#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_ECHOCAN 8 |
#define | AST_OPTION_FLAG_ACCEPT 1 |
#define | AST_OPTION_FLAG_ANSWER 5 |
#define | AST_OPTION_FLAG_QUERY 4 |
#define | AST_OPTION_FLAG_REJECT 2 |
#define | AST_OPTION_FLAG_REQUEST 0 |
#define | AST_OPTION_FLAG_WTF 6 |
#define | AST_OPTION_OPRMODE 7 |
#define | AST_OPTION_RELAXDTMF 3 |
#define | AST_OPTION_RXGAIN 6 |
#define | AST_OPTION_TDD 2 |
#define | AST_OPTION_TONE_VERIFY 1 |
#define | AST_OPTION_TXGAIN 5 |
#define | ast_smoother_feed(s, f) __ast_smoother_feed(s, f, 0) |
#define | ast_smoother_feed_be(s, f) __ast_smoother_feed(s, f, 1) |
#define | ast_smoother_feed_le(s, f) __ast_smoother_feed(s, f, 0) |
#define | AST_SMOOTHER_FLAG_BE (1 << 1) |
#define | AST_SMOOTHER_FLAG_G729 (1 << 0) |
Enumerations | |
enum | { AST_FRFLAG_HAS_TIMING_INFO = (1 << 0), AST_FRFLAG_FROM_TRANSLATOR = (1 << 1), AST_FRFLAG_FROM_DSP = (1 << 2), AST_FRFLAG_FROM_FILESTREAM = (1 << 3) } |
enum | ast_control_frame_type { AST_CONTROL_HANGUP = 1, AST_CONTROL_RING = 2, AST_CONTROL_RINGING = 3, AST_CONTROL_ANSWER = 4, AST_CONTROL_BUSY = 5, AST_CONTROL_TAKEOFFHOOK = 6, AST_CONTROL_OFFHOOK = 7, AST_CONTROL_CONGESTION = 8, AST_CONTROL_FLASH = 9, AST_CONTROL_WINK = 10, AST_CONTROL_OPTION = 11, AST_CONTROL_RADIO_KEY = 12, AST_CONTROL_RADIO_UNKEY = 13, AST_CONTROL_PROGRESS = 14, AST_CONTROL_PROCEEDING = 15, AST_CONTROL_HOLD = 16, AST_CONTROL_UNHOLD = 17, AST_CONTROL_VIDUPDATE = 18, AST_CONTROL_ATXFERCMD = 19, AST_CONTROL_SRCUPDATE = 20 } |
enum | ast_frame_type { AST_FRAME_DTMF_END = 1, AST_FRAME_VOICE, AST_FRAME_VIDEO, AST_FRAME_CONTROL, AST_FRAME_NULL, AST_FRAME_IAX, AST_FRAME_TEXT, AST_FRAME_IMAGE, AST_FRAME_HTML, AST_FRAME_CNG, AST_FRAME_MODEM, AST_FRAME_DTMF_BEGIN } |
Frame types. More... | |
Functions | |
int | __ast_smoother_feed (struct ast_smoother *s, struct ast_frame *f, int swap) |
char * | ast_codec2str (int codec) |
Get a name from a format Gets a name from a format. | |
int | ast_codec_choose (struct ast_codec_pref *pref, int formats, int find_best) |
Select the best audio format according to preference list from supplied options. If "find_best" is non-zero then if nothing is found, the "Best" format of the format list is selected, otherwise 0 is returned. | |
int | ast_codec_get_len (int format, int samples) |
Returns the number of bytes for the number of samples of the given format. | |
int | ast_codec_get_samples (struct ast_frame *f) |
Returns the number of samples contained in the frame. | |
static int | ast_codec_interp_len (int format) |
Gets duration in ms of interpolation frame for a format. | |
int | ast_codec_pref_append (struct ast_codec_pref *pref, int format) |
Append a audio codec to a preference list, removing it first if it was already there. | |
void | ast_codec_pref_convert (struct ast_codec_pref *pref, char *buf, size_t size, int right) |
Shift an audio codec preference list up or down 65 bytes so that it becomes an ASCII string. | |
ast_format_list | ast_codec_pref_getsize (struct ast_codec_pref *pref, int format) |
Get packet size for codec. | |
int | ast_codec_pref_index (struct ast_codec_pref *pref, int index) |
Codec located at a particular place in the preference index See Audio Codec Preferences. | |
void | ast_codec_pref_init (struct ast_codec_pref *pref) |
Initialize an audio codec preference to "no preference" See Audio Codec Preferences. | |
void | ast_codec_pref_prepend (struct ast_codec_pref *pref, int format, int only_if_existing) |
Prepend an audio codec to a preference list, removing it first if it was already there. | |
void | ast_codec_pref_remove (struct ast_codec_pref *pref, int format) |
Remove audio a codec from a preference list. | |
int | ast_codec_pref_setsize (struct ast_codec_pref *pref, int format, int framems) |
Set packet size for codec. | |
int | ast_codec_pref_string (struct ast_codec_pref *pref, char *buf, size_t size) |
Dump audio codec preference list into a string. | |
static force_inline int | ast_format_rate (int format) |
Get the sample rate for a given format. | |
int | ast_frame_adjust_volume (struct ast_frame *f, int adjustment) |
Adjusts the volume of the audio samples contained in a frame. | |
void | ast_frame_dump (const char *name, struct ast_frame *f, char *prefix) |
ast_frame * | ast_frame_enqueue (struct ast_frame *head, struct ast_frame *f, int maxlen, int dupe) |
Appends a frame to the end of a list of frames, truncating the maximum length of the list. | |
void | ast_frame_free (struct ast_frame *fr, int cache) |
Requests a frame to be allocated Frees a frame 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 248 of file frame.h.
Referenced by adpcmtolin_sample(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), convertcap(), vox_read(), and vox_write().
#define AST_FORMAT_ALAW (1 << 3) |
Raw A-law data (G.711)
Definition at line 244 of file frame.h.
Referenced by alawtolin_sample(), alawtoulaw_sample(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), cb_events(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_new(), dahdi_read(), dahdi_write(), find_transcoders(), is_encoder(), misdn_read(), misdn_set_opt_exec(), oh323_rtp_read(), pcm_seek(), pcm_write(), read_config(), and sms_generate().
#define AST_FORMAT_AUDIO_MASK ((1 << 16)-1) |
Maximum audio mask
Definition at line 268 of file frame.h.
