#include <sys/types.h>
#include <sys/time.h>
#include "asterisk/compiler.h"
#include "asterisk/endian.h"
#include "asterisk/linkedlists.h"
Go to the source code of this file.
Data Structures | |
struct | ast_codec_pref |
struct | ast_format_list |
Definition of supported media formats (codecs). More... | |
struct | ast_frame |
Data structure associated with a single frame of data. More... | |
struct | ast_option_header |
struct | oprmode |
Defines | |
#define | AST_FORMAT_ADPCM (1 << 5) |
#define | AST_FORMAT_ALAW (1 << 3) |
#define | AST_FORMAT_AUDIO_MASK ((1 << 16)-1) |
#define | AST_FORMAT_AUDIO_UNDEFINED ((1 << 13) | (1 << 14) | (1 << 15)) |
#define | AST_FORMAT_G722 (1 << 12) |
#define | AST_FORMAT_G723_1 (1 << 0) |
#define | AST_FORMAT_G726 (1 << 11) |
#define | AST_FORMAT_G726_AAL2 (1 << 4) |
#define | AST_FORMAT_G729A (1 << 8) |
#define | AST_FORMAT_GSM (1 << 1) |
#define | AST_FORMAT_H261 (1 << 18) |
#define | AST_FORMAT_H263 (1 << 19) |
#define | AST_FORMAT_H263_PLUS (1 << 20) |
#define | AST_FORMAT_H264 (1 << 21) |
#define | AST_FORMAT_ILBC (1 << 10) |
#define | AST_FORMAT_JPEG (1 << 16) |
#define | AST_FORMAT_LPC10 (1 << 7) |
#define | AST_FORMAT_MAX_AUDIO (1 << 15) |
#define | AST_FORMAT_MAX_VIDEO (1 << 24) |
#define | AST_FORMAT_PNG (1 << 17) |
#define | AST_FORMAT_SLINEAR (1 << 6) |
#define | AST_FORMAT_SPEEX (1 << 9) |
#define | AST_FORMAT_ULAW (1 << 2) |
#define | AST_FORMAT_VIDEO_MASK (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) |
#define | ast_frame_byteswap_be(fr) do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data, __f->data, __f->samples); } while(0) |
#define | ast_frame_byteswap_le(fr) do { ; } while(0) |
#define | AST_FRAME_DTMF AST_FRAME_DTMF_END |
#define | AST_FRAME_SET_BUFFER(fr, _base, _ofs, _datalen) |
#define | ast_frfree(fr) ast_frame_free(fr, 1) |
#define | AST_FRIENDLY_OFFSET 64 |
#define | AST_HTML_BEGIN 4 |
#define | AST_HTML_DATA 2 |
#define | AST_HTML_END 8 |
#define | AST_HTML_LDCOMPLETE 16 |
#define | AST_HTML_LINKREJECT 20 |
#define | AST_HTML_LINKURL 18 |
#define | AST_HTML_NOSUPPORT 17 |
#define | AST_HTML_UNLINK 19 |
#define | AST_HTML_URL 1 |
#define | AST_MALLOCD_DATA (1 << 1) |
#define | AST_MALLOCD_HDR (1 << 0) |
#define | AST_MALLOCD_SRC (1 << 2) |
#define | AST_MIN_OFFSET 32 |
#define | AST_MODEM_T38 1 |
#define | AST_MODEM_V150 2 |
#define | AST_OPTION_AUDIO_MODE 4 |
#define | AST_OPTION_CHANNEL_WRITE 9 |
Handle channel write data If a channel needs to process the data from a func_channel write operation after func_channel_write executes, it can define the setoption callback and process this option. A pointer to an ast_chan_write_info_t will be passed. | |
#define | AST_OPTION_ECHOCAN 8 |
#define | AST_OPTION_FLAG_ACCEPT 1 |
#define | AST_OPTION_FLAG_ANSWER 5 |
#define | AST_OPTION_FLAG_QUERY 4 |
#define | AST_OPTION_FLAG_REJECT 2 |
#define | AST_OPTION_FLAG_REQUEST 0 |
#define | AST_OPTION_FLAG_WTF 6 |
#define | AST_OPTION_OPRMODE 7 |
#define | AST_OPTION_RELAXDTMF 3 |
#define | AST_OPTION_RXGAIN 6 |
#define | AST_OPTION_TDD 2 |
#define | AST_OPTION_TONE_VERIFY 1 |
#define | AST_OPTION_TXGAIN 5 |
#define | ast_smoother_feed(s, f) __ast_smoother_feed(s, f, 0) |
#define | ast_smoother_feed_be(s, f) __ast_smoother_feed(s, f, 1) |
#define | ast_smoother_feed_le(s, f) __ast_smoother_feed(s, f, 0) |
#define | AST_SMOOTHER_FLAG_BE (1 << 1) |
#define | AST_SMOOTHER_FLAG_G729 (1 << 0) |
Enumerations | |
enum | { AST_FRFLAG_HAS_TIMING_INFO = (1 << 0) } |
enum | ast_control_frame_type { AST_CONTROL_HANGUP = 1, AST_CONTROL_RING = 2, AST_CONTROL_RINGING = 3, AST_CONTROL_ANSWER = 4, AST_CONTROL_BUSY = 5, AST_CONTROL_TAKEOFFHOOK = 6, AST_CONTROL_OFFHOOK = 7, AST_CONTROL_CONGESTION = 8, AST_CONTROL_FLASH = 9, AST_CONTROL_WINK = 10, AST_CONTROL_OPTION = 11, AST_CONTROL_RADIO_KEY = 12, AST_CONTROL_RADIO_UNKEY = 13, AST_CONTROL_PROGRESS = 14, AST_CONTROL_PROCEEDING = 15, AST_CONTROL_HOLD = 16, AST_CONTROL_UNHOLD = 17, AST_CONTROL_VIDUPDATE = 18, AST_CONTROL_SRCUPDATE = 20, AST_CONTROL_SRCCHANGE = 21, AST_CONTROL_END_OF_Q = 22 } |
enum | ast_frame_type { AST_FRAME_DTMF_END = 1, AST_FRAME_VOICE, AST_FRAME_VIDEO, AST_FRAME_CONTROL, AST_FRAME_NULL, AST_FRAME_IAX, AST_FRAME_TEXT, AST_FRAME_IMAGE, AST_FRAME_HTML, AST_FRAME_CNG, AST_FRAME_MODEM, AST_FRAME_DTMF_BEGIN } |
Frame types. More... | |
Functions | |
int | __ast_smoother_feed (struct ast_smoother *s, struct ast_frame *f, int swap) |
char * | ast_codec2str (int codec) |
Get a name from a format Gets a name from a format. | |
int | ast_codec_choose (struct ast_codec_pref *pref, int formats, int find_best) |
Select the best audio format according to preference list from supplied options. If "find_best" is non-zero then if nothing is found, the "Best" format of the format list is selected, otherwise 0 is returned. | |
int | ast_codec_get_len (int format, int samples) |
Returns the number of bytes for the number of samples of the given format. | |
int | ast_codec_get_samples (struct ast_frame *f) |
Returns the number of samples contained in the frame. | |
static int | ast_codec_interp_len (int format) |
Gets duration in ms of interpolation frame for a format. | |
int | ast_codec_pref_append (struct ast_codec_pref *pref, int format) |
Append a audio codec to a preference list, removing it first if it was already there. | |
void | ast_codec_pref_convert (struct ast_codec_pref *pref, char *buf, size_t size, int right) |
Shift an audio codec preference list up or down 65 bytes so that it becomes an ASCII string. | |
ast_format_list | ast_codec_pref_getsize (struct ast_codec_pref *pref, int format) |
Get packet size for codec. | |
int | ast_codec_pref_index (struct ast_codec_pref *pref, int index) |
Codec located at a particular place in the preference index See Audio Codec Preferences. | |
void | ast_codec_pref_init (struct ast_codec_pref *pref) |
Initialize an audio codec preference to "no preference" See Audio Codec Preferences. | |
void | ast_codec_pref_prepend (struct ast_codec_pref *pref, int format, int only_if_existing) |
Prepend an audio codec to a preference list, removing it first if it was already there. | |
void | ast_codec_pref_remove (struct ast_codec_pref *pref, int format) |
Remove audio a codec from a preference list. | |
int | ast_codec_pref_setsize (struct ast_codec_pref *pref, int format, int framems) |
Set packet size for codec. | |
int | ast_codec_pref_string (struct ast_codec_pref *pref, char *buf, size_t size) |
Dump audio codec preference list into a string. | |
static force_inline int | ast_format_rate (int format) |
Get the sample rate for a given format. | |
int | ast_frame_adjust_volume (struct ast_frame *f, int adjustment) |
Adjusts the volume of the audio samples contained in a frame. | |
void | ast_frame_dump (const char *name, struct ast_frame *f, char *prefix) |
ast_frame * | ast_frame_enqueue (struct ast_frame *head, struct ast_frame *f, int maxlen, int dupe) |
Appends a frame to the end of a list of frames, truncating the maximum length of the list. | |
void | ast_frame_free (struct ast_frame *fr, int cache) |
Requests a frame to be allocated Frees a frame or list of frames. | |
int | ast_frame_slinear_sum (struct ast_frame *f1, struct ast_frame *f2) |
Sums two frames of audio samples. | |
ast_frame * | ast_frdup (const struct ast_frame *fr) |
Copies a frame. | |
ast_frame * | ast_frisolate (struct ast_frame *fr) |
Makes a frame independent of any static storage. | |
ast_format_list * | ast_get_format_list (size_t *size) |
ast_format_list * | ast_get_format_list_index (int index) |
int | ast_getformatbyname (const char *name) |
Gets a format from a name. | |
char * | ast_getformatname (int format) |
Get the name of a format. | |
char * | ast_getformatname_multiple (char *buf, size_t size, int format) |
Get the names of a set of formats. | |
void | ast_parse_allow_disallow (struct ast_codec_pref *pref, int *mask, const char *list, int allowing) |
Parse an "allow" or "deny" line in a channel or device configuration and update the capabilities mask and pref if provided. Video codecs are not added to codec preference lists, since we can not transcode. | |
void | ast_smoother_free (struct ast_smoother *s) |
int | ast_smoother_get_flags (struct ast_smoother *smoother) |
ast_smoother * | ast_smoother_new (int bytes) |
ast_frame * | ast_smoother_read (struct ast_smoother *s) |
void | ast_smoother_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) |
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 237 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 233 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(), oh323_rtp_read(), pcm_seek(), pcm_write(), and sms_generate().
