 00222 ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) && 00223 !s->opt && 00224 !s->len && 00225 (f->offset >= AST_MIN_OFFSET)) { 00226 /* Optimize by sending the frame we just got 00227 on the next read, thus eliminating the douple 00228 copy */ 00229 if (swap) 00230 ast_swapcopy_samples(f->data.ptr, f->data.ptr, f->samples); 00231 s->opt = f; 00232 s->opt_needs_swap = swap ? 1 : 0; 00233 return 0; 00234 } 00235 00236 return smoother_frame_feed(s, f, swap); 00237 }
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 631 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, and ast_format_list::desc.
Referenced by moh_alloc(), show_codec_n(), and show_codecs().
00632 { 00633 int x; 00634 char *ret = "unknown"; 00635 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00636 if (AST_FORMAT_LIST[x].bits == codec) { 00637 ret = AST_FORMAT_LIST[x].desc; 00638 break; 00639 } 00640 } 00641 return ret; 00642 }
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 1221 of file frame.c.
References ARRAY_LEN, ast_best_codec(), ast_debug, AST_FORMAT_AUDIO_MASK, AST_FORMAT_LIST, ast_format_list::bits, and ast_codec_pref::order.
Referenced by __oh323_new(), gtalk_new(), jingle_new(), process_sdp(), sip_new(), and socket_process().
01222 { 01223 int x, ret = 0, slot; 01224 01225 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01226 slot = pref->order[x]; 01227 01228 if (!slot) 01229 break; 01230 if (formats & AST_FORMAT_LIST[slot-1].bits) { 01231 ret = AST_FORMAT_LIST[slot-1].bits; 01232 break; 01233 } 01234 } 01235 if (ret & AST_FORMAT_AUDIO_MASK) 01236 return ret; 01237 01238 ast_debug(4, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec"); 01239 01240 return find_best ? ast_best_codec(formats) : 0; 01241 }
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 1485 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_SLINEAR, AST_FORMAT_SLINEAR16, AST_FORMAT_ULAW, ast_getformatname(), ast_log(), len(), and LOG_WARNING.
Referenced by moh_generate(), and monmp3thread().
01486 { 01487 int len = 0; 01488 01489 /* XXX Still need speex, g723, and lpc10 XXX */ 01490 switch(format) { 01491 case AST_FORMAT_G723_1: 01492 len = (samples / 240) * 20; 01493 break; 01494 case AST_FORMAT_ILBC: 01495 len = (samples / 240) * 50; 01496 break; 01497 case AST_FORMAT_GSM: 01498 len = (samples / 160) * 33; 01499 break; 01500 case AST_FORMAT_G729A: 01501 len = samples / 8; 01502 break; 01503 case AST_FORMAT_SLINEAR: 01504 case AST_FORMAT_SLINEAR16: 01505 len = samples * 2; 01506 break; 01507 case AST_FORMAT_ULAW: 01508 case AST_FORMAT_ALAW: 01509 len = samples; 01510 break; 01511 case AST_FORMAT_G722: 01512 case AST_FORMAT_ADPCM: 01513 case AST_FORMAT_G726: 01514 case AST_FORMAT_G726_AAL2: 01515 len = samples / 2; 01516 break; 01517 default: 01518 ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format)); 01519 } 01520 01521 return len; 01522 }
int ast_codec_get_samples | ( | struct ast_frame * | f | ) |
Returns the number of samples contained in the frame.
Definition at line 1441 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_SLINEAR16, 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(), socket_process(), and socket_process_meta().
01442 { 01443 int samples=0; 01444 switch(f->subclass) { 01445 case AST_FORMAT_SPEEX: 01446 samples = speex_samples(f->data.ptr, f->datalen); 01447 break; 01448 case AST_FORMAT_G723_1: 01449 samples = g723_samples(f->data.ptr, f->datalen); 01450 break; 01451 case AST_FORMAT_ILBC: 01452 samples = 240 * (f->datalen / 50); 01453 break; 01454 case AST_FORMAT_GSM: 01455 samples = 160 * (f->datalen / 33); 01456 break; 01457 case AST_FORMAT_G729A: 01458 samples = f->datalen * 8; 01459 break; 01460 case AST_FORMAT_SLINEAR: 01461 case AST_FORMAT_SLINEAR16: 01462 samples = f->datalen / 2; 01463 break; 01464 case AST_FORMAT_LPC10: 01465 /* assumes that the RTP packet contains one LPC10 frame */ 01466 samples = 22 * 8; 01467 samples += (((char *)(f->data.ptr))[7] & 0x1) * 8; 01468 break; 01469 case AST_FORMAT_ULAW: 01470 case AST_FORMAT_ALAW: 01471 samples = f->datalen; 01472 break; 01473 case AST_FORMAT_G722: 01474 case AST_FORMAT_ADPCM: 01475 case AST_FORMAT_G726: 01476 case AST_FORMAT_G726_AAL2: 01477 samples = f->datalen * 2; 01478 break; 01479 default: 01480 ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass)); 01481 } 01482 return samples; 01483 }
static int ast_codec_interp_len | ( | int | format | ) | [inline, static] |
Gets duration in ms of interpolation frame for a format.
Definition at line 624 of file frame.h.
References AST_FORMAT_ILBC.
Referenced by __get_from_jb(), and jb_get_and_deliver().
00625 { 00626 return (format == AST_FORMAT_ILBC) ? 30 : 20; 00627 }
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 1081 of file frame.c.
References ARRAY_LEN, ast_codec_pref_remove(), AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow().
01082 { 01083 int x, newindex = 0; 01084 01085 ast_codec_pref_remove(pref, format); 01086 01087 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01088 if (AST_FORMAT_LIST[x].bits == format) { 01089 newindex = x + 1; 01090 break; 01091 } 01092 } 01093 01094 if (newindex) { 01095 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01096 if (!pref->order[x]) { 01097 pref->order[x] = newindex; 01098 break; 01099 } 01100 } 01101 } 01102 01103 return x; 01104 }
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 984 of file frame.c.