Referenced by add_sdp(), ast_best_codec(), ast_codec_choose(), ast_openstream_full(), ast_parse_allow_disallow(), ast_request(), ast_translate_available_formats(), ast_translator_best_choice(), begin_dial(), func_channel_read(), generator_force(), gtalk_rtp_read(), process_sdp(), set_format(), sip_call(), sip_rtp_read(), and sip_write().
#define AST_FORMAT_AUDIO_UNDEFINED ((1 << 13) | (1 << 14) | (1 << 15)) |
#define AST_FORMAT_G722 (1 << 12) |
G.722
Definition at line 262 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_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 238 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_samples(), ast_rtp_write(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_destroy(), dahdi_translate(), g723_read(), g723_write(), load_module(), phone_request(), phone_setup(), phone_write(), and register_translator().
#define AST_FORMAT_G726 (1 << 11) |
ADPCM (G.726, 32kbps, RFC3551 codeword packing)
Definition at line 260 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_set_rtpmap_type(), g726_read(), g726_write(), and g726tolin_sample().
#define AST_FORMAT_G726_AAL2 (1 << 4) |
ADPCM (G.726, 32kbps, AAL2 codeword packing)
Definition at line 246 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_lookup_mime_subtype(), ast_rtp_set_rtpmap_type(), codec_ast2skinny(), and codec_skinny2ast().
#define AST_FORMAT_G729A (1 << 8) |
G.729A audio
Definition at line 254 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_destroy(), dahdi_translate(), g729_read(), and g729_write().
#define AST_FORMAT_GSM (1 << 1) |
GSM compression
Definition at line 240 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), convertcap(), gsm_read(), gsm_write(), gsmtolin_sample(), wav_read(), and wav_write().
#define AST_FORMAT_H261 (1 << 18) |
H.261 Video
Definition at line 274 of file frame.h.
Referenced by codec_ast2skinny(), and codec_skinny2ast().
#define AST_FORMAT_H263 (1 << 19) |
H.263 Video
Definition at line 276 of file frame.h.
Referenced by codec_ast2skinny(), codec_skinny2ast(), h263_read(), and h263_write().
#define AST_FORMAT_H264 (1 << 21) |
#define AST_FORMAT_ILBC (1 << 10) |
iLBC Free Compression
Definition at line 258 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_codec_interp_len(), convertcap(), ilbc_read(), ilbc_write(), and ilbctolin_sample().
#define AST_FORMAT_JPEG (1 << 16) |
JPEG Images
Definition at line 270 of file frame.h.
Referenced by jpeg_read_image(), and jpeg_write_image().
#define AST_FORMAT_LPC10 (1 << 7) |
LPC10, 180 samples/frame
Definition at line 252 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_samples(), and lpc10tolin_sample().
#define AST_FORMAT_MAX_AUDIO (1 << 15) |
Maximum audio format
Definition at line 266 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 282 of file frame.h.
Referenced by add_sdp(), ast_openvstream(), and ast_translate_available_formats().
#define AST_FORMAT_PNG (1 << 17) |
#define AST_FORMAT_SLINEAR (1 << 6) |
Raw 16-bit Signed Linear (8000 Hz) PCM
Definition at line 250 of file frame.h.
Referenced by __ast_play_and_record(), __ast_register_translator(), action_originate(), agent_new(), alsa_new(), alsa_read(), alsa_request(), ast_audiohook_read_frame(), ast_best_codec(), ast_channel_make_compatible(), ast_channel_start_silence_generator(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_call_progress(), ast_dsp_digitdetect(), ast_dsp_process(), ast_dsp_silence(), ast_frame_adjust_volume(), ast_frame_slinear_sum(), ast_rtp_read(), ast_slinfactory_feed(), attempt_reconnect(), audio_audiohook_write_list(), audiohook_read_frame_both(), audiohook_read_frame_single(), background_detect_exec(), build_conf(), chanspy_exec(), conf_run(), connect_link(), dahdi_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(), misdn_set_opt_exec(), mixmonitor_thread(), moh_class_malloc(), mp3_exec(), nbs_request(), nbs_xwrite(), NBScat_exec(), ogg_vorbis_read(), ogg_vorbis_write(), oh323_rtp_read(), orig_app(), orig_exten(), oss_new(), oss_read(), oss_request(), parkandannounce_exec(), phone_new(), phone_read(), phone_request(), phone_setup(), phone_write(), playtones_alloc(), read_config(), rpt(), rpt_call(), rpt_tele_thread(), send_waveform_to_channel(), silence_generator_generate(), slinear_read(), slinear_write(), sms_generate(), socket_process(), speech_background(), speech_create(), spy_generate(), tonepair_alloc(), wav_read(), and wav_write().
#define AST_FORMAT_SPEEX (1 << 9) |
SpeeX Free Compression
Definition at line 256 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_samples(), ast_rtp_write(), convertcap(), and speextolin_sample().
#define AST_FORMAT_ULAW (1 << 2) |
Raw mu-law data (G.711)
Definition at line 242 of file frame.h.
Referenced by __adsi_transmit_messages(), adsi_careful_send(), alarmreceiver_exec(), ast_adsi_transmit_message_full(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), codec_ast2skinny(), codec_skinny2ast(), conf_run(), convertcap(), dahdi_new(), dahdi_read(), dahdi_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 283 of file frame.h.
Referenced by add_sdp(), ast_request(), ast_translate_available_formats(), check_user_full(), create_addr_from_peer(), func_channel_read(), gtalk_new(), gtalk_rtp_read(), sip_new(), and sip_rtp_read().
#define ast_frame_byteswap_be | ( | fr | ) | do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data, __f->data, __f->samples); } while(0) |
#define ast_frame_byteswap_le | ( | fr | ) | do { ; } while(0) |
#define AST_FRAME_DTMF AST_FRAME_DTMF_END |
Definition at line 125 of file frame.h.
Referenced by __action_dialoffhook(), __adsi_transmit_messages(), __ast_play_and_record(), agent_ack_sleep(), app_exec(), ast_audiohook_write_list(), ast_bridge_call(), ast_dsp_process(), ast_feature_request_and_dial(), ast_jb_put(), background_detect_exec(), cb_events(), channel_spy(), conf_exec(), conf_run(), console_dial(), console_dial_deprecated(), dahdi_bridge(), dahdi_read(), dictate_exec(), disa_exec(), do_immediate_setup(), echo_exec(), gtalk_handle_dtmf(), handle_recordfile(), handle_request(), handle_request_info(), mgcp_rtp_read(), misdn_bridge(), mp3_exec(), NBScat_exec(), oh323_rtp_read(), phone_exception(), process_ast_dsp(), receive_dtmf_digits(), rpt(), rpt_call(), send_waveform_to_channel(), sip_rtp_read(), speech_background(), ss_thread(), wait_for_answer(), and wait_for_winner().