#define AST_FORMAT_AUDIO_MASK ((1 << 16)-1) |
Maximum audio mask
Definition at line 257 of file frame.h.
Referenced by add_sdp(), ast_best_codec(), ast_closestream(), ast_codec_choose(), ast_openstream_full(), ast_parse_allow_disallow(), ast_request(), ast_translate_available_formats(), ast_translator_best_choice(), begin_dial(), func_channel_read(), generator_force(), gtalk_rtp_read(), process_sdp(), set_format(), sip_call(), sip_rtp_read(), and sip_write().
#define AST_FORMAT_AUDIO_UNDEFINED ((1 << 13) | (1 << 14) | (1 << 15)) |
#define AST_FORMAT_G722 (1 << 12) |
G.722
Definition at line 251 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_raw_write(), au_seek(), convertcap(), g722tolin_sample(), pcm_read(), and rtp_get_rate().
#define AST_FORMAT_G723_1 (1 << 0) |
G.723.1 compression
Definition at line 227 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_samples(), ast_rtp_write(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_destroy(), dahdi_translate(), 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 249 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 235 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 243 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(), and g729_write().
#define AST_FORMAT_GSM (1 << 1) |
GSM compression
Definition at line 229 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 263 of file frame.h.
Referenced by codec_ast2skinny(), and codec_skinny2ast().
#define AST_FORMAT_H263 (1 << 19) |
H.263 Video
Definition at line 265 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 247 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 259 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 241 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 255 of file frame.h.
Referenced by add_sdp(), ast_filehelper(), ast_openvstream(), ast_playstream(), ast_rtp_read(), ast_translate_available_formats(), ast_writestream(), filestream_destructor(), oh323_request(), phone_read(), sip_request_call(), skinny_request(), transmit_connect_with_sdp(), and transmit_modify_with_sdp().
#define AST_FORMAT_MAX_VIDEO (1 << 24) |
Maximum video format
Definition at line 271 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 239 of file frame.h.
Referenced by __ast_play_and_record(), action_originate(), agent_new(), alsa_new(), alsa_read(), alsa_request(), ast_audiohook_read_frame(), ast_best_codec(), ast_channel_make_compatible(), ast_channel_start_silence_generator(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_call_progress(), ast_dsp_digitdetect(), ast_dsp_process(), ast_dsp_silence(), ast_frame_adjust_volume(), ast_frame_slinear_sum(), ast_rtp_read(), ast_slinfactory_feed(), ast_write(), 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(), find_transcoders(), handle_recordfile(), iax_frame_wrap(), ices_exec(), init_outgoing(), is_encoder(), isAnsweringMachine(), linear_alloc(), linear_generator(), lintoadpcm_sample(), lintoalaw_sample(), lintog722_sample(), lintog726_sample(), lintogsm_sample(), lintoilbc_sample(), lintolpc10_sample(), lintospeex_sample(), lintoulaw_sample(), load_module(), measurenoise(), 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(), rpt(), rpt_call(), rpt_tele_thread(), send_waveform_to_channel(), silence_generator_generate(), slinear_read(), slinear_write(), sms_generate(), socket_process(), speech_background(), speech_create(), spy_generate(), tonepair_alloc(), wav_read(), and wav_write().
#define AST_FORMAT_SPEEX (1 << 9) |
SpeeX Free Compression
Definition at line 245 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 231 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(), calc_energy(), codec_ast2skinny(), codec_skinny2ast(), conf_run(), convertcap(), dahdi_new(), dahdi_read(), dahdi_translate(), dahdi_write(), disa_exec(), 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(), ulawtoalaw_sample(), and ulawtolin_sample().
#define AST_FORMAT_VIDEO_MASK (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) |
Definition at line 272 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 126 of file frame.h.
Referenced by __action_dialoffhook(), __adsi_transmit_messages(), __ast_play_and_record(), agent_ack_sleep(), app_exec(), ast_audiohook_write_list(), ast_bridge_call(), ast_dsp_process(), ast_jb_put(), background_detect_exec(), cb_events(), channel_spy(), conf_exec(), conf_run(), console_dial(), console_dial_deprecated(), dahdi_bridge(), dictate_exec(), disa_exec(), do_immediate_setup(), echo_exec(), feature_request_and_dial(), gtalk_handle_dtmf(), handle_recordfile(), handle_request(), handle_request_info(), mgcp_rtp_read(), misdn_bridge(), mp3_exec(), NBScat_exec(), oh323_rtp_read(), phone_exception(), process_ast_dsp(), receive_dtmf_digits(), rpt(), rpt_call(), send_waveform_to_channel(), sip_rtp_read(), speech_background(), ss_thread(), wait_for_answer(), and wait_for_winner().
#define AST_FRAME_SET_BUFFER | ( | fr, | |||
_base, | |||||
_ofs, | |||||
_datalen | ) |
Value:
Set the various field of a frame to point to a buffer. Typically you set the base address of the buffer, the offset as AST_FRIENDLY_OFFSET, and the datalen as the amount of bytes queued. The remaining things (to be done manually) is set the number of samples, which cannot be derived from the datalen unless you know the number of bits per sample.Definition at line 176 of file frame.h.
Referenced by g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), ilbc_read(), ogg_vorbis_read(), pcm_read(), slinear_read(), vox_read(), and wav_read().
#define ast_frfree | ( | fr | ) | ast_frame_free(fr, 1) |
Definition at line 407 of file frame.h.