References ast_codec_pref::order.
Referenced by check_access(), create_addr(), dump_prefs(), and socket_process().
00985 { 00986 int x, differential = (int) 'A', mem; 00987 char *from, *to; 00988 00989 if (right) { 00990 from = pref->order; 00991 to = buf; 00992 mem = size; 00993 } else { 00994 to = pref->order; 00995 from = buf; 00996 mem = 32; 00997 } 00998 00999 memset(to, 0, mem); 01000 for (x = 0; x < 32 ; x++) { 01001 if (!from[x]) 01002 break; 01003 to[x] = right ? (from[x] + differential) : (from[x] - differential); 01004 } 01005 }
struct ast_format_list ast_codec_pref_getsize | ( | struct ast_codec_pref * | pref, | |
int | format | |||
) |
Get packet size for codec.
Definition at line 1182 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_format_list::bits, ast_format_list::cur_ms, ast_format_list::def_ms, format, ast_format_list::inc_ms, ast_format_list::max_ms, and ast_format_list::min_ms.
Referenced by add_codec_to_sdp(), ast_rtp_bridge(), ast_rtp_codec_setpref(), ast_rtp_write(), handle_open_receive_channel_ack_message(), skinny_set_rtp_peer(), and transmit_connect().
01183 { 01184 int x, idx = -1, framems = 0; 01185 struct ast_format_list fmt = { 0, }; 01186 01187 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01188 if (AST_FORMAT_LIST[x].bits == format) { 01189 fmt = AST_FORMAT_LIST[x]; 01190 idx = x; 01191 break; 01192 } 01193 } 01194 01195 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01196 if (pref->order[x] == (idx + 1)) { 01197 framems = pref->framing[x]; 01198 break; 01199 } 01200 } 01201 01202 /* size validation */ 01203 if (!framems) 01204 framems = AST_FORMAT_LIST[idx].def_ms; 01205 01206 if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */ 01207 framems -= framems % AST_FORMAT_LIST[idx].inc_ms; 01208 01209 if (framems < AST_FORMAT_LIST[idx].min_ms) 01210 framems = AST_FORMAT_LIST[idx].min_ms; 01211 01212 if (framems > AST_FORMAT_LIST[idx].max_ms) 01213 framems = AST_FORMAT_LIST[idx].max_ms; 01214 01215 fmt.cur_ms = framems; 01216 01217 return fmt; 01218 }
int ast_codec_pref_index | ( | struct ast_codec_pref * | pref, | |
int | index | |||
) |
Codec located at a particular place in the preference index.
Definition at line 1042 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(), handle_cli_iax2_show_peer(), jingle_accept_call(), print_codec_to_cli(), and socket_process().
01043 { 01044 int slot = 0; 01045 01046 if ((idx >= 0) && (idx < sizeof(pref->order))) { 01047 slot = pref->order[idx]; 01048 } 01049 01050 return slot ? AST_FORMAT_LIST[slot - 1].bits : 0; 01051 }
void ast_codec_pref_init | ( | struct ast_codec_pref * | pref | ) |
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 1107 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_codec_pref::framing, and ast_codec_pref::order.
Referenced by create_addr().
01108 { 01109 int x, newindex = 0; 01110 01111 /* First step is to get the codecs "index number" */ 01112 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01113 if (AST_FORMAT_LIST[x].bits == format) { 01114 newindex = x + 1; 01115 break; 01116 } 01117 } 01118 /* Done if its unknown */ 01119 if (!newindex) 01120 return; 01121 01122 /* Now find any existing occurrence, or the end */ 01123 for (x = 0; x < 32; x++) { 01124 if (!pref->order[x] || pref->order[x] == newindex) 01125 break; 01126 } 01127 01128 if (only_if_existing && !pref->order[x]) 01129 return; 01130 01131 /* Move down to make space to insert - either all the way to the end, 01132 or as far as the existing location (which will be overwritten) */ 01133 for (; x > 0; x--) { 01134 pref->order[x] = pref->order[x - 1]; 01135 pref->framing[x] = pref->framing[x - 1]; 01136 } 01137 01138 /* And insert the new entry */ 01139 pref->order[0] = newindex; 01140 pref->framing[0] = 0; /* ? */ 01141 }
void ast_codec_pref_remove | ( | struct ast_codec_pref * | pref, | |
int | format | |||
) |
Remove audio a codec from a preference list.
Definition at line 1054 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_codec_pref_append(), and ast_parse_allow_disallow().
01055 { 01056 struct ast_codec_pref oldorder; 01057 int x, y = 0; 01058 int slot; 01059 int size; 01060 01061 if (!pref->order[0]) 01062 return; 01063 01064 memcpy(&oldorder, pref, sizeof(oldorder)); 01065 memset(pref, 0, sizeof(*pref)); 01066 01067 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01068 slot = oldorder.order[x]; 01069 size = oldorder.framing[x]; 01070 if (! slot) 01071 break; 01072 if (AST_FORMAT_LIST[slot-1].bits != format) { 01073 pref->order[y] = slot; 01074 pref->framing[y++] = size; 01075 } 01076 } 01077 01078 }
int ast_codec_pref_setsize | ( | struct ast_codec_pref * | pref, | |
int | format, | |||
int | framems | |||
) |
Set packet size for codec.
Definition at line 1144 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_format_list::def_ms, ast_codec_pref::framing, ast_format_list::inc_ms, ast_format_list::max_ms, ast_format_list::min_ms, and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow(), and process_sdp().