#define AST_FRAME_SET_BUFFER | ( | fr, | |||
_base, | |||||
_ofs, | |||||
_datalen | ) |
Value:
Set the various field of a frame to point to a buffer. Typically you set the base address of the buffer, the offset as AST_FRIENDLY_OFFSET, and the datalen as the amount of bytes queued. The remaining things (to be done manually) is set the number of samples, which cannot be derived from the datalen unless you know the number of bits per sample.Definition at line 187 of file frame.h.
Referenced by g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), ilbc_read(), ogg_vorbis_read(), pcm_read(), slinear_read(), vox_read(), and wav_read().
#define ast_frfree | ( | fr | ) | ast_frame_free(fr, 1) |
Definition at line 410 of file frame.h.
Referenced by __adsi_transmit_messages(), __ast_play_and_record(), __ast_queue_frame(), __ast_read(), __ast_request_and_dial(), adsi_careful_send(), agent_ack_sleep(), agent_read(), app_exec(), ast_audiohook_read_frame(), ast_autoservice_stop(), ast_bridge_call(), ast_channel_free(), ast_dsp_process(), ast_feature_request_and_dial(), ast_jb_destroy(), ast_jb_put(), ast_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_atxfer(), do_idle_thread(), do_parking_thread(), do_waiting(), echo_exec(), find_cache(), gen_generate(), handle_invite_replaces(), handle_recordfile(), iax_park_thread(), ices_exec(), isAnsweringMachine(), jb_empty_and_reset_adaptive(), jb_empty_and_reset_fixed(), jb_get_and_deliver(), masq_park_call(), measurenoise(), moh_files_generator(), monitor_dial(), mp3_exec(), NBScat_exec(), receive_dtmf_digits(), recordthread(), rpt(), run_agi(), send_tone_burst(), send_waveform_to_channel(), sendurl_exec(), speech_background(), spy_generate(), ss_thread(), wait_for_answer(), wait_for_hangup(), wait_for_winner(), waitforring_exec(), and waitstream_core().
#define AST_FRIENDLY_OFFSET 64 |
Definition at line 198 of file frame.h.
Referenced by __get_from_jb(), alsa_read(), ast_frdup(), ast_frisolate(), ast_prod(), ast_rtcp_read(), ast_rtp_read(), ast_smoother_read(), ast_trans_frameout(), ast_udptl_read(), conf_run(), dahdi_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 226 of file frame.h.
Referenced by ast_frame_dump(), and sendurl_exec().
#define AST_HTML_LINKREJECT 20 |
#define AST_HTML_LINKURL 18 |
#define AST_HTML_NOSUPPORT 17 |
Peer is unable to support HTML
Definition at line 228 of file frame.h.
Referenced by ast_frame_dump(), and sendurl_exec().
#define AST_HTML_UNLINK 19 |
#define AST_HTML_URL 1 |
Sending a URL
Definition at line 218 of file frame.h.
Referenced by ast_channel_sendurl(), and ast_frame_dump().
#define AST_MALLOCD_DATA (1 << 1) |
Need the data be free'd?
Definition at line 206 of file frame.h.
Referenced by __frame_free(), and ast_frisolate().
#define AST_MALLOCD_HDR (1 << 0) |
Need the header be free'd?
Definition at line 204 of file frame.h.
Referenced by __frame_free(), ast_frame_header_new(), ast_frdup(), and ast_frisolate().
#define AST_MALLOCD_SRC (1 << 2) |
Need the source be free'd? (haha!)
Definition at line 208 of file frame.h.
Referenced by __frame_free(), and ast_frisolate().
#define AST_MIN_OFFSET 32 |
#define AST_MODEM_T38 1 |
T.38 Fax-over-IP
Definition at line 212 of file frame.h.
Referenced by ast_frame_dump(), and udptl_rx_packet().
#define AST_MODEM_V150 2 |
#define AST_OPTION_AUDIO_MODE 4 |
Set (or clear) Audio (Not-Clear) Mode
Definition at line 330 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 352 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 327 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 346 of file frame.h.
Referenced by dahdi_setoption(), func_channel_write(), iax2_setoption(), play_record_review(), reset_volumes(), set_talk_volume(), and vm_forwardoptions().
#define AST_OPTION_TDD 2 |
Put a compatible channel into TDD (TTY for the hearing-impared) mode
Definition at line 324 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 321 of file frame.h.
Referenced by conf_run(), dahdi_hangup(), dahdi_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 338 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 308 of file frame.h.
Referenced by __ast_smoother_feed(), ast_smoother_read(), and smoother_frame_feed().
anonymous enum |
Definition at line 127 of file frame.h.
00127 { 00128 /*! This frame contains valid timing information */ 00129 AST_FRFLAG_HAS_TIMING_INFO = (1 << 0), 00130 /*! This frame came from a translator and is still the original frame. 00131 * The translator can not be free'd if the frame inside of it still has 00132 * this flag set. */ 00133 AST_FRFLAG_FROM_TRANSLATOR = (1 << 1), 00134 /*! This frame came from a dsp and is still the original frame. 00135 * The dsp cannot be free'd if the frame inside of it still has 00136 * this flag set. */ 00137 AST_FRFLAG_FROM_DSP = (1 << 2), 00138 /*! This frame came from a filestream and is still the original frame. 00139 * The filestream cannot be free'd if the frame inside of it still has 00140 * this flag set. */ 00141 AST_FRFLAG_FROM_FILESTREAM = (1 << 3), 00142 };
Definition at line 285 of file frame.h.