Referenced by __adsi_transmit_messages(), __ast_play_and_record(), __ast_queue_frame(), __ast_read(), __ast_request_and_dial(), adsi_careful_send(), agent_ack_sleep(), agent_read(), app_exec(), ast_audiohook_read_frame(), ast_autoservice_stop(), ast_bridge_call(), ast_channel_clear_softhangup(), ast_channel_free(), ast_dsp_process(), 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(), calc_cost(), channel_spy(), check_goto_on_transfer(), cli_audio_convert(), cli_audio_convert_deprecated(), conf_exec(), conf_flush(), conf_free(), conf_run(), create_jb(), dahdi_bridge(), dictate_exec(), disa_exec(), do_idle_thread(), do_parking_thread(), do_waiting(), echo_exec(), feature_request_and_dial(), find_cache(), gen_generate(), handle_recordfile(), iax_park_thread(), ices_exec(), isAnsweringMachine(), jb_empty_and_reset_adaptive(), jb_empty_and_reset_fixed(), jb_get_and_deliver(), masq_park_call(), measurenoise(), moh_files_generator(), monitor_dial(), mp3_exec(), NBScat_exec(), read_frame(), receive_dtmf_digits(), recordthread(), rpt(), run_agi(), send_tone_burst(), send_waveform_to_channel(), sendurl_exec(), speech_background(), spy_generate(), ss_thread(), wait_for_answer(), wait_for_hangup(), wait_for_winner(), waitforring_exec(), and waitstream_core().
#define AST_FRIENDLY_OFFSET 64 |
Definition at line 187 of file frame.h.
Referenced by __get_from_jb(), adjust_frame_for_plc(), alsa_read(), ast_frdup(), ast_frisolate(), ast_prod(), ast_rtcp_read(), ast_rtp_read(), ast_smoother_read(), ast_trans_frameout(), ast_udptl_read(), conf_run(), dahdi_decoder_frameout(), dahdi_encoder_frameout(), dahdi_read(), g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), iax_frame_wrap(), ilbc_read(), jb_get_and_deliver(), linear_generator(), milliwatt_generate(), moh_generate(), mohalloc(), mp3_exec(), NBScat_exec(), newpvt(), ogg_vorbis_read(), oss_read(), pcm_read(), phone_read(), process_rfc3389(), send_tone_burst(), send_waveform_to_channel(), slinear_read(), sms_generate(), vox_read(), 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 215 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 217 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 207 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 195 of file frame.h.
Referenced by __frame_free(), and ast_frisolate().
#define AST_MALLOCD_HDR (1 << 0) |
Need the header be free'd?
Definition at line 193 of file frame.h.
Referenced by __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 197 of file frame.h.
Referenced by __frame_free(), and ast_frisolate().
#define AST_MIN_OFFSET 32 |
#define AST_MODEM_T38 1 |
T.38 Fax-over-IP
Definition at line 201 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 320 of file frame.h.
Referenced by ast_bridge_call(), dahdi_hangup(), dahdi_setoption(), and iax2_setoption().
#define AST_OPTION_CHANNEL_WRITE 9 |
Handle channel write data If a channel needs to process the data from a func_channel write operation after func_channel_write executes, it can define the setoption callback and process this option. A pointer to an ast_chan_write_info_t will be passed.
Definition at line 349 of file frame.h.
Referenced by func_channel_write(), and local_setoption().
#define AST_OPTION_ECHOCAN 8 |
Explicitly enable or disable echo cancelation for the given channel
Definition at line 342 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 317 of file frame.h.
Referenced by ast_bridge_call(), dahdi_setoption(), iax2_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 336 of file frame.h.
Referenced by dahdi_setoption(), func_channel_write_real(), iax2_setoption(), play_record_review(), reset_volumes(), set_talk_volume(), and vm_forwardoptions().
#define AST_OPTION_TDD 2 |
Put a compatible channel into TDD (TTY for the hearing-impared) mode
Definition at line 314 of file frame.h.
Referenced by ast_bridge_call(), dahdi_hangup(), dahdi_setoption(), handle_tddmode(), and iax2_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 311 of file frame.h.
Referenced by ast_bridge_call(), conf_run(), dahdi_hangup(), dahdi_setoption(), iax2_setoption(), and rpt().
#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 328 of file frame.h.
Referenced by common_exec(), dahdi_setoption(), func_channel_write_real(), 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 298 of file frame.h.
Referenced by __ast_smoother_feed(), ast_smoother_read(), and smoother_frame_feed().
anonymous enum |
Definition at line 128 of file frame.h.
00128 { 00129 /*! This frame contains valid timing information */ 00130 AST_FRFLAG_HAS_TIMING_INFO = (1 << 0), 00131 };
Definition at line 274 of file frame.h.
00274 { 00275 AST_CONTROL_HANGUP = 1, /*!< Other end has hungup */ 00276 AST_CONTROL_RING = 2, /*!< Local ring */ 00277 AST_CONTROL_RINGING = 3, /*!< Remote end is ringing */ 00278 AST_CONTROL_ANSWER = 4, /*!< Remote end has answered */ 00279 AST_CONTROL_BUSY = 5, /*!< Remote end is busy */ 00280 AST_CONTROL_TAKEOFFHOOK = 6, /*!< Make it go off hook */ 00281 AST_CONTROL_OFFHOOK = 7, /*!< Line is off hook */ 00282 AST_CONTROL_CONGESTION = 8, /*!< Congestion (circuits busy) */ 00283 AST_CONTROL_FLASH = 9, /*!< Flash hook */ 00284 AST_CONTROL_WINK = 10, /*!< Wink */ 00285 AST_CONTROL_OPTION = 11, /*!< Set a low-level option */ 00286 AST_CONTROL_RADIO_KEY = 12, /*!< Key Radio */ 00287 AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */ 00288 AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */ 00289 AST_CONTROL_PROCEEDING = 15, /*!< Indicate CALL PROCEEDING */ 00290 AST_CONTROL_HOLD = 16, /*!< Indicate call is placed on hold */ 00291 AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold */ 00292 AST_CONTROL_VIDUPDATE = 18, /*!< Indicate video frame update */ 00293 AST_CONTROL_SRCUPDATE = 20, /*!< Indicate source of media has changed */ 00294 AST_CONTROL_SRCCHANGE = 21, /*!< Media has changed and requires a new RTP SSRC */ 00295 AST_CONTROL_END_OF_Q = 22, /*!< Indicate that this position was the end of the channel queue for a softhangup. */ 00296 };
enum ast_frame_type |
Frame types.
Definition at line 99 of file frame.h.
00099 { 00100 /*! DTMF end event, subclass is the digit */ 00101 AST_FRAME_DTMF_END = 1, 00102 /*! Voice data, subclass is AST_FORMAT_* */ 00103 AST_FRAME_VOICE, 00104 /*! Video frame, maybe?? :) */ 00105 AST_FRAME_VIDEO, 00106 /*! A control frame, subclass is AST_CONTROL_* */ 00107 AST_FRAME_CONTROL, 00108 /*! An empty, useless frame */ 00109 AST_FRAME_NULL, 00110 /*! Inter Asterisk Exchange private frame type */ 00111 AST_FRAME_IAX, 00112 /*! Text messages */ 00113 AST_FRAME_TEXT, 00114 /*! Image Frames */ 00115 AST_FRAME_IMAGE, 00116 /*! HTML Frame */ 00117 AST_FRAME_HTML, 00118 /*! Comfort Noise frame (subclass is level of CNG in -dBov), 00119 body may include zero or more 8-bit quantization coefficients */ 00120 AST_FRAME_CNG, 00121 /*! Modem-over-IP data streams */ 00122 AST_FRAME_MODEM, 00123 /*! DTMF begin event, subclass is the digit */ 00124 AST_FRAME_DTMF_BEGIN, 00125 };
int __ast_smoother_feed | ( | struct ast_smoother * | s, | |
struct ast_frame * | f, | |||
int | swap | |||
) |
Definition at line 211 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.
00212 { 00213 if (f->frametype != AST_FRAME_VOICE) { 00214 ast_log(LOG_WARNING, "Huh? Can't smooth a non-voice frame!\n"); 00215 return -1; 00216 } 00217 if (!s->format) { 00218 s->format = f->subclass; 00219 s->samplesperbyte = (float)f->samples / (float)f->datalen; 00220 } else if (s->format != f->subclass) { 00221 ast_log(LOG_WARNING, "Smoother was working on %d format frames, now trying to feed %d?\n", s->format, f->subclass); 00222 return -1; 00223 } 00224 if (s->len + f->datalen > SMOOTHER_SIZE) { 00225 ast_log(LOG_WARNING, "Out of smoother space\n"); 00226 return -1; 00227 } 00228 if (((f->datalen == s->size) || 00229 ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) && 00230 !s->opt && 00231 !s->len && 00232 (f->offset >= AST_MIN_OFFSET)) { 00233 /* Optimize by sending the frame we just got 00234 on the next read, thus eliminating the douple 00235 copy */ 00236 if (swap) 00237 ast_swapcopy_samples(f->data, f->data, f->samples); 00238 s->opt = f; 00239 s->opt_needs_swap = swap ? 1 : 0; 00240 return 0; 00241 } 00242 00243 return smoother_frame_feed(s, f, swap); 00244 }
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 645 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().