01145 { 01146 int x, idx = -1; 01147 01148 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01149 if (AST_FORMAT_LIST[x].bits == format) { 01150 idx = x; 01151 break; 01152 } 01153 } 01154 01155 if (idx < 0) 01156 return -1; 01157 01158 /* size validation */ 01159 if (!framems) 01160 framems = AST_FORMAT_LIST[idx].def_ms; 01161 01162 if (AST_FORMAT_LIST[idx].inc_ms && framems % AST_FORMAT_LIST[idx].inc_ms) /* avoid division by zero */ 01163 framems -= framems % AST_FORMAT_LIST[idx].inc_ms; 01164 01165 if (framems < AST_FORMAT_LIST[idx].min_ms) 01166 framems = AST_FORMAT_LIST[idx].min_ms; 01167 01168 if (framems > AST_FORMAT_LIST[idx].max_ms) 01169 framems = AST_FORMAT_LIST[idx].max_ms; 01170 01171 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01172 if (pref->order[x] == (idx + 1)) { 01173 pref->framing[x] = framems; 01174 break; 01175 } 01176 } 01177 01178 return x; 01179 }
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 1007 of file frame.c.
References ast_codec_pref_index(), and ast_getformatname().
Referenced by dump_prefs(), and socket_process().
01008 { 01009 int x, codec; 01010 size_t total_len, slen; 01011 char *formatname; 01012 01013 memset(buf,0,size); 01014 total_len = size; 01015 buf[0] = '('; 01016 total_len--; 01017 for(x = 0; x < 32 ; x++) { 01018 if (total_len <= 0) 01019 break; 01020 if (!(codec = ast_codec_pref_index(pref,x))) 01021 break; 01022 if ((formatname = ast_getformatname(codec))) { 01023 slen = strlen(formatname); 01024 if (slen > total_len) 01025 break; 01026 strncat(buf, formatname, total_len - 1); /* safe */ 01027 total_len -= slen; 01028 } 01029 if (total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) { 01030 strncat(buf, "|", total_len - 1); /* safe */ 01031 total_len--; 01032 } 01033 } 01034 if (total_len) { 01035 strncat(buf, ")", total_len - 1); /* safe */ 01036 total_len--; 01037 } 01038 01039 return size - total_len; 01040 }
static force_inline int ast_format_rate | ( | int | format | ) | [static] |
Get the sample rate for a given format.
Definition at line 651 of file frame.h.
References AST_FORMAT_G722, and AST_FORMAT_SLINEAR16.
Referenced by ast_read_generator_actions(), ast_readaudio_callback(), ast_readvideo_callback(), ast_rtp_read(), ast_smoother_read(), ast_translate(), calc_cost(), and generator_force().
00652 { 00653 if (format == AST_FORMAT_G722 || format == AST_FORMAT_SLINEAR16) 00654 return 16000; 00655 00656 return 8000; 00657 }
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 1524 of file frame.c.
References AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_slinear_saturated_divide(), ast_slinear_saturated_multiply(), and f.
Referenced by audiohook_read_frame_single(), audiohook_volume_callback(), conf_run(), and volume_callback().
01525 { 01526 int count; 01527 short *fdata = f->data.ptr; 01528 short adjust_value = abs(adjustment); 01529 01530 if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR)) 01531 return -1; 01532 01533 if (!adjustment) 01534 return 0; 01535 01536 for (count = 0; count < f->samples; count++) { 01537 if (adjustment > 0) { 01538 ast_slinear_saturated_multiply(&fdata[count], &adjust_value); 01539 } else if (adjustment < 0) { 01540 ast_slinear_saturated_divide(&fdata[count], &adjust_value); 01541 } 01542 } 01543 01544 return 0; 01545 }
void ast_frame_dump | ( | const char * | name, | |
struct ast_frame * | f, | |||
char * | prefix | |||
) |
Dump a frame for debugging purposes
Definition at line 733 of file frame.c.
References AST_CONTROL_ANSWER, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_FLASH, AST_CONTROL_HANGUP, AST_CONTROL_HOLD, AST_CONTROL_OFFHOOK, AST_CONTROL_OPTION, AST_CONTROL_RADIO_KEY, AST_CONTROL_RADIO_UNKEY, AST_CONTROL_RING, AST_CONTROL_RINGING, AST_CONTROL_T38, AST_CONTROL_TAKEOFFHOOK, AST_CONTROL_UNHOLD, AST_CONTROL_WINK, ast_copy_string(), AST_FRAME_CONTROL, AST_FRAME_DTMF_BEGIN, AST_FRAME_DTMF_END, AST_FRAME_HTML, AST_FRAME_IAX, AST_FRAME_IMAGE, AST_FRAME_MODEM, AST_FRAME_NULL, AST_FRAME_TEXT, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_getformatname(), AST_HTML_BEGIN, AST_HTML_DATA, AST_HTML_END, AST_HTML_LDCOMPLETE, AST_HTML_LINKREJECT, AST_HTML_LINKURL, AST_HTML_NOSUPPORT, AST_HTML_UNLINK, AST_HTML_URL, AST_MODEM_T38, AST_MODEM_V150, ast_strlen_zero(), AST_T38_NEGOTIATED, AST_T38_REFUSED, AST_T38_REQUEST_NEGOTIATE, AST_T38_REQUEST_TERMINATE, AST_T38_TERMINATED, ast_verbose, COLOR_BLACK, COLOR_BRCYAN, COLOR_BRGREEN, COLOR_BRMAGENTA, COLOR_BRRED, COLOR_YELLOW, f, and term_color().
Referenced by __ast_read(), and ast_write().