00285 { 00286 AST_CONTROL_HANGUP = 1, /*!< Other end has hungup */ 00287 AST_CONTROL_RING = 2, /*!< Local ring */ 00288 AST_CONTROL_RINGING = 3, /*!< Remote end is ringing */ 00289 AST_CONTROL_ANSWER = 4, /*!< Remote end has answered */ 00290 AST_CONTROL_BUSY = 5, /*!< Remote end is busy */ 00291 AST_CONTROL_TAKEOFFHOOK = 6, /*!< Make it go off hook */ 00292 AST_CONTROL_OFFHOOK = 7, /*!< Line is off hook */ 00293 AST_CONTROL_CONGESTION = 8, /*!< Congestion (circuits busy) */ 00294 AST_CONTROL_FLASH = 9, /*!< Flash hook */ 00295 AST_CONTROL_WINK = 10, /*!< Wink */ 00296 AST_CONTROL_OPTION = 11, /*!< Set a low-level option */ 00297 AST_CONTROL_RADIO_KEY = 12, /*!< Key Radio */ 00298 AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */ 00299 AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */ 00300 AST_CONTROL_PROCEEDING = 15, /*!< Indicate CALL PROCEEDING */ 00301 AST_CONTROL_HOLD = 16, /*!< Indicate call is placed on hold */ 00302 AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold */ 00303 AST_CONTROL_VIDUPDATE = 18, /*!< Indicate video frame update */ 00304 AST_CONTROL_ATXFERCMD = 19, /*!< AMI triggered attended transfer */ 00305 AST_CONTROL_SRCUPDATE = 20, /*!< Indicate source of media has changed */ 00306 };
enum ast_frame_type |
Frame types.
Definition at line 98 of file frame.h.
00098 { 00099 /*! DTMF end event, subclass is the digit */ 00100 AST_FRAME_DTMF_END = 1, 00101 /*! Voice data, subclass is AST_FORMAT_* */ 00102 AST_FRAME_VOICE, 00103 /*! Video frame, maybe?? :) */ 00104 AST_FRAME_VIDEO, 00105 /*! A control frame, subclass is AST_CONTROL_* */ 00106 AST_FRAME_CONTROL, 00107 /*! An empty, useless frame */ 00108 AST_FRAME_NULL, 00109 /*! Inter Asterisk Exchange private frame type */ 00110 AST_FRAME_IAX, 00111 /*! Text messages */ 00112 AST_FRAME_TEXT, 00113 /*! Image Frames */ 00114 AST_FRAME_IMAGE, 00115 /*! HTML Frame */ 00116 AST_FRAME_HTML, 00117 /*! Comfort Noise frame (subclass is level of CNG in -dBov), 00118 body may include zero or more 8-bit quantization coefficients */ 00119 AST_FRAME_CNG, 00120 /*! Modem-over-IP data streams */ 00121 AST_FRAME_MODEM, 00122 /*! DTMF begin event, subclass is the digit */ 00123 AST_FRAME_DTMF_BEGIN, 00124 };
int __ast_smoother_feed | ( | struct ast_smoother * | s, | |
struct ast_frame * | f, | |||
int | swap | |||
) |
Definition at line 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 656 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().
00657 { 00658 int x; 00659 char *ret = "unknown"; 00660 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00661 if(AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == codec) { 00662 ret = AST_FORMAT_LIST[x].desc; 00663 break; 00664 } 00665 } 00666 return ret; 00667 }
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 1295 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().
01296 { 01297 int x, ret = 0, slot; 01298 01299 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01300 slot = pref->order[x]; 01301 01302 if (!slot) 01303 break; 01304 if (formats & AST_FORMAT_LIST[slot-1].bits) { 01305 ret = AST_FORMAT_LIST[slot-1].bits; 01306 break; 01307 } 01308 } 01309 if(ret & AST_FORMAT_AUDIO_MASK) 01310 return ret; 01311 01312 if (option_debug > 3) 01313 ast_log(LOG_DEBUG, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec"); 01314 01315 return find_best ? ast_best_codec(formats) : 0; 01316 }
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 1554 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().
01555 { 01556 int len = 0; 01557 01558 /* XXX Still need speex, g723, and lpc10 XXX */ 01559 switch(format) { 01560 case AST_FORMAT_ILBC: 01561 len = (samples / 240) * 50; 01562 break; 01563 case AST_FORMAT_GSM: 01564 len = (samples / 160) * 33; 01565 break; 01566 case AST_FORMAT_G729A: 01567 len = samples / 8; 01568 break; 01569 case AST_FORMAT_SLINEAR: 01570 len = samples * 2; 01571 break; 01572 case AST_FORMAT_ULAW: 01573 case AST_FORMAT_ALAW: 01574 len = samples; 01575 break; 01576 case AST_FORMAT_G722: 01577 case AST_FORMAT_ADPCM: 01578 case AST_FORMAT_G726: 01579 case AST_FORMAT_G726_AAL2: 01580 len = samples / 2; 01581 break; 01582 default: 01583 ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format)); 01584 } 01585 01586 return len; 01587 }
int ast_codec_get_samples | ( | struct ast_frame * | f | ) |
Returns the number of samples contained in the frame.
Definition at line 1511 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().
01512 { 01513 int samples=0; 01514 switch(f->subclass) { 01515 case AST_FORMAT_SPEEX: 01516 samples = speex_samples(f->data, f->datalen); 01517 break; 01518 case AST_FORMAT_G723_1: 01519 samples = g723_samples(f->data, f->datalen); 01520 break; 01521 case AST_FORMAT_ILBC: 01522 samples = 240 * (f->datalen / 50); 01523 break; 01524 case AST_FORMAT_GSM: 01525 samples = 160 * (f->datalen / 33); 01526 break; 01527 case AST_FORMAT_G729A: 01528 samples = f->datalen * 8; 01529 break; 01530 case AST_FORMAT_SLINEAR: 01531 samples = f->datalen / 2; 01532 break; 01533 case AST_FORMAT_LPC10: 01534 /* assumes that the RTP packet contains one LPC10 frame */ 01535 samples = 22 * 8; 01536 samples += (((char *)(f->data))[7] & 0x1) * 8; 01537 break; 01538 case AST_FORMAT_ULAW: 01539 case AST_FORMAT_ALAW: 01540 samples = f->datalen; 01541 break; 01542 case AST_FORMAT_G722: 01543 case AST_FORMAT_ADPCM: 01544 case AST_FORMAT_G726: 01545 case AST_FORMAT_G726_AAL2: 01546 samples = f->datalen * 2; 01547 break; 01548 default: 01549 ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass)); 01550 } 01551 return samples; 01552 }
static int ast_codec_interp_len | ( | int | format | ) | [inline, static] |
Gets duration in ms of interpolation frame for a format.
Definition at line 576 of file frame.h.
References AST_FORMAT_ILBC.
Referenced by __get_from_jb(), and jb_get_and_deliver().
00577 { 00578 return (format == AST_FORMAT_ILBC) ? 30 : 20; 00579 }
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 1154 of file frame.c.
References ast_codec_pref_remove(), AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow().