00646 { 00647 int x; 00648 char *ret = "unknown"; 00649 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00650 if(AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == codec) { 00651 ret = AST_FORMAT_LIST[x].desc; 00652 break; 00653 } 00654 } 00655 return ret; 00656 }
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 1284 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().
01285 { 01286 int x, ret = 0, slot; 01287 01288 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01289 slot = pref->order[x]; 01290 01291 if (!slot) 01292 break; 01293 if (formats & AST_FORMAT_LIST[slot-1].bits) { 01294 ret = AST_FORMAT_LIST[slot-1].bits; 01295 break; 01296 } 01297 } 01298 if(ret & AST_FORMAT_AUDIO_MASK) 01299 return ret; 01300 01301 if (option_debug > 3) 01302 ast_log(LOG_DEBUG, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec"); 01303 01304 return find_best ? ast_best_codec(formats) : 0; 01305 }
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 1545 of file frame.c.
References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G722, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_ILBC, AST_FORMAT_SLINEAR, AST_FORMAT_ULAW, ast_getformatname(), ast_log(), len(), and LOG_WARNING.
Referenced by moh_generate(), and monmp3thread().
01546 { 01547 int len = 0; 01548 01549 /* XXX Still need speex, g723, and lpc10 XXX */ 01550 switch(format) { 01551 case AST_FORMAT_ILBC: 01552 len = (samples / 240) * 50; 01553 break; 01554 case AST_FORMAT_GSM: 01555 len = (samples / 160) * 33; 01556 break; 01557 case AST_FORMAT_G729A: 01558 len = samples / 8; 01559 break; 01560 case AST_FORMAT_SLINEAR: 01561 len = samples * 2; 01562 break; 01563 case AST_FORMAT_ULAW: 01564 case AST_FORMAT_ALAW: 01565 len = samples; 01566 break; 01567 case AST_FORMAT_G722: 01568 case AST_FORMAT_ADPCM: 01569 case AST_FORMAT_G726: 01570 case AST_FORMAT_G726_AAL2: 01571 len = samples / 2; 01572 break; 01573 default: 01574 ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format)); 01575 } 01576 01577 return len; 01578 }
int ast_codec_get_samples | ( | struct ast_frame * | f | ) |
Returns the number of samples contained in the frame.
Definition at line 1502 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().
01503 { 01504 int samples=0; 01505 switch(f->subclass) { 01506 case AST_FORMAT_SPEEX: 01507 samples = speex_samples(f->data, f->datalen); 01508 break; 01509 case AST_FORMAT_G723_1: 01510 samples = g723_samples(f->data, f->datalen); 01511 break; 01512 case AST_FORMAT_ILBC: 01513 samples = 240 * (f->datalen / 50); 01514 break; 01515 case AST_FORMAT_GSM: 01516 samples = 160 * (f->datalen / 33); 01517 break; 01518 case AST_FORMAT_G729A: 01519 samples = f->datalen * 8; 01520 break; 01521 case AST_FORMAT_SLINEAR: 01522 samples = f->datalen / 2; 01523 break; 01524 case AST_FORMAT_LPC10: 01525 /* assumes that the RTP packet contains one LPC10 frame */ 01526 samples = 22 * 8; 01527 samples += (((char *)(f->data))[7] & 0x1) * 8; 01528 break; 01529 case AST_FORMAT_ULAW: 01530 case AST_FORMAT_ALAW: 01531 samples = f->datalen; 01532 break; 01533 case AST_FORMAT_G722: 01534 case AST_FORMAT_ADPCM: 01535 case AST_FORMAT_G726: 01536 case AST_FORMAT_G726_AAL2: 01537 samples = f->datalen * 2; 01538 break; 01539 default: 01540 ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass)); 01541 } 01542 return samples; 01543 }
static int ast_codec_interp_len | ( | int | format | ) | [inline, static] |
Gets duration in ms of interpolation frame for a format.
Definition at line 581 of file frame.h.
References AST_FORMAT_ILBC.
Referenced by __get_from_jb(), and jb_get_and_deliver().
00582 { 00583 return (format == AST_FORMAT_ILBC) ? 30 : 20; 00584 }
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 1143 of file frame.c.
References ast_codec_pref_remove(), AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow().
01144 { 01145 int x, newindex = 0; 01146 01147 ast_codec_pref_remove(pref, format); 01148 01149 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01150 if(AST_FORMAT_LIST[x].bits == format) { 01151 newindex = x + 1; 01152 break; 01153 } 01154 } 01155 01156 if(newindex) { 01157 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01158 if(!pref->order[x]) { 01159 pref->order[x] = newindex; 01160 break; 01161 } 01162 } 01163 } 01164 01165 return x; 01166 }
void ast_codec_pref_convert | ( | struct ast_codec_pref * | pref, | |
char * | buf, | |||
size_t | size, | |||
int | right | |||
) |
Shift an audio codec preference list up or down 65 bytes so that it becomes an ASCII string.
pref | A codec preference list structure | |
buf | A string denoting codec preference, appropriate for use in line transmission | |
size | Size of buf | |
right | Boolean: if 0, convert from buf to pref; if 1, convert from pref to buf. |
Definition at line 1045 of file frame.c.
References ast_codec_pref::order.
Referenced by check_access(), create_addr(), dump_prefs(), and socket_process().
01046 { 01047 int x, differential = (int) 'A', mem; 01048 char *from, *to; 01049 01050 if(right) { 01051 from = pref->order; 01052 to = buf; 01053 mem = size; 01054 } else { 01055 to = pref->order; 01056 from = buf; 01057 mem = 32; 01058 } 01059 01060 memset(to, 0, mem); 01061 for (x = 0; x < 32 ; x++) { 01062 if(!from[x]) 01063 break; 01064 to[x] = right ? (from[x] + differential) : (from[x] - differential); 01065 } 01066 }
struct ast_format_list ast_codec_pref_getsize | ( | struct ast_codec_pref * | pref, | |
int | format | |||
) |
Get packet size for codec.
Definition at line 1245 of file frame.c.
References AST_FORMAT_LIST, ast_format_list::bits, and format.
Referenced by add_codec_to_sdp(), ast_rtp_bridge(), ast_rtp_codec_setpref(), ast_rtp_write(), handle_open_receive_channel_ack_message(), and transmit_connect().
01246 { 01247 int x, index = -1, framems = 0; 01248 struct ast_format_list fmt = {0}; 01249 01250 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01251 if(AST_FORMAT_LIST[x].bits == format) { 01252 fmt = AST_FORMAT_LIST[x]; 01253 index = x; 01254 break; 01255 } 01256 } 01257 01258 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01259 if(pref->order[x] == (index + 1)) { 01260 framems = pref->framing[x]; 01261 break; 01262 } 01263 } 01264 01265 /* size validation */ 01266 if(!framems) 01267 framems = AST_FORMAT_LIST[index].def_ms; 01268 01269 if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */ 01270 framems -= framems % AST_FORMAT_LIST[index].inc_ms; 01271 01272 if(framems < AST_FORMAT_LIST[index].min_ms) 01273 framems = AST_FORMAT_LIST[index].min_ms; 01274 01275 if(framems > AST_FORMAT_LIST[index].max_ms) 01276 framems = AST_FORMAT_LIST[index].max_ms; 01277 01278 fmt.cur_ms = framems; 01279 01280 return fmt; 01281 }
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 1103 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().
01104 { 01105 int slot = 0; 01106 01107 01108 if((index >= 0) && (index < sizeof(pref->order))) { 01109 slot = pref->order[index]; 01110 } 01111 01112 return slot ? AST_FORMAT_LIST[slot-1].bits : 0; 01113 }
void ast_codec_pref_init | ( | struct ast_codec_pref * | pref | ) |
Initialize an audio codec preference to "no preference" See Audio Codec Preferences.
void ast_codec_pref_prepend | ( | struct ast_codec_pref * | pref, | |
int | format, | |||
int | only_if_existing | |||
) |
Prepend an audio codec to a preference list, removing it first if it was already there.
