#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. | |
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_write(), au_seek(), convertcap(), g722tolin_sample(), and pcm_read().
#define AST_FORMAT_G723_1 (1 << 0) |
G.723.1 compression
Definition at line 238 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_samples(), ast_rtp_write(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), 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_new(), 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_recvtext(), ast_rtp_write(), ast_safe_sleep_conditional(), ast_send_image(), ast_slinfactory_destroy(), ast_slinfactory_feed(), ast_slinfactory_flush(), ast_slinfactory_read(), ast_tonepair(), ast_translate(), ast_udptl_bridge(), ast_waitfordigit_full(), ast_write(), ast_writestream(), async_wait(), audio_audiohook_write_list(), autoservice_run(), background_detect_exec(), bridge_native_loop(), bridge_p2p_loop(), calc_cost(), channel_spy(), check_goto_on_transfer(), conf_exec(), conf_flush(), conf_free(), conf_run(), create_jb(), dahdi_bridge(), dictate_exec(), disa_exec(), do_atxfer(), do_idle_thread(), do_parking_thread(), do_waiting(), echo_exec(), find_cache(), gen_generate(), handle_invite_replaces(), handle_recordfile(), iax_park_thread(), ices_exec(), isAnsweringMachine(), jb_empty_and_reset_adaptive(), jb_empty_and_reset_fixed(), jb_get_and_deliver(), masq_park_call(), measurenoise(), moh_files_generator(), monitor_dial(), mp3_exec(), NBScat_exec(), receive_dtmf_digits(), recordthread(), rpt(), run_agi(), send_tone_burst(), send_waveform_to_channel(), sendurl_exec(), speech_background(), spy_generate(), ss_thread(), wait_for_answer(), wait_for_hangup(), wait_for_winner(), waitforring_exec(), and waitstream_core().
#define AST_FRIENDLY_OFFSET 64 |
Definition at line 198 of file frame.h.
Referenced by __get_from_jb(), alsa_read(), ast_frdup(), ast_frisolate(), ast_prod(), ast_rtcp_read(), ast_rtp_read(), ast_smoother_read(), ast_trans_frameout(), ast_udptl_read(), conf_run(), dahdi_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 ast_frame_free(), and ast_frisolate().
#define AST_MALLOCD_HDR (1 << 0) |
Need the header be free'd?
Definition at line 204 of file frame.h.
Referenced by ast_frame_free(), ast_frame_header_new(), ast_frdup(), and ast_frisolate().
#define AST_MALLOCD_SRC (1 << 2) |
Need the source be free'd? (haha!)
Definition at line 208 of file frame.h.
Referenced by ast_frame_free(), and ast_frisolate().
#define AST_MIN_OFFSET 32 |
#define AST_MODEM_T38 1 |
T.38 Fax-over-IP
Definition at line 212 of file frame.h.
Referenced by ast_frame_dump(), and udptl_rx_packet().
#define AST_MODEM_V150 2 |
#define AST_OPTION_AUDIO_MODE 4 |
Set (or clear) Audio (Not-Clear) Mode
Definition at line 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(), rpt(), and try_calling().
#define AST_OPTION_TXGAIN 5 |
Set channel transmit gain Option data is a single signed char representing number of decibels (dB) to set gain to (on top of any gain specified in channel driver)
Definition at line 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 216 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.
00217 { 00218 if (f->frametype != AST_FRAME_VOICE) { 00219 ast_log(LOG_WARNING, "Huh? Can't smooth a non-voice frame!\n"); 00220 return -1; 00221 } 00222 if (!s->format) { 00223 s->format = f->subclass; 00224 s->samplesperbyte = (float)f->samples / (float)f->datalen; 00225 } else if (s->format != f->subclass) { 00226 ast_log(LOG_WARNING, "Smoother was working on %d format frames, now trying to feed %d?\n", s->format, f->subclass); 00227 return -1; 00228 } 00229 if (s->len + f->datalen > SMOOTHER_SIZE) { 00230 ast_log(LOG_WARNING, "Out of smoother space\n"); 00231 return -1; 00232 } 00233 if (((f->datalen == s->size) || 00234 ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) && 00235 !s->opt && 00236 !s->len && 00237 (f->offset >= AST_MIN_OFFSET)) { 00238 /* Optimize by sending the frame we just got 00239 on the next read, thus eliminating the douple 00240 copy */ 00241 if (swap) 00242 ast_swapcopy_samples(f->data, f->data, f->samples); 00243 s->opt = f; 00244 s->opt_needs_swap = swap ? 1 : 0; 00245 return 0; 00246 } 00247 00248 return smoother_frame_feed(s, f, swap); 00249 }
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 639 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().