01155 { 01156 int x, newindex = 0; 01157 01158 ast_codec_pref_remove(pref, format); 01159 01160 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01161 if(AST_FORMAT_LIST[x].bits == format) { 01162 newindex = x + 1; 01163 break; 01164 } 01165 } 01166 01167 if(newindex) { 01168 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01169 if(!pref->order[x]) { 01170 pref->order[x] = newindex; 01171 break; 01172 } 01173 } 01174 } 01175 01176 return x; 01177 }
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 1056 of file frame.c.
References ast_codec_pref::order.
Referenced by check_access(), create_addr(), dump_prefs(), and socket_process().
01057 { 01058 int x, differential = (int) 'A', mem; 01059 char *from, *to; 01060 01061 if(right) { 01062 from = pref->order; 01063 to = buf; 01064 mem = size; 01065 } else { 01066 to = pref->order; 01067 from = buf; 01068 mem = 32; 01069 } 01070 01071 memset(to, 0, mem); 01072 for (x = 0; x < 32 ; x++) { 01073 if(!from[x]) 01074 break; 01075 to[x] = right ? (from[x] + differential) : (from[x] - differential); 01076 } 01077 }
struct ast_format_list ast_codec_pref_getsize | ( | struct ast_codec_pref * | pref, | |
int | format | |||
) |
Get packet size for codec.
Definition at line 1256 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().
01257 { 01258 int x, index = -1, framems = 0; 01259 struct ast_format_list fmt = {0}; 01260 01261 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01262 if(AST_FORMAT_LIST[x].bits == format) { 01263 fmt = AST_FORMAT_LIST[x]; 01264 index = x; 01265 break; 01266 } 01267 } 01268 01269 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01270 if(pref->order[x] == (index + 1)) { 01271 framems = pref->framing[x]; 01272 break; 01273 } 01274 } 01275 01276 /* size validation */ 01277 if(!framems) 01278 framems = AST_FORMAT_LIST[index].def_ms; 01279 01280 if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */ 01281 framems -= framems % AST_FORMAT_LIST[index].inc_ms; 01282 01283 if(framems < AST_FORMAT_LIST[index].min_ms) 01284 framems = AST_FORMAT_LIST[index].min_ms; 01285 01286 if(framems > AST_FORMAT_LIST[index].max_ms) 01287 framems = AST_FORMAT_LIST[index].max_ms; 01288 01289 fmt.cur_ms = framems; 01290 01291 return fmt; 01292 }
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 1114 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().
01115 { 01116 int slot = 0; 01117 01118 01119 if((index >= 0) && (index < sizeof(pref->order))) { 01120 slot = pref->order[index]; 01121 } 01122 01123 return slot ? AST_FORMAT_LIST[slot-1].bits : 0; 01124 }
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 1180 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_codec_pref::framing, and ast_codec_pref::order.
Referenced by create_addr().
01181 { 01182 int x, newindex = 0; 01183 01184 /* First step is to get the codecs "index number" */ 01185 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01186 if (AST_FORMAT_LIST[x].bits == format) { 01187 newindex = x + 1; 01188 break; 01189 } 01190 } 01191 /* Done if its unknown */ 01192 if (!newindex) 01193 return; 01194 01195 /* Now find any existing occurrence, or the end */ 01196 for (x = 0; x < 32; x++) { 01197 if (!pref->order[x] || pref->order[x] == newindex) 01198 break; 01199 } 01200 01201 if (only_if_existing && !pref->order[x]) 01202 return; 01203 01204 /* Move down to make space to insert - either all the way to the end, 01205 or as far as the existing location (which will be overwritten) */ 01206 for (; x > 0; x--) { 01207 pref->order[x] = pref->order[x - 1]; 01208 pref->framing[x] = pref->framing[x - 1]; 01209 } 01210 01211 /* And insert the new entry */ 01212 pref->order[0] = newindex; 01213 pref->framing[0] = 0; /* ? */ 01214 }
void ast_codec_pref_remove | ( | struct ast_codec_pref * | pref, | |
int | format | |||
) |
Remove audio a codec from a preference list.
Definition at line 1127 of file frame.c.
References AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_codec_pref_append(), and ast_parse_allow_disallow().
01128 { 01129 struct ast_codec_pref oldorder; 01130 int x, y = 0; 01131 int slot; 01132 int size; 01133 01134 if(!pref->order[0]) 01135 return; 01136 01137 memcpy(&oldorder, pref, sizeof(oldorder)); 01138 memset(pref, 0, sizeof(*pref)); 01139 01140 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01141 slot = oldorder.order[x]; 01142 size = oldorder.framing[x]; 01143 if(! slot) 01144 break; 01145 if(AST_FORMAT_LIST[slot-1].bits != format) { 01146 pref->order[y] = slot; 01147 pref->framing[y++] = size; 01148 } 01149 } 01150 01151 }
int ast_codec_pref_setsize | ( | struct ast_codec_pref * | pref, | |
int | format, | |||
int | framems | |||
) |
Set packet size for codec.
Definition at line 1217 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().
01218 { 01219 int x, index = -1; 01220 01221 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01222 if(AST_FORMAT_LIST[x].bits == format) { 01223 index = x; 01224 break; 01225 } 01226 } 01227 01228 if(index < 0) 01229 return -1; 01230 01231 /* size validation */ 01232 if(!framems) 01233 framems = AST_FORMAT_LIST[index].def_ms; 01234 01235 if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */ 01236 framems -= framems % AST_FORMAT_LIST[index].inc_ms; 01237 01238 if(framems < AST_FORMAT_LIST[index].min_ms) 01239 framems = AST_FORMAT_LIST[index].min_ms; 01240 01241 if(framems > AST_FORMAT_LIST[index].max_ms) 01242 framems = AST_FORMAT_LIST[index].max_ms; 01243 01244 01245 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01246 if(pref->order[x] == (index + 1)) { 01247 pref->framing[x] = framems; 01248 break; 01249 } 01250 } 01251 01252 return x; 01253 }
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 1079 of file frame.c.
References ast_codec_pref_index(), and ast_getformatname().
Referenced by dump_prefs(), and socket_process().