Definition at line 1169 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_codec_pref::framing, and ast_codec_pref::order.
Referenced by create_addr().
01170 { 01171 int x, newindex = 0; 01172 01173 /* First step is to get the codecs "index number" */ 01174 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01175 if (AST_FORMAT_LIST[x].bits == format) { 01176 newindex = x + 1; 01177 break; 01178 } 01179 } 01180 /* Done if its unknown */ 01181 if (!newindex) 01182 return; 01183 01184 /* Now find any existing occurrence, or the end */ 01185 for (x = 0; x < 32; x++) { 01186 if (!pref->order[x] || pref->order[x] == newindex) 01187 break; 01188 } 01189 01190 if (only_if_existing && !pref->order[x]) 01191 return; 01192 01193 /* Move down to make space to insert - either all the way to the end, 01194 or as far as the existing location (which will be overwritten) */ 01195 for (; x > 0; x--) { 01196 pref->order[x] = pref->order[x - 1]; 01197 pref->framing[x] = pref->framing[x - 1]; 01198 } 01199 01200 /* And insert the new entry */ 01201 pref->order[0] = newindex; 01202 pref->framing[0] = 0; /* ? */ 01203 }
void ast_codec_pref_remove | ( | struct ast_codec_pref * | pref, | |
int | format | |||
) |
Remove audio a codec from a preference list.
Definition at line 1116 of file frame.c.
References AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_codec_pref_append(), and ast_parse_allow_disallow().
01117 { 01118 struct ast_codec_pref oldorder; 01119 int x, y = 0; 01120 int slot; 01121 int size; 01122 01123 if(!pref->order[0]) 01124 return; 01125 01126 memcpy(&oldorder, pref, sizeof(oldorder)); 01127 memset(pref, 0, sizeof(*pref)); 01128 01129 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01130 slot = oldorder.order[x]; 01131 size = oldorder.framing[x]; 01132 if(! slot) 01133 break; 01134 if(AST_FORMAT_LIST[slot-1].bits != format) { 01135 pref->order[y] = slot; 01136 pref->framing[y++] = size; 01137 } 01138 } 01139 01140 }
int ast_codec_pref_setsize | ( | struct ast_codec_pref * | pref, | |
int | format, | |||
int | framems | |||
) |
Set packet size for codec.
Definition at line 1206 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_a_audio().
01207 { 01208 int x, index = -1; 01209 01210 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01211 if(AST_FORMAT_LIST[x].bits == format) { 01212 index = x; 01213 break; 01214 } 01215 } 01216 01217 if(index < 0) 01218 return -1; 01219 01220 /* size validation */ 01221 if(!framems) 01222 framems = AST_FORMAT_LIST[index].def_ms; 01223 01224 if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */ 01225 framems -= framems % AST_FORMAT_LIST[index].inc_ms; 01226 01227 if(framems < AST_FORMAT_LIST[index].min_ms) 01228 framems = AST_FORMAT_LIST[index].min_ms; 01229 01230 if(framems > AST_FORMAT_LIST[index].max_ms) 01231 framems = AST_FORMAT_LIST[index].max_ms; 01232 01233 01234 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01235 if(pref->order[x] == (index + 1)) { 01236 pref->framing[x] = framems; 01237 break; 01238 } 01239 } 01240 01241 return x; 01242 }
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 1068 of file frame.c.
References ast_codec_pref_index(), and ast_getformatname().
Referenced by dump_prefs(), and socket_process().
01069 { 01070 int x, codec; 01071 size_t total_len, slen; 01072 char *formatname; 01073 01074 memset(buf,0,size); 01075 total_len = size; 01076 buf[0] = '('; 01077 total_len--; 01078 for(x = 0; x < 32 ; x++) { 01079 if(total_len <= 0) 01080 break; 01081 if(!(codec = ast_codec_pref_index(pref,x))) 01082 break; 01083 if((formatname = ast_getformatname(codec))) { 01084 slen = strlen(formatname); 01085 if(slen > total_len) 01086 break; 01087 strncat(buf, formatname, total_len - 1); /* safe */ 01088 total_len -= slen; 01089 } 01090 if(total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) { 01091 strncat(buf, "|", total_len - 1); /* safe */ 01092 total_len--; 01093 } 01094 } 01095 if(total_len) { 01096 strncat(buf, ")", total_len - 1); /* safe */ 01097 total_len--; 01098 } 01099 01100 return size - total_len; 01101 }
static force_inline int ast_format_rate | ( | int | format | ) | [static] |
Get the sample rate for a given format.
Definition at line 608 of file frame.h.
References AST_FORMAT_G722.
Referenced by __get_from_jb(), ast_read_generator_actions(), ast_readaudio_callback(), ast_readvideo_callback(), ast_rtp_read(), ast_translate(), calc_cost(), calc_timestamp(), generator_force(), rtp_get_rate(), and schedule_delivery().
00609 { 00610 if (format == AST_FORMAT_G722) 00611 return 16000; 00612 00613 return 8000; 00614 }
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 1580 of file frame.c.
References AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_slinear_saturated_divide(), ast_slinear_saturated_multiply(), and f.
Referenced by audiohook_read_frame_single(), and conf_run().
01581 { 01582 int count; 01583 short *fdata = f->data; 01584 short adjust_value = abs(adjustment); 01585 01586 if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR)) 01587 return -1; 01588 01589 if (!adjustment) 01590 return 0; 01591 01592 for (count = 0; count < f->samples; count++) { 01593 if (adjustment > 0) { 01594 ast_slinear_saturated_multiply(&fdata[count], &adjust_value); 01595 } else if (adjustment < 0) { 01596 ast_slinear_saturated_divide(&fdata[count], &adjust_value); 01597 } 01598 } 01599 01600 return 0; 01601 }
void ast_frame_dump | ( | const char * | name, | |
struct ast_frame * | f, | |||
char * | prefix | |||
) |
Dump a frame for debugging purposes
Definition at line 799 of file frame.c.
References AST_CONTROL_ANSWER, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_FLASH, AST_CONTROL_HANGUP, AST_CONTROL_HOLD, AST_CONTROL_OFFHOOK, AST_CONTROL_OPTION, AST_CONTROL_PROCEEDING, AST_CONTROL_PROGRESS, AST_CONTROL_RADIO_KEY, AST_CONTROL_RADIO_UNKEY, AST_CONTROL_RING, AST_CONTROL_RINGING, AST_CONTROL_TAKEOFFHOOK, AST_CONTROL_UNHOLD, AST_CONTROL_WINK, ast_copy_string(), AST_FRAME_CONTROL, AST_FRAME_DTMF_BEGIN, AST_FRAME_DTMF_END, AST_FRAME_HTML, AST_FRAME_IAX, AST_FRAME_IMAGE, AST_FRAME_MODEM, AST_FRAME_NULL, AST_FRAME_TEXT, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_getformatname(), AST_HTML_BEGIN, AST_HTML_DATA, AST_HTML_END, AST_HTML_LDCOMPLETE, AST_HTML_LINKREJECT, AST_HTML_LINKURL, AST_HTML_NOSUPPORT, AST_HTML_UNLINK, AST_HTML_URL, AST_MODEM_T38, AST_MODEM_V150, ast_strlen_zero(), ast_verbose(), COLOR_BLACK, COLOR_BRCYAN, COLOR_BRGREEN, COLOR_BRMAGENTA, COLOR_BRRED, COLOR_YELLOW, f, and term_color().
Referenced by __ast_read(), and ast_write().