00640 { 00641 int x; 00642 char *ret = "unknown"; 00643 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00644 if(AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == codec) { 00645 ret = AST_FORMAT_LIST[x].desc; 00646 break; 00647 } 00648 } 00649 return ret; 00650 }
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 1314 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().
01315 { 01316 int x, ret = 0, slot; 01317 01318 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01319 slot = pref->order[x]; 01320 01321 if (!slot) 01322 break; 01323 if (formats & AST_FORMAT_LIST[slot-1].bits) { 01324 ret = AST_FORMAT_LIST[slot-1].bits; 01325 break; 01326 } 01327 } 01328 if(ret & AST_FORMAT_AUDIO_MASK) 01329 return ret; 01330 01331 if (option_debug > 3) 01332 ast_log(LOG_DEBUG, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec"); 01333 01334 return find_best ? ast_best_codec(formats) : 0; 01335 }
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 1573 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().
01574 { 01575 int len = 0; 01576 01577 /* XXX Still need speex, g723, and lpc10 XXX */ 01578 switch(format) { 01579 case AST_FORMAT_ILBC: 01580 len = (samples / 240) * 50; 01581 break; 01582 case AST_FORMAT_GSM: 01583 len = (samples / 160) * 33; 01584 break; 01585 case AST_FORMAT_G729A: 01586 len = samples / 8; 01587 break; 01588 case AST_FORMAT_SLINEAR: 01589 len = samples * 2; 01590 break; 01591 case AST_FORMAT_ULAW: 01592 case AST_FORMAT_ALAW: 01593 len = samples; 01594 break; 01595 case AST_FORMAT_G722: 01596 case AST_FORMAT_ADPCM: 01597 case AST_FORMAT_G726: 01598 case AST_FORMAT_G726_AAL2: 01599 len = samples / 2; 01600 break; 01601 default: 01602 ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format)); 01603 } 01604 01605 return len; 01606 }
int ast_codec_get_samples | ( | struct ast_frame * | f | ) |
Returns the number of samples contained in the frame.
Definition at line 1530 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().
01531 { 01532 int samples=0; 01533 switch(f->subclass) { 01534 case AST_FORMAT_SPEEX: 01535 samples = speex_samples(f->data, f->datalen); 01536 break; 01537 case AST_FORMAT_G723_1: 01538 samples = g723_samples(f->data, f->datalen); 01539 break; 01540 case AST_FORMAT_ILBC: 01541 samples = 240 * (f->datalen / 50); 01542 break; 01543 case AST_FORMAT_GSM: 01544 samples = 160 * (f->datalen / 33); 01545 break; 01546 case AST_FORMAT_G729A: 01547 samples = f->datalen * 8; 01548 break; 01549 case AST_FORMAT_SLINEAR: 01550 samples = f->datalen / 2; 01551 break; 01552 case AST_FORMAT_LPC10: 01553 /* assumes that the RTP packet contains one LPC10 frame */ 01554 samples = 22 * 8; 01555 samples += (((char *)(f->data))[7] & 0x1) * 8; 01556 break; 01557 case AST_FORMAT_ULAW: 01558 case AST_FORMAT_ALAW: 01559 samples = f->datalen; 01560 break; 01561 case AST_FORMAT_G722: 01562 case AST_FORMAT_ADPCM: 01563 case AST_FORMAT_G726: 01564 case AST_FORMAT_G726_AAL2: 01565 samples = f->datalen * 2; 01566 break; 01567 default: 01568 ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass)); 01569 } 01570 return samples; 01571 }
static int ast_codec_interp_len | ( | int | format | ) | [inline, static] |
Gets duration in ms of interpolation frame for a format.
Definition at line 571 of file frame.h.
References AST_FORMAT_ILBC.