01080 { 01081 int x, codec; 01082 size_t total_len, slen; 01083 char *formatname; 01084 01085 memset(buf,0,size); 01086 total_len = size; 01087 buf[0] = '('; 01088 total_len--; 01089 for(x = 0; x < 32 ; x++) { 01090 if(total_len <= 0) 01091 break; 01092 if(!(codec = ast_codec_pref_index(pref,x))) 01093 break; 01094 if((formatname = ast_getformatname(codec))) { 01095 slen = strlen(formatname); 01096 if(slen > total_len) 01097 break; 01098 strncat(buf, formatname, total_len - 1); /* safe */ 01099 total_len -= slen; 01100 } 01101 if(total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) { 01102 strncat(buf, "|", total_len - 1); /* safe */ 01103 total_len--; 01104 } 01105 } 01106 if(total_len) { 01107 strncat(buf, ")", total_len - 1); /* safe */ 01108 total_len--; 01109 } 01110 01111 return size - total_len; 01112 }
static force_inline int ast_format_rate | ( | int | format | ) | [static] |
Get the sample rate for a given format.
Definition at line 603 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().
00604 { 00605 if (format == AST_FORMAT_G722) 00606 return 16000; 00607 00608 return 8000; 00609 }
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 1589 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().
01590 { 01591 int count; 01592 short *fdata = f->data; 01593 short adjust_value = abs(adjustment); 01594 01595 if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR)) 01596 return -1; 01597 01598 if (!adjustment) 01599 return 0; 01600 01601 for (count = 0; count < f->samples; count++) { 01602 if (adjustment > 0) { 01603 ast_slinear_saturated_multiply(&fdata[count], &adjust_value); 01604 } else if (adjustment < 0) { 01605 ast_slinear_saturated_divide(&fdata[count], &adjust_value); 01606 } 01607 } 01608 01609 return 0; 01610 }
void ast_frame_dump | ( | const char * | name, | |
struct ast_frame * | f, | |||
char * | prefix | |||
) |
Dump a frame for debugging purposes
Definition at line 810 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().
00811 { 00812 const char noname[] = "unknown"; 00813 char ftype[40] = "Unknown Frametype"; 00814 char cft[80]; 00815 char subclass[40] = "Unknown Subclass"; 00816 char csub[80]; 00817 char moreinfo[40] = ""; 00818 char cn[60]; 00819 char cp[40]; 00820 char cmn[40]; 00821 00822 if (!name) 00823 name = noname; 00824 00825 00826 if (!f) { 00827 ast_verbose("%s [ %s (NULL) ] [%s]\n", 00828 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00829 term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00830 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00831 return; 00832 } 00833 /* XXX We should probably print one each of voice and video when the format changes XXX */ 00834 if (f->frametype == AST_FRAME_VOICE) 00835 return; 00836 if (f->frametype == AST_FRAME_VIDEO) 00837 return; 00838 switch(f->frametype) { 00839 case AST_FRAME_DTMF_BEGIN: 00840 strcpy(ftype, "DTMF Begin"); 00841 subclass[0] = f->subclass; 00842 subclass[1] = '\0'; 00843 break; 00844 case AST_FRAME_DTMF_END: 00845 strcpy(ftype, "DTMF End"); 00846 subclass[0] = f->subclass; 00847 subclass[1] = '\0'; 00848 break; 00849 case AST_FRAME_CONTROL: 00850 strcpy(ftype, "Control"); 00851 switch(f->subclass) { 00852 case AST_CONTROL_HANGUP: 00853 strcpy(subclass, "Hangup"); 00854 break; 00855 case AST_CONTROL_RING: 00856 strcpy(subclass, "Ring"); 00857 break; 00858 case AST_CONTROL_RINGING: 00859 strcpy(subclass, "Ringing"); 00860 break; 00861 case AST_CONTROL_ANSWER: 00862 strcpy(subclass, "Answer"); 00863 break; 00864 case AST_CONTROL_BUSY: 00865 strcpy(subclass, "Busy"); 00866 break; 00867 case AST_CONTROL_TAKEOFFHOOK: 00868 strcpy(subclass, "Take Off Hook"); 00869 break; 00870 case AST_CONTROL_OFFHOOK: 00871 strcpy(subclass, "Line Off Hook"); 00872 break; 00873 case AST_CONTROL_CONGESTION: 00874 strcpy(subclass, "Congestion"); 00875 break; 00876 case AST_CONTROL_FLASH: 00877 strcpy(subclass, "Flash"); 00878 break; 00879 case AST_CONTROL_WINK: 00880 strcpy(subclass, "Wink"); 00881 break; 00882 case AST_CONTROL_OPTION: 00883 strcpy(subclass, "Option"); 00884 break; 00885 case AST_CONTROL_RADIO_KEY: 00886 strcpy(subclass, "Key Radio"); 00887 break; 00888 case AST_CONTROL_RADIO_UNKEY: 00889 strcpy(subclass, "Unkey Radio"); 00890 break; 00891 case AST_CONTROL_PROGRESS: 00892 strcpy(subclass, "Call Progress"); 00893 break; 00894 case AST_CONTROL_PROCEEDING: 00895 strcpy(subclass, "Proceeding"); 00896 break; 00897 case AST_CONTROL_HOLD: 00898 strcpy(subclass, "Hold"); 00899 break; 00900 case AST_CONTROL_UNHOLD: 00901 strcpy(subclass, "UnHold"); 00902 break; 00903 case -1: 00904 strcpy(subclass, "Stop generators"); 00905 break; 00906 default: 00907 snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass); 00908 } 00909 break; 00910 case AST_FRAME_NULL: 00911 strcpy(ftype, "Null Frame"); 00912 strcpy(subclass, "N/A"); 00913 break; 00914 case AST_FRAME_IAX: 00915 /* Should never happen */ 00916 strcpy(ftype, "IAX Specific"); 00917 snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass); 00918 break; 00919 case AST_FRAME_TEXT: 00920 strcpy(ftype, "Text"); 00921 strcpy(subclass, "N/A"); 00922 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00923 break; 00924 case AST_FRAME_IMAGE: 00925 strcpy(ftype, "Image"); 00926 snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass)); 00927 break; 00928 case AST_FRAME_HTML: 00929 strcpy(ftype, "HTML"); 00930 switch(f->subclass) { 00931 case AST_HTML_URL: 00932 strcpy(subclass, "URL"); 00933 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00934 break; 00935 case AST_HTML_DATA: 00936 strcpy(subclass, "Data"); 00937 break; 00938 case AST_HTML_BEGIN: 00939 strcpy(subclass, "Begin"); 00940 break; 00941 case AST_HTML_END: 00942 strcpy(subclass, "End"); 00943 break; 00944 case AST_HTML_LDCOMPLETE: 00945 strcpy(subclass, "Load Complete"); 00946 break; 00947 case AST_HTML_NOSUPPORT: 00948 strcpy(subclass, "No Support"); 00949 break; 00950 case AST_HTML_LINKURL: 00951 strcpy(subclass, "Link URL"); 00952 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00953 break; 00954 case AST_HTML_UNLINK: 00955 strcpy(subclass, "Unlink"); 00956 break; 00957 case AST_HTML_LINKREJECT: 00958 strcpy(subclass, "Link Reject"); 00959 break; 00960 default: 00961 snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass); 00962 break; 00963 } 00964 break; 00965 case AST_FRAME_MODEM: 00966 strcpy(ftype, "Modem"); 00967 switch (f->subclass) { 00968 case AST_MODEM_T38: 00969 strcpy(subclass, "T.38"); 00970 break; 00971 case AST_MODEM_V150: 00972 strcpy(subclass, "V.150"); 00973 break; 00974 default: 00975 snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass); 00976 break; 00977 } 00978 break; 00979 default: 00980 snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype); 00981 } 00982 if (!ast_strlen_zero(moreinfo)) 00983 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n", 00984 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00985 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00986 f->frametype, 00987 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00988 f->subclass, 00989 term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)), 00990 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00991 else 00992 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n", 00993 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00994 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00995 f->frametype, 00996 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00997 f->subclass, 00998 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00999 }
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 385 of file frame.c.