00734 { 00735 const char noname[] = "unknown"; 00736 char ftype[40] = "Unknown Frametype"; 00737 char cft[80]; 00738 char subclass[40] = "Unknown Subclass"; 00739 char csub[80]; 00740 char moreinfo[40] = ""; 00741 char cn[60]; 00742 char cp[40]; 00743 char cmn[40]; 00744 const char *message = "Unknown"; 00745 00746 if (!name) 00747 name = noname; 00748 00749 00750 if (!f) { 00751 ast_verbose("%s [ %s (NULL) ] [%s]\n", 00752 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00753 term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00754 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00755 return; 00756 } 00757 /* XXX We should probably print one each of voice and video when the format changes XXX */ 00758 if (f->frametype == AST_FRAME_VOICE) 00759 return; 00760 if (f->frametype == AST_FRAME_VIDEO) 00761 return; 00762 switch(f->frametype) { 00763 case AST_FRAME_DTMF_BEGIN: 00764 strcpy(ftype, "DTMF Begin"); 00765 subclass[0] = f->subclass; 00766 subclass[1] = '\0'; 00767 break; 00768 case AST_FRAME_DTMF_END: 00769 strcpy(ftype, "DTMF End"); 00770 subclass[0] = f->subclass; 00771 subclass[1] = '\0'; 00772 break; 00773 case AST_FRAME_CONTROL: 00774 strcpy(ftype, "Control"); 00775 switch(f->subclass) { 00776 case AST_CONTROL_HANGUP: 00777 strcpy(subclass, "Hangup"); 00778 break; 00779 case AST_CONTROL_RING: 00780 strcpy(subclass, "Ring"); 00781 break; 00782 case AST_CONTROL_RINGING: 00783 strcpy(subclass, "Ringing"); 00784 break; 00785 case AST_CONTROL_ANSWER: 00786 strcpy(subclass, "Answer"); 00787 break; 00788 case AST_CONTROL_BUSY: 00789 strcpy(subclass, "Busy"); 00790 break; 00791 case AST_CONTROL_TAKEOFFHOOK: 00792 strcpy(subclass, "Take Off Hook"); 00793 break; 00794 case AST_CONTROL_OFFHOOK: 00795 strcpy(subclass, "Line Off Hook"); 00796 break; 00797 case AST_CONTROL_CONGESTION: 00798 strcpy(subclass, "Congestion"); 00799 break; 00800 case AST_CONTROL_FLASH: 00801 strcpy(subclass, "Flash"); 00802 break; 00803 case AST_CONTROL_WINK: 00804 strcpy(subclass, "Wink"); 00805 break; 00806 case AST_CONTROL_OPTION: 00807 strcpy(subclass, "Option"); 00808 break; 00809 case AST_CONTROL_RADIO_KEY: 00810 strcpy(subclass, "Key Radio"); 00811 break; 00812 case AST_CONTROL_RADIO_UNKEY: 00813 strcpy(subclass, "Unkey Radio"); 00814 break; 00815 case AST_CONTROL_HOLD: 00816 strcpy(subclass, "Hold"); 00817 break; 00818 case AST_CONTROL_UNHOLD: 00819 strcpy(subclass, "Unhold"); 00820 break; 00821 case AST_CONTROL_T38: 00822 if (f->datalen != sizeof(enum ast_control_t38)) { 00823 message = "Invalid"; 00824 } else { 00825 enum ast_control_t38 state = *((enum ast_control_t38 *) f->data.ptr); 00826 if (state == AST_T38_REQUEST_NEGOTIATE) 00827 message = "Negotiation Requested"; 00828 else if (state == AST_T38_REQUEST_TERMINATE) 00829 message = "Negotiation Request Terminated"; 00830 else if (state == AST_T38_NEGOTIATED) 00831 message = "Negotiated"; 00832 else if (state == AST_T38_TERMINATED) 00833 message = "Terminated"; 00834 else if (state == AST_T38_REFUSED) 00835 message = "Refused"; 00836 } 00837 snprintf(subclass, sizeof(subclass), "T38/%s", message); 00838 break; 00839 case -1: 00840 strcpy(subclass, "Stop generators"); 00841 break; 00842 default: 00843 snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass); 00844 } 00845 break; 00846 case AST_FRAME_NULL: 00847 strcpy(ftype, "Null Frame"); 00848 strcpy(subclass, "N/A"); 00849 break; 00850 case AST_FRAME_IAX: 00851 /* Should never happen */ 00852 strcpy(ftype, "IAX Specific"); 00853 snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass); 00854 break; 00855 case AST_FRAME_TEXT: 00856 strcpy(ftype, "Text"); 00857 strcpy(subclass, "N/A"); 00858 ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); 00859 break; 00860 case AST_FRAME_IMAGE: 00861 strcpy(ftype, "Image"); 00862 snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass)); 00863 break; 00864 case AST_FRAME_HTML: 00865 strcpy(ftype, "HTML"); 00866 switch(f->subclass) { 00867 case AST_HTML_URL: 00868 strcpy(subclass, "URL"); 00869 ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); 00870 break; 00871 case AST_HTML_DATA: 00872 strcpy(subclass, "Data"); 00873 break; 00874 case AST_HTML_BEGIN: 00875 strcpy(subclass, "Begin"); 00876 break; 00877 case AST_HTML_END: 00878 strcpy(subclass, "End"); 00879 break; 00880 case AST_HTML_LDCOMPLETE: 00881 strcpy(subclass, "Load Complete"); 00882 break; 00883 case AST_HTML_NOSUPPORT: 00884 strcpy(subclass, "No Support"); 00885 break; 00886 case AST_HTML_LINKURL: 00887 strcpy(subclass, "Link URL"); 00888 ast_copy_string(moreinfo, f->data.ptr, sizeof(moreinfo)); 00889 break; 00890 case AST_HTML_UNLINK: 00891 strcpy(subclass, "Unlink"); 00892 break; 00893 case AST_HTML_LINKREJECT: 00894 strcpy(subclass, "Link Reject"); 00895 break; 00896 default: 00897 snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass); 00898 break; 00899 } 00900 break; 00901 case AST_FRAME_MODEM: 00902 strcpy(ftype, "Modem"); 00903 switch (f->subclass) { 00904 case AST_MODEM_T38: 00905 strcpy(subclass, "T.38"); 00906 break; 00907 case AST_MODEM_V150: 00908 strcpy(subclass, "V.150"); 00909 break; 00910 default: 00911 snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass); 00912 break; 00913 } 00914 break; 00915 default: 00916 snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype); 00917 } 00918 if (!ast_strlen_zero(moreinfo)) 00919 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n", 00920 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00921 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00922 f->frametype, 00923 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00924 f->subclass, 00925 term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)), 00926 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00927 else 00928 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n", 00929 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00930 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00931 f->frametype, 00932 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00933 f->subclass, 00934 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00935 }
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 342 of file frame.c.