Referenced by __get_from_jb(), and jb_get_and_deliver().
00572 { 00573 return (format == AST_FORMAT_ILBC) ? 30 : 20; 00574 }
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 1173 of file frame.c.
References ast_codec_pref_remove(), AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow().
01174 { 01175 int x, newindex = 0; 01176 01177 ast_codec_pref_remove(pref, format); 01178 01179 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01180 if(AST_FORMAT_LIST[x].bits == format) { 01181 newindex = x + 1; 01182 break; 01183 } 01184 } 01185 01186 if(newindex) { 01187 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01188 if(!pref->order[x]) { 01189 pref->order[x] = newindex; 01190 break; 01191 } 01192 } 01193 } 01194 01195 return x; 01196 }
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 1075 of file frame.c.
References ast_codec_pref::order.
Referenced by check_access(), create_addr(), dump_prefs(), and socket_process().
01076 { 01077 int x, differential = (int) 'A', mem; 01078 char *from, *to; 01079 01080 if(right) { 01081 from = pref->order; 01082 to = buf; 01083 mem = size; 01084 } else { 01085 to = pref->order; 01086 from = buf; 01087 mem = 32; 01088 } 01089 01090 memset(to, 0, mem); 01091 for (x = 0; x < 32 ; x++) { 01092 if(!from[x]) 01093 break; 01094 to[x] = right ? (from[x] + differential) : (from[x] - differential); 01095 } 01096 }
struct ast_format_list ast_codec_pref_getsize | ( | struct ast_codec_pref * | pref, | |
int | format | |||
) |
Get packet size for codec.
Definition at line 1275 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().
01276 { 01277 int x, index = -1, framems = 0; 01278 struct ast_format_list fmt = {0}; 01279 01280 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01281 if(AST_FORMAT_LIST[x].bits == format) { 01282 fmt = AST_FORMAT_LIST[x]; 01283 index = x; 01284 break; 01285 } 01286 } 01287 01288 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01289 if(pref->order[x] == (index + 1)) { 01290 framems = pref->framing[x]; 01291 break; 01292 } 01293 } 01294 01295 /* size validation */ 01296 if(!framems) 01297 framems = AST_FORMAT_LIST[index].def_ms; 01298 01299 if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */ 01300 framems -= framems % AST_FORMAT_LIST[index].inc_ms; 01301 01302 if(framems < AST_FORMAT_LIST[index].min_ms) 01303 framems = AST_FORMAT_LIST[index].min_ms; 01304 01305 if(framems > AST_FORMAT_LIST[index].max_ms) 01306 framems = AST_FORMAT_LIST[index].max_ms; 01307 01308 fmt.cur_ms = framems; 01309 01310 return fmt; 01311 }
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 1133 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().
01134 { 01135 int slot = 0; 01136 01137 01138 if((index >= 0) && (index < sizeof(pref->order))) { 01139 slot = pref->order[index]; 01140 } 01141 01142 return slot ? AST_FORMAT_LIST[slot-1].bits : 0; 01143 }
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 1199 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_codec_pref::framing, and ast_codec_pref::order.
Referenced by create_addr().
01200 { 01201 int x, newindex = 0; 01202 01203 /* First step is to get the codecs "index number" */ 01204 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01205 if (AST_FORMAT_LIST[x].bits == format) { 01206 newindex = x + 1; 01207 break; 01208 } 01209 } 01210 /* Done if its unknown */ 01211 if (!newindex) 01212 return; 01213 01214 /* Now find any existing occurrence, or the end */ 01215 for (x = 0; x < 32; x++) { 01216 if (!pref->order[x] || pref->order[x] == newindex) 01217 break; 01218 } 01219 01220 if (only_if_existing && !pref->order[x]) 01221 return; 01222 01223 /* Move down to make space to insert - either all the way to the end, 01224 or as far as the existing location (which will be overwritten) */ 01225 for (; x > 0; x--) { 01226 pref->order[x] = pref->order[x - 1]; 01227 pref->framing[x] = pref->framing[x - 1]; 01228 } 01229 01230 /* And insert the new entry */ 01231 pref->order[0] = newindex; 01232 pref->framing[0] = 0; /* ? */ 01233 }
void ast_codec_pref_remove | ( | struct ast_codec_pref * | pref, | |
int | format | |||
) |
Remove audio a codec from a preference list.