References __frame_free(), AST_LIST_NEXT, ast_frame::frame_list, and ast_frame::next.
Referenced by mixmonitor_thread().
00386 { 00387 struct ast_frame *next; 00388 00389 for (next = AST_LIST_NEXT(frame, frame_list); 00390 frame; 00391 frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) { 00392 __frame_free(frame, cache); 00393 } 00394 }
Sums two frames of audio samples.
f1 | The first frame (which will contain the result) | |
f2 | The second frame |
Definition at line 1612 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.
01613 { 01614 int count; 01615 short *data1, *data2; 01616 01617 if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR)) 01618 return -1; 01619 01620 if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR)) 01621 return -1; 01622 01623 if (f1->samples != f2->samples) 01624 return -1; 01625 01626 for (count = 0, data1 = f1->data, data2 = f2->data; 01627 count < f1->samples; 01628 count++, data1++, data2++) 01629 ast_slinear_saturated_add(data1, data2); 01630 01631 return 0; 01632 }
Copies a frame.
fr | frame to copy Duplicates a frame -- should only rarely be used, typically frisolate is good enough |
Definition at line 482 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(), recordthread(), and rpt().
00483 { 00484 struct ast_frame *out = NULL; 00485 int len, srclen = 0; 00486 void *buf = NULL; 00487 00488 #if !defined(LOW_MEMORY) 00489 struct ast_frame_cache *frames; 00490 #endif 00491 00492 /* Start with standard stuff */ 00493 len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00494 /* If we have a source, add space for it */ 00495 /* 00496 * XXX Watch out here - if we receive a src which is not terminated 00497 * properly, we can be easily attacked. Should limit the size we deal with. 00498 */ 00499 if (f->src) 00500 srclen = strlen(f->src); 00501 if (srclen > 0) 00502 len += srclen + 1; 00503 00504 #if !defined(LOW_MEMORY) 00505 if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) { 00506 AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) { 00507 if (out->mallocd_hdr_len >= len) { 00508 size_t mallocd_len = out->mallocd_hdr_len; 00509 AST_LIST_REMOVE_CURRENT(&frames->list, frame_list); 00510 memset(out, 0, sizeof(*out)); 00511 out->mallocd_hdr_len = mallocd_len; 00512 buf = out; 00513 frames->size--; 00514 break; 00515 } 00516 } 00517 AST_LIST_TRAVERSE_SAFE_END; 00518 } 00519 #endif 00520 00521 if (!buf) { 00522 if (!(buf = ast_calloc_cache(1, len))) 00523 return NULL; 00524 out = buf; 00525 out->mallocd_hdr_len = len; 00526 } 00527 00528 out->frametype = f->frametype; 00529 out->subclass = f->subclass; 00530 out->datalen = f->datalen; 00531 out->samples = f->samples; 00532 out->delivery = f->delivery; 00533 /* Set us as having malloc'd header only, so it will eventually 00534 get freed. */ 00535 out->mallocd = AST_MALLOCD_HDR; 00536 out->offset = AST_FRIENDLY_OFFSET; 00537 if (out->datalen) { 00538 out->data = buf + sizeof(*out) + AST_FRIENDLY_OFFSET; 00539 memcpy(out->data, f->data, out->datalen); 00540 } 00541 if (srclen > 0) { 00542 /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */ 00543 char *src; 00544 out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00545 src = (char *) out->src; 00546 /* Must have space since we allocated for it */ 00547 strcpy(src, f->src); 00548 } 00549 ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO); 00550 out->ts = f->ts; 00551 out->len = f->len; 00552 out->seqno = f->seqno; 00553 return out; 00554 }
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 401 of file frame.c.
References ast_clear_flag, ast_copy_flags, ast_frame_header_new(), ast_frdup(), AST_FRFLAG_FROM_DSP, AST_FRFLAG_FROM_FILESTREAM, AST_FRFLAG_FROM_TRANSLATOR, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, ast_malloc, AST_MALLOCD_DATA, AST_MALLOCD_HDR, AST_MALLOCD_SRC, ast_strdup, ast_test_flag, ast_frame::data, ast_frame::datalen, ast_frame::frametype, free, ast_frame::len, ast_frame::mallocd, ast_frame::offset, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, and ast_frame::ts.
Referenced by ast_slinfactory_feed(), autoservice_run(), and jpeg_read_image().