References ast_dsp_frame_freed(), ast_filestream_frame_freed(), ast_free, 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, ast_frame::mallocd, ast_frame::offset, ast_frame::ptr, and ast_frame::src.
Referenced by mixmonitor_thread().
00343 { 00344 if (ast_test_flag(fr, AST_FRFLAG_FROM_TRANSLATOR)) { 00345 ast_translate_frame_freed(fr); 00346 } else if (ast_test_flag(fr, AST_FRFLAG_FROM_DSP)) { 00347 ast_dsp_frame_freed(fr); 00348 } else if (ast_test_flag(fr, AST_FRFLAG_FROM_FILESTREAM)) { 00349 ast_filestream_frame_freed(fr); 00350 } 00351 00352 if (!fr->mallocd) 00353 return; 00354 00355 #if !defined(LOW_MEMORY) 00356 if (cache && fr->mallocd == AST_MALLOCD_HDR) { 00357 /* Cool, only the header is malloc'd, let's just cache those for now 00358 * to keep things simple... */ 00359 struct ast_frame_cache *frames; 00360 00361 if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames))) 00362 && frames->size < FRAME_CACHE_MAX_SIZE) { 00363 AST_LIST_INSERT_HEAD(&frames->list, fr, frame_list); 00364 frames->size++; 00365 return; 00366 } 00367 } 00368 #endif 00369 00370 if (fr->mallocd & AST_MALLOCD_DATA) { 00371 if (fr->data.ptr) 00372 ast_free(fr->data.ptr - fr->offset); 00373 } 00374 if (fr->mallocd & AST_MALLOCD_SRC) { 00375 if (fr->src) 00376 ast_free((char *)fr->src); 00377 } 00378 if (fr->mallocd & AST_MALLOCD_HDR) { 00379 #ifdef TRACE_FRAMES 00380 AST_LIST_LOCK(&headerlist); 00381 headers--; 00382 AST_LIST_REMOVE(&headerlist, fr, frame_list); 00383 AST_LIST_UNLOCK(&headerlist); 00384 #endif 00385 ast_free(fr); 00386 } 00387 }
Sums two frames of audio samples.
f1 | The first frame (which will contain the result) | |
f2 | The second frame |
Definition at line 1547 of file frame.c.
References AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_slinear_saturated_add(), ast_frame::data, ast_frame::frametype, ast_frame::ptr, ast_frame::samples, and ast_frame::subclass.
01548 { 01549 int count; 01550 short *data1, *data2; 01551 01552 if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR)) 01553 return -1; 01554 01555 if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR)) 01556 return -1; 01557 01558 if (f1->samples != f2->samples) 01559 return -1; 01560 01561 for (count = 0, data1 = f1->data.ptr, data2 = f2->data.ptr; 01562 count < f1->samples; 01563 count++, data1++, data2++) 01564 ast_slinear_saturated_add(data1, data2); 01565 01566 return 0; 01567 }
Copies a frame.
fr | frame to copy Duplicates a frame -- should only rarely be used, typically frisolate is good enough |
Definition at line 453 of file frame.c.
References ast_calloc_cache, ast_copy_flags, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, AST_LIST_REMOVE_CURRENT, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, AST_MALLOCD_HDR, ast_threadstorage_get(), buf, ast_frame::data, ast_frame::datalen, ast_frame::delivery, f, frame_cache, frames, ast_frame::frametype, ast_frame::len, len(), ast_frame::mallocd, ast_frame::mallocd_hdr_len, ast_frame::offset, ast_frame::ptr, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, ast_frame::ts, and ast_frame::uint32.
Referenced by __ast_queue_frame(), ast_jb_put(), ast_rtp_write(), ast_slinfactory_feed(), audiohook_read_frame_single(), autoservice_run(), recordthread(), rpt(), and transmit_audio().