Definition at line 1146 of file frame.c.
References AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_codec_pref_append(), and ast_parse_allow_disallow().
01147 { 01148 struct ast_codec_pref oldorder; 01149 int x, y = 0; 01150 int slot; 01151 int size; 01152 01153 if(!pref->order[0]) 01154 return; 01155 01156 memcpy(&oldorder, pref, sizeof(oldorder)); 01157 memset(pref, 0, sizeof(*pref)); 01158 01159 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01160 slot = oldorder.order[x]; 01161 size = oldorder.framing[x]; 01162 if(! slot) 01163 break; 01164 if(AST_FORMAT_LIST[slot-1].bits != format) { 01165 pref->order[y] = slot; 01166 pref->framing[y++] = size; 01167 } 01168 } 01169 01170 }
int ast_codec_pref_setsize | ( | struct ast_codec_pref * | pref, | |
int | format, | |||
int | framems | |||
) |
Set packet size for codec.
Definition at line 1236 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().
01237 { 01238 int x, index = -1; 01239 01240 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01241 if(AST_FORMAT_LIST[x].bits == format) { 01242 index = x; 01243 break; 01244 } 01245 } 01246 01247 if(index < 0) 01248 return -1; 01249 01250 /* size validation */ 01251 if(!framems) 01252 framems = AST_FORMAT_LIST[index].def_ms; 01253 01254 if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */ 01255 framems -= framems % AST_FORMAT_LIST[index].inc_ms; 01256 01257 if(framems < AST_FORMAT_LIST[index].min_ms) 01258 framems = AST_FORMAT_LIST[index].min_ms; 01259 01260 if(framems > AST_FORMAT_LIST[index].max_ms) 01261 framems = AST_FORMAT_LIST[index].max_ms; 01262 01263 01264 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 01265 if(pref->order[x] == (index + 1)) { 01266 pref->framing[x] = framems; 01267 break; 01268 } 01269 } 01270 01271 return x; 01272 }
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 1098 of file frame.c.
References ast_codec_pref_index(), and ast_getformatname().
Referenced by dump_prefs(), and socket_process().
01099 { 01100 int x, codec; 01101 size_t total_len, slen; 01102 char *formatname; 01103 01104 memset(buf,0,size); 01105 total_len = size; 01106 buf[0] = '('; 01107 total_len--; 01108 for(x = 0; x < 32 ; x++) { 01109 if(total_len <= 0) 01110 break; 01111 if(!(codec = ast_codec_pref_index(pref,x))) 01112 break; 01113 if((formatname = ast_getformatname(codec))) { 01114 slen = strlen(formatname); 01115 if(slen > total_len) 01116 break; 01117 strncat(buf, formatname, total_len - 1); /* safe */ 01118 total_len -= slen; 01119 } 01120 if(total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) { 01121 strncat(buf, "|", total_len - 1); /* safe */ 01122 total_len--; 01123 } 01124 } 01125 if(total_len) { 01126 strncat(buf, ")", total_len - 1); /* safe */ 01127 total_len--; 01128 } 01129 01130 return size - total_len; 01131 }
static force_inline int ast_format_rate | ( | int | format | ) | [static] |
Get the sample rate for a given format.
Definition at line 598 of file frame.h.
References AST_FORMAT_G722.
Referenced by ast_read_generator_actions(), ast_readaudio_callback(), ast_readvideo_callback(), ast_rtp_read(), ast_translate(), calc_cost(), and generator_force().
00599 { 00600 if (format == AST_FORMAT_G722) 00601 return 16000; 00602 00603 return 8000; 00604 }
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 1608 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().
01609 { 01610 int count; 01611 short *fdata = f->data; 01612 short adjust_value = abs(adjustment); 01613 01614 if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR)) 01615 return -1; 01616 01617 if (!adjustment) 01618 return 0; 01619 01620 for (count = 0; count < f->samples; count++) { 01621 if (adjustment > 0) { 01622 ast_slinear_saturated_multiply(&fdata[count], &adjust_value); 01623 } else if (adjustment < 0) { 01624 ast_slinear_saturated_divide(&fdata[count], &adjust_value); 01625 } 01626 } 01627 01628 return 0; 01629 }
void ast_frame_dump | ( | const char * | name, | |
struct ast_frame * | f, | |||
char * | prefix | |||
) |
Dump a frame for debugging purposes
Definition at line 793 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().