00800 { 00801 const char noname[] = "unknown"; 00802 char ftype[40] = "Unknown Frametype"; 00803 char cft[80]; 00804 char subclass[40] = "Unknown Subclass"; 00805 char csub[80]; 00806 char moreinfo[40] = ""; 00807 char cn[60]; 00808 char cp[40]; 00809 char cmn[40]; 00810 00811 if (!name) 00812 name = noname; 00813 00814 00815 if (!f) { 00816 ast_verbose("%s [ %s (NULL) ] [%s]\n", 00817 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00818 term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00819 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00820 return; 00821 } 00822 /* XXX We should probably print one each of voice and video when the format changes XXX */ 00823 if (f->frametype == AST_FRAME_VOICE) 00824 return; 00825 if (f->frametype == AST_FRAME_VIDEO) 00826 return; 00827 switch(f->frametype) { 00828 case AST_FRAME_DTMF_BEGIN: 00829 strcpy(ftype, "DTMF Begin"); 00830 subclass[0] = f->subclass; 00831 subclass[1] = '\0'; 00832 break; 00833 case AST_FRAME_DTMF_END: 00834 strcpy(ftype, "DTMF End"); 00835 subclass[0] = f->subclass; 00836 subclass[1] = '\0'; 00837 break; 00838 case AST_FRAME_CONTROL: 00839 strcpy(ftype, "Control"); 00840 switch(f->subclass) { 00841 case AST_CONTROL_HANGUP: 00842 strcpy(subclass, "Hangup"); 00843 break; 00844 case AST_CONTROL_RING: 00845 strcpy(subclass, "Ring"); 00846 break; 00847 case AST_CONTROL_RINGING: 00848 strcpy(subclass, "Ringing"); 00849 break; 00850 case AST_CONTROL_ANSWER: 00851 strcpy(subclass, "Answer"); 00852 break; 00853 case AST_CONTROL_BUSY: 00854 strcpy(subclass, "Busy"); 00855 break; 00856 case AST_CONTROL_TAKEOFFHOOK: 00857 strcpy(subclass, "Take Off Hook"); 00858 break; 00859 case AST_CONTROL_OFFHOOK: 00860 strcpy(subclass, "Line Off Hook"); 00861 break; 00862 case AST_CONTROL_CONGESTION: 00863 strcpy(subclass, "Congestion"); 00864 break; 00865 case AST_CONTROL_FLASH: 00866 strcpy(subclass, "Flash"); 00867 break; 00868 case AST_CONTROL_WINK: 00869 strcpy(subclass, "Wink"); 00870 break; 00871 case AST_CONTROL_OPTION: 00872 strcpy(subclass, "Option"); 00873 break; 00874 case AST_CONTROL_RADIO_KEY: 00875 strcpy(subclass, "Key Radio"); 00876 break; 00877 case AST_CONTROL_RADIO_UNKEY: 00878 strcpy(subclass, "Unkey Radio"); 00879 break; 00880 case AST_CONTROL_PROGRESS: 00881 strcpy(subclass, "Call Progress"); 00882 break; 00883 case AST_CONTROL_PROCEEDING: 00884 strcpy(subclass, "Proceeding"); 00885 break; 00886 case AST_CONTROL_HOLD: 00887 strcpy(subclass, "Hold"); 00888 break; 00889 case AST_CONTROL_UNHOLD: 00890 strcpy(subclass, "UnHold"); 00891 break; 00892 case -1: 00893 strcpy(subclass, "Stop generators"); 00894 break; 00895 default: 00896 snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass); 00897 } 00898 break; 00899 case AST_FRAME_NULL: 00900 strcpy(ftype, "Null Frame"); 00901 strcpy(subclass, "N/A"); 00902 break; 00903 case AST_FRAME_IAX: 00904 /* Should never happen */ 00905 strcpy(ftype, "IAX Specific"); 00906 snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass); 00907 break; 00908 case AST_FRAME_TEXT: 00909 strcpy(ftype, "Text"); 00910 strcpy(subclass, "N/A"); 00911 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00912 break; 00913 case AST_FRAME_IMAGE: 00914 strcpy(ftype, "Image"); 00915 snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass)); 00916 break; 00917 case AST_FRAME_HTML: 00918 strcpy(ftype, "HTML"); 00919 switch(f->subclass) { 00920 case AST_HTML_URL: 00921 strcpy(subclass, "URL"); 00922 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00923 break; 00924 case AST_HTML_DATA: 00925 strcpy(subclass, "Data"); 00926 break; 00927 case AST_HTML_BEGIN: 00928 strcpy(subclass, "Begin"); 00929 break; 00930 case AST_HTML_END: 00931 strcpy(subclass, "End"); 00932 break; 00933 case AST_HTML_LDCOMPLETE: 00934 strcpy(subclass, "Load Complete"); 00935 break; 00936 case AST_HTML_NOSUPPORT: 00937 strcpy(subclass, "No Support"); 00938 break; 00939 case AST_HTML_LINKURL: 00940 strcpy(subclass, "Link URL"); 00941 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00942 break; 00943 case AST_HTML_UNLINK: 00944 strcpy(subclass, "Unlink"); 00945 break; 00946 case AST_HTML_LINKREJECT: 00947 strcpy(subclass, "Link Reject"); 00948 break; 00949 default: 00950 snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass); 00951 break; 00952 } 00953 break; 00954 case AST_FRAME_MODEM: 00955 strcpy(ftype, "Modem"); 00956 switch (f->subclass) { 00957 case AST_MODEM_T38: 00958 strcpy(subclass, "T.38"); 00959 break; 00960 case AST_MODEM_V150: 00961 strcpy(subclass, "V.150"); 00962 break; 00963 default: 00964 snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass); 00965 break; 00966 } 00967 break; 00968 default: 00969 snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype); 00970 } 00971 if (!ast_strlen_zero(moreinfo)) 00972 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n", 00973 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00974 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00975 f->frametype, 00976 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00977 f->subclass, 00978 term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)), 00979 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00980 else 00981 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n", 00982 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00983 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00984 f->frametype, 00985 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00986 f->subclass, 00987 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00988 }
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 or list of frames.
fr | Frame to free, or head of list to free | |
cache | Whether to consider this frame for frame caching |
Definition at line 377 of file frame.c.
References __frame_free(), AST_LIST_NEXT, and ast_frame::next.
Referenced by mixmonitor_thread().
00378 { 00379 struct ast_frame *next; 00380 00381 for (next = AST_LIST_NEXT(frame, frame_list); 00382 frame; 00383 frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) { 00384 __frame_free(frame, cache); 00385 } 00386 }
Sums two frames of audio samples.
f1 | The first frame (which will contain the result) | |
f2 | The second frame |
Definition at line 1603 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.
01604 { 01605 int count; 01606 short *data1, *data2; 01607 01608 if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR)) 01609 return -1; 01610 01611 if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR)) 01612 return -1; 01613 01614 if (f1->samples != f2->samples) 01615 return -1; 01616 01617 for (count = 0, data1 = f1->data, data2 = f2->data; 01618 count < f1->samples; 01619 count++, data1++, data2++) 01620 ast_slinear_saturated_add(data1, data2); 01621 01622 return 0; 01623 }
Copies a frame.
fr | frame to copy Duplicates a frame -- should only rarely be used, typically frisolate is good enough |
Definition at line 471 of file frame.c.
References ast_calloc_cache, ast_copy_flags, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, AST_LIST_REMOVE_CURRENT, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, AST_MALLOCD_HDR, ast_threadstorage_get(), ast_frame::data, ast_frame::datalen, ast_frame::delivery, f, frame_cache, frames, ast_frame::frametype, ast_frame::len, len(), ast_frame::mallocd, ast_frame::mallocd_hdr_len, ast_frame::offset, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, and ast_frame::ts.
Referenced by __ast_queue_frame(), ast_frisolate(), ast_jb_put(), ast_rtp_write(), ast_slinfactory_feed(), audiohook_read_frame_single(), autoservice_run(), process_rfc2833(), recordthread(), and rpt().