00454 { 00455 struct ast_frame *out = NULL; 00456 int len, srclen = 0; 00457 void *buf = NULL; 00458 00459 #if !defined(LOW_MEMORY) 00460 struct ast_frame_cache *frames; 00461 #endif 00462 00463 /* Start with standard stuff */ 00464 len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00465 /* If we have a source, add space for it */ 00466 /* 00467 * XXX Watch out here - if we receive a src which is not terminated 00468 * properly, we can be easily attacked. Should limit the size we deal with. 00469 */ 00470 if (f->src) 00471 srclen = strlen(f->src); 00472 if (srclen > 0) 00473 len += srclen + 1; 00474 00475 #if !defined(LOW_MEMORY) 00476 if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) { 00477 AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) { 00478 if (out->mallocd_hdr_len >= len) { 00479 size_t mallocd_len = out->mallocd_hdr_len; 00480 00481 AST_LIST_REMOVE_CURRENT(frame_list); 00482 memset(out, 0, sizeof(*out)); 00483 out->mallocd_hdr_len = mallocd_len; 00484 buf = out; 00485 frames->size--; 00486 break; 00487 } 00488 } 00489 AST_LIST_TRAVERSE_SAFE_END; 00490 } 00491 #endif 00492 00493 if (!buf) { 00494 if (!(buf = ast_calloc_cache(1, len))) 00495 return NULL; 00496 out = buf; 00497 out->mallocd_hdr_len = len; 00498 } 00499 00500 out->frametype = f->frametype; 00501 out->subclass = f->subclass; 00502 out->datalen = f->datalen; 00503 out->samples = f->samples; 00504 out->delivery = f->delivery; 00505 /* Set us as having malloc'd header only, so it will eventually 00506 get freed. */ 00507 out->mallocd = AST_MALLOCD_HDR; 00508 out->offset = AST_FRIENDLY_OFFSET; 00509 if (out->datalen) { 00510 out->data.ptr = buf + sizeof(*out) + AST_FRIENDLY_OFFSET; 00511 memcpy(out->data.ptr, f->data.ptr, out->datalen); 00512 } else { 00513 out->data.uint32 = f->data.uint32; 00514 } 00515 if (srclen > 0) { 00516 /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */ 00517 char *src; 00518 out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00519 src = (char *) out->src; 00520 /* Must have space since we allocated for it */ 00521 strcpy(src, f->src); 00522 } 00523 ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO); 00524 out->ts = f->ts; 00525 out->len = f->len; 00526 out->seqno = f->seqno; 00527 return out; 00528 }
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 394 of file frame.c.
References ast_clear_flag, ast_copy_flags, ast_frame_header_new(), ast_free, AST_FRFLAG_FROM_DSP, AST_FRFLAG_FROM_TRANSLATOR, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, ast_malloc, AST_MALLOCD_DATA, AST_MALLOCD_HDR, AST_MALLOCD_SRC, ast_strdup, ast_test_flag, ast_frame::data, ast_frame::datalen, ast_frame::frametype, ast_frame::len, ast_frame::mallocd, ast_frame::offset, ast_frame::ptr, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, and ast_frame::ts.
Referenced by __ast_answer(), and jpeg_read_image().
00395 { 00396 struct ast_frame *out; 00397 void *newdata; 00398 00399 ast_clear_flag(fr, AST_FRFLAG_FROM_TRANSLATOR); 00400 ast_clear_flag(fr, AST_FRFLAG_FROM_DSP); 00401 00402 if (!(fr->mallocd & AST_MALLOCD_HDR)) { 00403 /* Allocate a new header if needed */ 00404 if (!(out = ast_frame_header_new())) 00405 return NULL; 00406 out->frametype = fr->frametype; 00407 out->subclass = fr->subclass; 00408 out->datalen = fr->datalen; 00409 out->samples = fr->samples; 00410 out->offset = fr->offset; 00411 out->data = fr->data; 00412 /* Copy the timing data */ 00413 ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO); 00414 if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) { 00415 out->ts = fr->ts; 00416 out->len = fr->len; 00417 out->seqno = fr->seqno; 00418 } 00419 } else 00420 out = fr; 00421 00422 if (!(fr->mallocd & AST_MALLOCD_SRC)) { 00423 if (fr->src) { 00424 if (!(out->src = ast_strdup(fr->src))) { 00425 if (out != fr) 00426 ast_free(out); 00427 return NULL; 00428 } 00429 } 00430 } else 00431 out->src = fr->src; 00432 00433 if (!(fr->mallocd & AST_MALLOCD_DATA)) { 00434 if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) { 00435 if (out->src != fr->src) 00436 ast_free((void *) out->src); 00437 if (out != fr) 00438 ast_free(out); 00439 return NULL; 00440 } 00441 newdata += AST_FRIENDLY_OFFSET; 00442 out->offset = AST_FRIENDLY_OFFSET; 00443 out->datalen = fr->datalen; 00444 memcpy(newdata, fr->data.ptr, fr->datalen); 00445 out->data.ptr = newdata; 00446 } 00447 00448 out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA; 00449 00450 return out; 00451 }
struct ast_format_list* ast_get_format_list | ( | size_t * | size | ) |
Definition at line 546 of file frame.c.
References ARRAY_LEN, and AST_FORMAT_LIST.
00547 { 00548 *size = ARRAY_LEN(AST_FORMAT_LIST); 00549 return AST_FORMAT_LIST; 00550 }
struct ast_format_list* ast_get_format_list_index | ( | int | index | ) |
Definition at line 541 of file frame.c.
References AST_FORMAT_LIST.
00542 { 00543 return &AST_FORMAT_LIST[idx]; 00544 }
int ast_getformatbyname | ( | const char * | name | ) |
Gets a format from a name.
name | string of format |
Definition at line 613 of file frame.c.
References ARRAY_LEN, ast_expand_codec_alias(), AST_FORMAT_LIST, ast_format_list::bits, and format.
Referenced by ast_parse_allow_disallow(), iax_template_parse(), load_moh_classes(), local_ast_moh_start(), reload_config(), and try_suggested_sip_codec().
00614 { 00615 int x, all, format = 0; 00616 00617 all = strcasecmp(name, "all") ? 0 : 1; 00618 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00619 if (all || 00620 !strcasecmp(AST_FORMAT_LIST[x].name,name) || 00621 !strcasecmp(AST_FORMAT_LIST[x].name, ast_expand_codec_alias(name))) { 00622 format |= AST_FORMAT_LIST[x].bits; 00623 if (!all) 00624 break; 00625 } 00626 } 00627 00628 return format; 00629 }
char* ast_getformatname | ( | int | format | ) |
Get the name of a format.
format | id of format |
Definition at line 552 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_format_list::bits, and ast_format_list::name.