00402 { 00403 struct ast_frame *out; 00404 void *newdata; 00405 00406 /* if none of the existing frame is malloc'd, let ast_frdup() do it 00407 since it is more efficient 00408 */ 00409 if (fr->mallocd == 0) { 00410 return ast_frdup(fr); 00411 } 00412 00413 /* if everything is already malloc'd, we are done */ 00414 if ((fr->mallocd & (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) == 00415 (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) { 00416 return fr; 00417 } 00418 00419 if (!(fr->mallocd & AST_MALLOCD_HDR)) { 00420 /* Allocate a new header if needed */ 00421 if (!(out = ast_frame_header_new())) { 00422 return NULL; 00423 } 00424 out->frametype = fr->frametype; 00425 out->subclass = fr->subclass; 00426 out->datalen = fr->datalen; 00427 out->samples = fr->samples; 00428 out->offset = fr->offset; 00429 /* Copy the timing data */ 00430 ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO); 00431 if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) { 00432 out->ts = fr->ts; 00433 out->len = fr->len; 00434 out->seqno = fr->seqno; 00435 } 00436 } else { 00437 ast_clear_flag(fr, AST_FRFLAG_FROM_TRANSLATOR); 00438 ast_clear_flag(fr, AST_FRFLAG_FROM_DSP); 00439 ast_clear_flag(fr, AST_FRFLAG_FROM_FILESTREAM); 00440 out = fr; 00441 } 00442 00443 if (!(fr->mallocd & AST_MALLOCD_SRC) && fr->src) { 00444 if (!(out->src = ast_strdup(fr->src))) { 00445 if (out != fr) { 00446 free(out); 00447 } 00448 return NULL; 00449 } 00450 } else { 00451 out->src = fr->src; 00452 fr->src = NULL; 00453 fr->mallocd &= ~AST_MALLOCD_SRC; 00454 } 00455 00456 if (!(fr->mallocd & AST_MALLOCD_DATA)) { 00457 if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) { 00458 if (out->src != fr->src) { 00459 free((void *) out->src); 00460 } 00461 if (out != fr) { 00462 free(out); 00463 } 00464 return NULL; 00465 } 00466 newdata += AST_FRIENDLY_OFFSET; 00467 out->offset = AST_FRIENDLY_OFFSET; 00468 out->datalen = fr->datalen; 00469 memcpy(newdata, fr->data, fr->datalen); 00470 out->data = newdata; 00471 } else { 00472 out->data = fr->data; 00473 fr->data = NULL; 00474 fr->mallocd &= ~AST_MALLOCD_DATA; 00475 } 00476 00477 out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA; 00478 00479 return out; 00480 }
struct ast_format_list* ast_get_format_list | ( | size_t * | size | ) |
Definition at line 572 of file frame.c.
References AST_FORMAT_LIST.
00573 { 00574 *size = (sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0])); 00575 return AST_FORMAT_LIST; 00576 }
struct ast_format_list* ast_get_format_list_index | ( | int | index | ) |
Definition at line 567 of file frame.c.
References AST_FORMAT_LIST.
00568 { 00569 return &AST_FORMAT_LIST[index]; 00570 }
int ast_getformatbyname | ( | const char * | name | ) |
Gets a format from a name.
name | string of format |
Definition at line 638 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().
00639 { 00640 int x, all, format = 0; 00641 00642 all = strcasecmp(name, "all") ? 0 : 1; 00643 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00644 if(AST_FORMAT_LIST[x].visible && (all || 00645 !strcasecmp(AST_FORMAT_LIST[x].name,name) || 00646 !strcasecmp(AST_FORMAT_LIST[x].name,ast_expand_codec_alias(name)))) { 00647 format |= AST_FORMAT_LIST[x].bits; 00648 if(!all) 00649 break; 00650 } 00651 } 00652 00653 return format; 00654 }
char* ast_getformatname | ( | int | format | ) |
Get the name of a format.
format | id of format |
Definition at line 578 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().
00579 { 00580 int x; 00581 char *ret = "unknown"; 00582 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00583 if(AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == format) { 00584 ret = AST_FORMAT_LIST[x].name; 00585 break; 00586 } 00587 } 00588 return ret; 00589 }
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 591 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().
00592 { 00593 int x; 00594 unsigned len; 00595 char *start, *end = buf; 00596 00597 if (!size) 00598 return buf; 00599 snprintf(end, size, "0x%x (", format); 00600 len = strlen(end); 00601 end += len; 00602 size -= len; 00603 start = end; 00604 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00605 if (AST_FORMAT_LIST[x].visible && (AST_FORMAT_LIST[x].bits & format)) { 00606 snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name); 00607 len = strlen(end); 00608 end += len; 00609 size -= len; 00610 } 00611 } 00612 if (start == end) 00613 snprintf(start, size, "nothing)"); 00614 else if (size > 1) 00615 *(end -1) = ')'; 00616 return buf; 00617 }
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 1318 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().
01319 { 01320 char *parse = NULL, *this = NULL, *psize = NULL; 01321 int format = 0, framems = 0; 01322 01323 parse = ast_strdupa(list); 01324 while ((this = strsep(&parse, ","))) { 01325 framems = 0; 01326 if ((psize = strrchr(this, ':'))) { 01327 *psize++ = '\0'; 01328 if (option_debug) 01329 ast_log(LOG_DEBUG,"Packetization for codec: %s is %s\n", this, psize); 01330 framems = atoi(psize); 01331 if (framems < 0) 01332 framems = 0; 01333 } 01334 if (!(format = ast_getformatbyname(this))) { 01335 ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this); 01336 continue; 01337 } 01338 01339 if (mask) { 01340 if (allowing) 01341 *mask |= format; 01342 else 01343 *mask &= ~format; 01344 } 01345 01346 /* Set up a preference list for audio. Do not include video in preferences 01347 since we can not transcode video and have to use whatever is offered 01348 */ 01349 if (pref && (format & AST_FORMAT_AUDIO_MASK)) { 01350 if (strcasecmp(this, "all")) { 01351 if (allowing) { 01352 ast_codec_pref_append(pref, format); 01353 ast_codec_pref_setsize(pref, format, framems); 01354 } 01355 else 01356 ast_codec_pref_remove(pref, format); 01357 } else if (!allowing) { 01358 memset(pref, 0, sizeof(*pref)); 01359 } 01360 } 01361 } 01362 }
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 556 of file frame.c.
Referenced by __ast_smoother_feed(), iax_frame_wrap(), phone_write_buf(), and smoother_frame_feed().
00557 { 00558 int i; 00559 unsigned short *dst_s = dst; 00560 const unsigned short *src_s = src; 00561 00562 for (i = 0; i < samples; i++) 00563 dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8); 00564 }
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_new(), agent_read(), ast_channel_masquerade(), ast_channel_setwhentohangup(), ast_do_masquerade(), ast_rtcp_read(), ast_rtp_read(), ast_softhangup_nolock(), ast_udptl_read(), conf_run(), features_read(), gtalk_rtp_read(), handle_request_invite(), handle_response_invite(), local_read(), mgcp_rtp_read(), oh323_read(), oh323_rtp_read(), process_rfc2833(), process_sdp(), send_dtmf(), sip_read(), sip_rtp_read(), skinny_rtp_read(), and wakeup_sub().