00472 { 00473 struct ast_frame *out = NULL; 00474 int len, srclen = 0; 00475 void *buf = NULL; 00476 00477 #if !defined(LOW_MEMORY) 00478 struct ast_frame_cache *frames; 00479 #endif 00480 00481 /* Start with standard stuff */ 00482 len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00483 /* If we have a source, add space for it */ 00484 /* 00485 * XXX Watch out here - if we receive a src which is not terminated 00486 * properly, we can be easily attacked. Should limit the size we deal with. 00487 */ 00488 if (f->src) 00489 srclen = strlen(f->src); 00490 if (srclen > 0) 00491 len += srclen + 1; 00492 00493 #if !defined(LOW_MEMORY) 00494 if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) { 00495 AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) { 00496 if (out->mallocd_hdr_len >= len) { 00497 size_t mallocd_len = out->mallocd_hdr_len; 00498 AST_LIST_REMOVE_CURRENT(&frames->list, frame_list); 00499 memset(out, 0, sizeof(*out)); 00500 out->mallocd_hdr_len = mallocd_len; 00501 buf = out; 00502 frames->size--; 00503 break; 00504 } 00505 } 00506 AST_LIST_TRAVERSE_SAFE_END; 00507 } 00508 #endif 00509 00510 if (!buf) { 00511 if (!(buf = ast_calloc_cache(1, len))) 00512 return NULL; 00513 out = buf; 00514 out->mallocd_hdr_len = len; 00515 } 00516 00517 out->frametype = f->frametype; 00518 out->subclass = f->subclass; 00519 out->datalen = f->datalen; 00520 out->samples = f->samples; 00521 out->delivery = f->delivery; 00522 /* Set us as having malloc'd header only, so it will eventually 00523 get freed. */ 00524 out->mallocd = AST_MALLOCD_HDR; 00525 out->offset = AST_FRIENDLY_OFFSET; 00526 if (out->datalen) { 00527 out->data = buf + sizeof(*out) + AST_FRIENDLY_OFFSET; 00528 memcpy(out->data, f->data, out->datalen); 00529 } 00530 if (srclen > 0) { 00531 /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */ 00532 char *src; 00533 out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00534 src = (char *) out->src; 00535 /* Must have space since we allocated for it */ 00536 strcpy(src, f->src); 00537 } 00538 ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO); 00539 out->ts = f->ts; 00540 out->len = f->len; 00541 out->seqno = f->seqno; 00542 return out; 00543 }
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 393 of file frame.c.
References ast_copy_flags, ast_frame_header_new(), ast_frdup(), AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, ast_malloc, AST_MALLOCD_DATA, AST_MALLOCD_HDR, AST_MALLOCD_SRC, ast_strdup, ast_test_flag, ast_frame::data, ast_frame::datalen, ast_frame::frametype, free, ast_frame::len, ast_frame::mallocd, ast_frame::offset, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, and ast_frame::ts.
Referenced by ast_dsp_process(), ast_rtp_read(), ast_safe_sleep_conditional(), ast_slinfactory_feed(), ast_trans_frameout(), ast_write(), autoservice_run(), dahdi_decoder_frameout(), dahdi_encoder_frameout(), feature_request_and_dial(), jpeg_read_image(), and read_frame().
00394 { 00395 struct ast_frame *out; 00396 void *newdata; 00397 00398 /* if none of the existing frame is malloc'd, let ast_frdup() do it 00399 since it is more efficient 00400 */ 00401 if (fr->mallocd == 0) { 00402 return ast_frdup(fr); 00403 } 00404 00405 /* if everything is already malloc'd, we are done */ 00406 if ((fr->mallocd & (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) == 00407 (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) { 00408 return fr; 00409 } 00410 00411 if (!(fr->mallocd & AST_MALLOCD_HDR)) { 00412 /* Allocate a new header if needed */ 00413 if (!(out = ast_frame_header_new())) { 00414 return NULL; 00415 } 00416 out->frametype = fr->frametype; 00417 out->subclass = fr->subclass; 00418 out->datalen = fr->datalen; 00419 out->samples = fr->samples; 00420 out->offset = fr->offset; 00421 /* Copy the timing data */ 00422 ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO); 00423 if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) { 00424 out->ts = fr->ts; 00425 out->len = fr->len; 00426 out->seqno = fr->seqno; 00427 } 00428 } else { 00429 out = fr; 00430 } 00431 00432 if (!(fr->mallocd & AST_MALLOCD_SRC) && fr->src) { 00433 if (!(out->src = ast_strdup(fr->src))) { 00434 if (out != fr) { 00435 free(out); 00436 } 00437 return NULL; 00438 } 00439 } else { 00440 out->src = fr->src; 00441 fr->src = NULL; 00442 fr->mallocd &= ~AST_MALLOCD_SRC; 00443 } 00444 00445 if (!(fr->mallocd & AST_MALLOCD_DATA)) { 00446 if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) { 00447 if (out->src != fr->src) { 00448 free((void *) out->src); 00449 } 00450 if (out != fr) { 00451 free(out); 00452 } 00453 return NULL; 00454 } 00455 newdata += AST_FRIENDLY_OFFSET; 00456 out->offset = AST_FRIENDLY_OFFSET; 00457 out->datalen = fr->datalen; 00458 memcpy(newdata, fr->data, fr->datalen); 00459 out->data = newdata; 00460 } else { 00461 out->data = fr->data; 00462 fr->data = NULL; 00463 fr->mallocd &= ~AST_MALLOCD_DATA; 00464 } 00465 00466 out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA; 00467 00468 return out; 00469 }
struct ast_format_list* ast_get_format_list | ( | size_t * | size | ) |
Definition at line 561 of file frame.c.
References AST_FORMAT_LIST.
00562 { 00563 *size = (sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0])); 00564 return AST_FORMAT_LIST; 00565 }
struct ast_format_list* ast_get_format_list_index | ( | int | index | ) |
Definition at line 556 of file frame.c.
References AST_FORMAT_LIST.
00557 { 00558 return &AST_FORMAT_LIST[index]; 00559 }
int ast_getformatbyname | ( | const char * | name | ) |
Gets a format from a name.
name | string of format |
Definition at line 627 of file frame.c.
References ast_expand_codec_alias(), AST_FORMAT_LIST, and format.
Referenced by ast_parse_allow_disallow(), iax_template_parse(), reload_config(), and try_suggested_sip_codec().
00628 { 00629 int x, all, format = 0; 00630 00631 all = strcasecmp(name, "all") ? 0 : 1; 00632 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00633 if(AST_FORMAT_LIST[x].visible && (all || 00634 !strcasecmp(AST_FORMAT_LIST[x].name,name) || 00635 !strcasecmp(AST_FORMAT_LIST[x].name,ast_expand_codec_alias(name)))) { 00636 format |= AST_FORMAT_LIST[x].bits; 00637 if(!all) 00638 break; 00639 } 00640 } 00641 00642 return format; 00643 }
char* ast_getformatname | ( | int | format | ) |
Get the name of a format.
format | id of format |
Definition at line 567 of file frame.c.
References AST_FORMAT_LIST, ast_format_list::bits, name, and ast_format_list::visible.
Referenced by __ast_play_and_record(), __ast_read(), __ast_register_translator(), __login_exec(), _sip_show_peer(), add_codec_to_answer(), add_codec_to_sdp(), agent_call(), ast_codec_get_len(), ast_codec_get_samples(), ast_codec_pref_string(), ast_dsp_process(), ast_frame_dump(), ast_openvstream(), ast_rtp_write(), ast_slinfactory_feed(), ast_streamfile(), ast_translator_build_path(), ast_unregister_translator(), ast_writestream(), background_detect_exec(), dahdi_read(), do_waiting(), eagi_exec(), func_channel_read(), function_iaxpeer(), function_sippeer(), gtalk_show_channels(), iax2_request(), iax2_show_channels(), iax2_show_peer(), iax_show_provisioning(), moh_classes_show(), moh_release(), oh323_rtp_read(), phone_setup(), print_codec_to_cli(), rebuild_matrix(), register_translator(), set_format(), set_local_capabilities(), set_peer_capabilities(), show_codecs(), show_codecs_deprecated(), show_file_formats(), show_file_formats_deprecated(), show_image_formats(), show_image_formats_deprecated(), show_translation(), show_translation_deprecated(), sip_request_call(), sip_rtp_read(), and socket_process().
00568 { 00569 int x; 00570 char *ret = "unknown"; 00571 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00572 if(AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == format) { 00573 ret = AST_FORMAT_LIST[x].name; 00574 break; 00575 } 00576 } 00577 return ret; 00578 }
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 580 of file frame.c.
References AST_FORMAT_LIST, ast_format_list::bits, len(), name, and ast_format_list::visible.