Referenced by __ast_play_and_record(), __ast_read(), __ast_register_translator(), _sip_show_peer(), add_codec_to_answer(), add_codec_to_sdp(), add_tcodec_to_sdp(), add_vcodec_to_sdp(), agent_call(), ast_codec_get_len(), ast_codec_get_samples(), ast_codec_pref_string(), ast_dsp_process(), ast_frame_dump(), ast_openvstream(), ast_rtp_write(), ast_slinfactory_feed(), ast_streamfile(), ast_translator_build_path(), ast_unregister_translator(), ast_writestream(), background_detect_exec(), dahdi_read(), do_waiting(), eagi_exec(), func_channel_read(), function_iaxpeer(), function_sippeer(), gtalk_show_channels(), handle_cli_core_show_file_formats(), handle_cli_core_show_translation(), handle_cli_iax2_show_channels(), handle_cli_iax2_show_peer(), handle_cli_moh_show_classes(), handle_core_show_image_formats(), iax2_request(), iax_show_provisioning(), jingle_show_channels(), login_exec(), moh_release(), oh323_rtp_read(), phone_setup(), print_codec_to_cli(), rebuild_matrix(), register_translator(), set_format(), set_peer_capabilities(), show_codecs(), sip_request_call(), sip_rtp_read(), socket_process(), start_rtp(), unistim_request(), unistim_rtp_read(), and unistim_write().
00553 { 00554 int x; 00555 char *ret = "unknown"; 00556 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00557 if (AST_FORMAT_LIST[x].bits == format) { 00558 ret = AST_FORMAT_LIST[x].name; 00559 break; 00560 } 00561 } 00562 return ret; 00563 }
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 565 of file frame.c.
References ARRAY_LEN, ast_copy_string(), AST_FORMAT_LIST, ast_format_list::bits, len(), and name.
Referenced by __ast_read(), _sip_show_peer(), add_sdp(), ast_streamfile(), function_iaxpeer(), function_sippeer(), gtalk_is_answered(), gtalk_newcall(), handle_cli_iax2_show_peer(), handle_showchan(), handle_skinny_show_line(), process_sdp(), serialize_showchan(), set_format(), show_channels_cb(), sip_new(), sip_request_call(), sip_show_channel(), sip_show_settings(), and sip_write().
00566 { 00567 int x; 00568 unsigned len; 00569 char *start, *end = buf; 00570 00571 if (!size) 00572 return buf; 00573 snprintf(end, size, "0x%x (", format); 00574 len = strlen(end); 00575 end += len; 00576 size -= len; 00577 start = end; 00578 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00579 if (AST_FORMAT_LIST[x].bits & format) { 00580 snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name); 00581 len = strlen(end); 00582 end += len; 00583 size -= len; 00584 } 00585 } 00586 if (start == end) 00587 ast_copy_string(start, "nothing)", size); 00588 else if (size > 1) 00589 *(end -1) = ')'; 00590 return buf; 00591 }
int ast_parse_allow_disallow | ( | struct ast_codec_pref * | pref, | |
int * | mask, | |||
const char * | list, | |||
int | allowing | |||
) |
Parse an "allow" or "deny" line in a channel or device configuration and update the capabilities mask and pref if provided. Video codecs are not added to codec preference lists, since we can not transcode.
Definition at line 1243 of file frame.c.
References ast_codec_pref_append(), ast_codec_pref_remove(), ast_codec_pref_setsize(), ast_debug, AST_FORMAT_AUDIO_MASK, ast_getformatbyname(), ast_log(), ast_strdupa, format, LOG_WARNING, parse(), and strsep().
Referenced by action_originate(), apply_outgoing(), build_device(), build_peer(), build_user(), gtalk_create_member(), gtalk_load_config(), jingle_create_member(), jingle_load_config(), reload_config(), set_config(), and update_common_options().
01244 { 01245 int errors = 0; 01246 char *parse = NULL, *this = NULL, *psize = NULL; 01247 int format = 0, framems = 0; 01248 01249 parse = ast_strdupa(list); 01250 while ((this = strsep(&parse, ","))) { 01251 framems = 0; 01252 if ((psize = strrchr(this, ':'))) { 01253 *psize++ = '\0'; 01254 ast_debug(1, "Packetization for codec: %s is %s\n", this, psize); 01255 framems = atoi(psize); 01256 if (framems < 0) { 01257 framems = 0; 01258 errors++; 01259 ast_log(LOG_WARNING, "Bad packetization value for codec %s\n", this); 01260 } 01261 } 01262 if (!(format = ast_getformatbyname(this))) { 01263 ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this); 01264 errors++; 01265 continue; 01266 } 01267 01268 if (mask) { 01269 if (allowing) 01270 *mask |= format; 01271 else 01272 *mask &= ~format; 01273 } 01274 01275 /* Set up a preference list for audio. Do not include video in preferences 01276 since we can not transcode video and have to use whatever is offered 01277 */ 01278 if (pref && (format & AST_FORMAT_AUDIO_MASK)) { 01279 if (strcasecmp(this, "all")) { 01280 if (allowing) { 01281 ast_codec_pref_append(pref, format); 01282 ast_codec_pref_setsize(pref, format, framems); 01283 } 01284 else 01285 ast_codec_pref_remove(pref, format); 01286 } else if (!allowing) { 01287 memset(pref, 0, sizeof(*pref)); 01288 } 01289 } 01290 } 01291 return errors; 01292 }
void ast_smoother_free | ( | struct ast_smoother * | s | ) |
int ast_smoother_get_flags | ( | struct ast_smoother * | smoother | ) |
struct ast_smoother* ast_smoother_new | ( | int | bytes | ) |
Definition at line 179 of file frame.c.
References ast_malloc, ast_smoother_reset(), and s.
Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().
00180 { 00181 struct ast_smoother *s; 00182 if (size < 1) 00183 return NULL; 00184 if ((s = ast_malloc(sizeof(*s)))) 00185 ast_smoother_reset(s, size); 00186 return s; 00187 }
struct ast_frame* ast_smoother_read | ( | struct ast_smoother * | s | ) |
Definition at line 239 of file frame.c.