Referenced by __ast_read(), __sip_show_channels(), _sip_show_peer(), add_sdp(), ast_streamfile(), function_iaxpeer(), function_sippeer(), gtalk_is_answered(), gtalk_newcall(), handle_showchan(), handle_showchan_deprecated(), iax2_show_peer(), process_sdp(), serialize_showchan(), set_format(), sip_new(), sip_request_call(), sip_show_channel(), sip_show_settings(), and sip_write().
00581 { 00582 int x; 00583 unsigned len; 00584 char *start, *end = buf; 00585 00586 if (!size) 00587 return buf; 00588 snprintf(end, size, "0x%x (", format); 00589 len = strlen(end); 00590 end += len; 00591 size -= len; 00592 start = end; 00593 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00594 if (AST_FORMAT_LIST[x].visible && (AST_FORMAT_LIST[x].bits & format)) { 00595 snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name); 00596 len = strlen(end); 00597 end += len; 00598 size -= len; 00599 } 00600 } 00601 if (start == end) 00602 snprintf(start, size, "nothing)"); 00603 else if (size > 1) 00604 *(end -1) = ')'; 00605 return buf; 00606 }
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 1307 of file frame.c.
References ast_codec_pref_append(), ast_codec_pref_remove(), ast_codec_pref_setsize(), AST_FORMAT_AUDIO_MASK, ast_getformatbyname(), ast_log(), ast_strdupa, format, LOG_DEBUG, LOG_WARNING, option_debug, and parse().
Referenced by action_originate(), apply_outgoing(), build_device(), build_peer(), build_user(), gtalk_create_member(), gtalk_load_config(), reload_config(), set_config(), and update_common_options().
01308 { 01309 char *parse = NULL, *this = NULL, *psize = NULL; 01310 int format = 0, framems = 0; 01311 01312 parse = ast_strdupa(list); 01313 while ((this = strsep(&parse, ","))) { 01314 framems = 0; 01315 if ((psize = strrchr(this, ':'))) { 01316 *psize++ = '\0'; 01317 if (option_debug) 01318 ast_log(LOG_DEBUG,"Packetization for codec: %s is %s\n", this, psize); 01319 framems = atoi(psize); 01320 if (framems < 0) 01321 framems = 0; 01322 } 01323 if (!(format = ast_getformatbyname(this))) { 01324 ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this); 01325 continue; 01326 } 01327 01328 if (mask) { 01329 if (allowing) 01330 *mask |= format; 01331 else 01332 *mask &= ~format; 01333 } 01334 01335 /* Set up a preference list for audio. Do not include video in preferences 01336 since we can not transcode video and have to use whatever is offered 01337 */ 01338 if (pref && (format & AST_FORMAT_AUDIO_MASK)) { 01339 if (strcasecmp(this, "all")) { 01340 if (allowing) { 01341 ast_codec_pref_append(pref, format); 01342 ast_codec_pref_setsize(pref, format, framems); 01343 } 01344 else 01345 ast_codec_pref_remove(pref, format); 01346 } else if (!allowing) { 01347 memset(pref, 0, sizeof(*pref)); 01348 } 01349 } 01350 } 01351 }
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 186 of file frame.c.
References ast_malloc, ast_smoother_reset(), and s.
Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().
00187 { 00188 struct ast_smoother *s; 00189 if (size < 1) 00190 return NULL; 00191 if ((s = ast_malloc(sizeof(*s)))) 00192 ast_smoother_reset(s, size); 00193 return s; 00194 }
struct ast_frame* ast_smoother_read | ( | struct ast_smoother * | s | ) |
Definition at line 246 of file frame.c.
References AST_FRAME_VOICE, AST_FRIENDLY_OFFSET, ast_log(), ast_samp2tv(), AST_SMOOTHER_FLAG_G729, ast_tvadd(), ast_tvzero(), len(), LOG_WARNING, and s.
Referenced by ast_rtp_write().
00247 { 00248 struct ast_frame *opt; 00249 int len; 00250 00251 /* IF we have an optimization frame, send it */ 00252 if (s->opt) { 00253 if (s->opt->offset < AST_FRIENDLY_OFFSET) 00254 ast_log(LOG_WARNING, "Returning a frame of inappropriate offset (%d).\n", 00255 s->opt->offset); 00256 opt = s->opt; 00257 s->opt = NULL; 00258 return opt; 00259 } 00260 00261 /* Make sure we have enough data */ 00262 if (s->len < s->size) { 00263 /* Or, if this is a G.729 frame with VAD on it, send it immediately anyway */ 00264 if (!((s->flags & AST_SMOOTHER_FLAG_G729) && (s->len % 10))) 00265 return NULL; 00266 } 00267 len = s->size; 00268 if (len > s->len) 00269 len = s->len; 00270 /* Make frame */ 00271 s->f.frametype = AST_FRAME_VOICE; 00272 s->f.subclass = s->format; 00273 s->f.data = s->framedata + AST_FRIENDLY_OFFSET; 00274 s->f.offset = AST_FRIENDLY_OFFSET; 00275 s->f.datalen = len; 00276 /* Samples will be improper given VAD, but with VAD the concept really doesn't even exist */ 00277 s->f.samples = len * s->samplesperbyte; /* XXX rounding */ 00278 s->f.delivery = s->delivery; 00279 /* Fill Data */ 00280 memcpy(s->f.data, s->data, len); 00281 s->len -= len; 00282 /* Move remaining data to the front if applicable */ 00283 if (s->len) { 00284 /* In principle this should all be fine because if we are sending 00285 G.729 VAD, the next timestamp will take over anyawy */ 00286 memmove(s->data, s->data + len, s->len); 00287 if (!ast_tvzero(s->delivery)) { 00288 /* If we have delivery time, increment it, otherwise, leave it at 0 */ 00289 s->delivery = ast_tvadd(s->delivery, ast_samp2tv(s->f.samples, 8000)); 00290 } 00291 } 00292 /* Return frame */ 00293 return &s->f; 00294 }
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 164 of file frame.c.
References s, and smoother_frame_feed().
Referenced by ast_rtp_codec_setpref().
00165 { 00166 /* if there is no change, then nothing to do */ 00167 if (s->size == bytes) { 00168 return; 00169 } 00170 /* set the new desired output size */ 00171 s->size = bytes; 00172 /* if there is no 'optimized' frame in the smoother, 00173 * then there is nothing left to do 00174 */ 00175 if (!s->opt) { 00176 return; 00177 } 00178 /* there is an 'optimized' frame here at the old size, 00179 * but it must now be put into the buffer so the data 00180 * can be extracted at the new size 00181 */ 00182 smoother_frame_feed(s, s->opt, s->opt_needs_swap); 00183 s->opt = NULL; 00184 }
void ast_smoother_reset | ( | struct ast_smoother * | s, | |
int | bytes | |||
) |
Definition at line 158 of file frame.c.
References s.
Referenced by ast_smoother_new().
00159 { 00160 memset(s, 0, sizeof(*s)); 00161 s->size = bytes; 00162 }
void ast_smoother_set_flags | ( | struct ast_smoother * | smoother, | |
int | flags | |||
) |
Definition at line 201 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 206 of file frame.c.
References s.
Referenced by ast_rtp_write().
00207 { 00208 return (s->flags & flag); 00209 }
void ast_swapcopy_samples | ( | void * | dst, | |
const void * | src, | |||
int | samples | |||
) |
Definition at line 545 of file frame.c.
Referenced by __ast_smoother_feed(), iax_frame_wrap(), phone_write_buf(), and smoother_frame_feed().
00546 { 00547 int i; 00548 unsigned short *dst_s = dst; 00549 const unsigned short *src_s = src; 00550 00551 for (i = 0; i < samples; i++) 00552 dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8); 00553 }
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 134 of file frame.c.
Referenced by __ast_read(), __oh323_rtp_create(), __oh323_update_info(), agent_read(), agent_request(), ast_channel_masquerade(), ast_channel_setwhentohangup(), ast_do_masquerade(), ast_rtcp_read(), ast_rtp_read(), ast_softhangup_nolock(), ast_udptl_read(), conf_run(), create_dtmf_frame(), dahdi_handle_event(), features_read(), gtalk_rtp_read(), handle_request_invite(), handle_response_invite(), local_read(), mgcp_rtp_read(), oh323_read(), oh323_rtp_read(), process_sdp(), sip_read(), sip_rtp_read(), skinny_rtp_read(), and wakeup_sub().