References ast_format_rate(), AST_FRAME_VOICE, AST_FRIENDLY_OFFSET, ast_log(), ast_samp2tv(), AST_SMOOTHER_FLAG_G729, ast_tvadd(), ast_tvzero(), len(), LOG_WARNING, and s.
Referenced by ast_rtp_write().
00240 { 00241 struct ast_frame *opt; 00242 int len; 00243 00244 /* IF we have an optimization frame, send it */ 00245 if (s->opt) { 00246 if (s->opt->offset < AST_FRIENDLY_OFFSET) 00247 ast_log(LOG_WARNING, "Returning a frame of inappropriate offset (%d).\n", 00248 s->opt->offset); 00249 opt = s->opt; 00250 s->opt = NULL; 00251 return opt; 00252 } 00253 00254 /* Make sure we have enough data */ 00255 if (s->len < s->size) { 00256 /* Or, if this is a G.729 frame with VAD on it, send it immediately anyway */ 00257 if (!((s->flags & AST_SMOOTHER_FLAG_G729) && (s->size % 10))) 00258 return NULL; 00259 } 00260 len = s->size; 00261 if (len > s->len) 00262 len = s->len; 00263 /* Make frame */ 00264 s->f.frametype = AST_FRAME_VOICE; 00265 s->f.subclass = s->format; 00266 s->f.data.ptr = s->framedata + AST_FRIENDLY_OFFSET; 00267 s->f.offset = AST_FRIENDLY_OFFSET; 00268 s->f.datalen = len; 00269 /* Samples will be improper given VAD, but with VAD the concept really doesn't even exist */ 00270 s->f.samples = len * s->samplesperbyte; /* XXX rounding */ 00271 s->f.delivery = s->delivery; 00272 /* Fill Data */ 00273 memcpy(s->f.data.ptr, s->data, len); 00274 s->len -= len; 00275 /* Move remaining data to the front if applicable */ 00276 if (s->len) { 00277 /* In principle this should all be fine because if we are sending 00278 G.729 VAD, the next timestamp will take over anyawy */ 00279 memmove(s->data, s->data + len, s->len); 00280 if (!ast_tvzero(s->delivery)) { 00281 /* If we have delivery time, increment it, otherwise, leave it at 0 */ 00282 s->delivery = ast_tvadd(s->delivery, ast_samp2tv(s->f.samples, ast_format_rate(s->format))); 00283 } 00284 } 00285 /* Return frame */ 00286 return &s->f; 00287 }
void ast_smoother_reconfigure | ( | struct ast_smoother * | s, | |
int | bytes | |||
) |
Reconfigure an existing smoother to output a different number of bytes per frame.
s | the smoother to reconfigure | |
bytes | the desired number of bytes per output frame |
Definition at line 157 of file frame.c.
References s, and smoother_frame_feed().
Referenced by ast_rtp_codec_setpref().
00158 { 00159 /* if there is no change, then nothing to do */ 00160 if (s->size == bytes) { 00161 return; 00162 } 00163 /* set the new desired output size */ 00164 s->size = bytes; 00165 /* if there is no 'optimized' frame in the smoother, 00166 * then there is nothing left to do 00167 */ 00168 if (!s->opt) { 00169 return; 00170 } 00171 /* there is an 'optimized' frame here at the old size, 00172 * but it must now be put into the buffer so the data 00173 * can be extracted at the new size 00174 */ 00175 smoother_frame_feed(s, s->opt, s->opt_needs_swap); 00176 s->opt = NULL; 00177 }
void ast_smoother_reset | ( | struct ast_smoother * | s, | |
int | bytes | |||
) |
Definition at line 151 of file frame.c.
References s.
Referenced by ast_smoother_new().
00152 { 00153 memset(s, 0, sizeof(*s)); 00154 s->size = bytes; 00155 }
void ast_smoother_set_flags | ( | struct ast_smoother * | smoother, | |
int | flags | |||
) |
Definition at line 194 of file frame.c.
References s.
Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().
int ast_smoother_test_flag | ( | struct ast_smoother * | s, | |
int | flag | |||
) |
Definition at line 199 of file frame.c.
References s.
Referenced by ast_rtp_write().
00200 { 00201 return (s->flags & flag); 00202 }
void ast_swapcopy_samples | ( | void * | dst, | |
const void * | src, | |||
int | samples | |||
) |
Definition at line 530 of file frame.c.
Referenced by __ast_smoother_feed(), iax_frame_wrap(), phone_write_buf(), and smoother_frame_feed().
00531 { 00532 int i; 00533 unsigned short *dst_s = dst; 00534 const unsigned short *src_s = src; 00535 00536 for (i = 0; i < samples; i++) 00537 dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8); 00538 }
struct ast_frame ast_null_frame |
Queueing a null frame is fairly common, so we declare a global null frame object for this purpose instead of having to declare one on the stack
Definition at line 127 of file frame.c.
Referenced by __ast_read(), __oh323_rtp_create(), __oh323_update_info(), agent_new(), agent_read(), ast_channel_masquerade(), ast_channel_setwhentohangup_tv(), ast_do_masquerade(), ast_rtcp_read(), ast_rtp_read(), ast_softhangup_nolock(), ast_udptl_read(), conf_run(), console_read(), gtalk_rtp_read(), handle_request_invite(), handle_response_invite(), iax2_read(), jingle_rtp_read(), local_read(), mgcp_rtp_read(), oh323_read(), oh323_rtp_read(), process_rfc2833(), process_sdp(), send_dtmf(), sip_read(), sip_rtp_read(), skinny_rtp_read(), unistim_rtp_read(), and wakeup_sub().