00794 { 00795 const char noname[] = "unknown"; 00796 char ftype[40] = "Unknown Frametype"; 00797 char cft[80]; 00798 char subclass[40] = "Unknown Subclass"; 00799 char csub[80]; 00800 char moreinfo[40] = ""; 00801 char cn[60]; 00802 char cp[40]; 00803 char cmn[40]; 00804 00805 if (!name) 00806 name = noname; 00807 00808 00809 if (!f) { 00810 ast_verbose("%s [ %s (NULL) ] [%s]\n", 00811 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00812 term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00813 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00814 return; 00815 } 00816 /* XXX We should probably print one each of voice and video when the format changes XXX */ 00817 if (f->frametype == AST_FRAME_VOICE) 00818 return; 00819 if (f->frametype == AST_FRAME_VIDEO) 00820 return; 00821 switch(f->frametype) { 00822 case AST_FRAME_DTMF_BEGIN: 00823 strcpy(ftype, "DTMF Begin"); 00824 subclass[0] = f->subclass; 00825 subclass[1] = '\0'; 00826 break; 00827 case AST_FRAME_DTMF_END: 00828 strcpy(ftype, "DTMF End"); 00829 subclass[0] = f->subclass; 00830 subclass[1] = '\0'; 00831 break; 00832 case AST_FRAME_CONTROL: 00833 strcpy(ftype, "Control"); 00834 switch(f->subclass) { 00835 case AST_CONTROL_HANGUP: 00836 strcpy(subclass, "Hangup"); 00837 break; 00838 case AST_CONTROL_RING: 00839 strcpy(subclass, "Ring"); 00840 break; 00841 case AST_CONTROL_RINGING: 00842 strcpy(subclass, "Ringing"); 00843 break; 00844 case AST_CONTROL_ANSWER: 00845 strcpy(subclass, "Answer"); 00846 break; 00847 case AST_CONTROL_BUSY: 00848 strcpy(subclass, "Busy"); 00849 break; 00850 case AST_CONTROL_TAKEOFFHOOK: 00851 strcpy(subclass, "Take Off Hook"); 00852 break; 00853 case AST_CONTROL_OFFHOOK: 00854 strcpy(subclass, "Line Off Hook"); 00855 break; 00856 case AST_CONTROL_CONGESTION: 00857 strcpy(subclass, "Congestion"); 00858 break; 00859 case AST_CONTROL_FLASH: 00860 strcpy(subclass, "Flash"); 00861 break; 00862 case AST_CONTROL_WINK: 00863 strcpy(subclass, "Wink"); 00864 break; 00865 case AST_CONTROL_OPTION: 00866 strcpy(subclass, "Option"); 00867 break; 00868 case AST_CONTROL_RADIO_KEY: 00869 strcpy(subclass, "Key Radio"); 00870 break; 00871 case AST_CONTROL_RADIO_UNKEY: 00872 strcpy(subclass, "Unkey Radio"); 00873 break; 00874 case AST_CONTROL_PROGRESS: 00875 strcpy(subclass, "Call Progress"); 00876 break; 00877 case AST_CONTROL_PROCEEDING: 00878 strcpy(subclass, "Proceeding"); 00879 break; 00880 case AST_CONTROL_HOLD: 00881 strcpy(subclass, "Hold"); 00882 break; 00883 case AST_CONTROL_UNHOLD: 00884 strcpy(subclass, "UnHold"); 00885 break; 00886 case -1: 00887 strcpy(subclass, "Stop generators"); 00888 break; 00889 default: 00890 snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass); 00891 } 00892 break; 00893 case AST_FRAME_NULL: 00894 strcpy(ftype, "Null Frame"); 00895 strcpy(subclass, "N/A"); 00896 break; 00897 case AST_FRAME_IAX: 00898 /* Should never happen */ 00899 strcpy(ftype, "IAX Specific"); 00900 snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass); 00901 break; 00902 case AST_FRAME_TEXT: 00903 strcpy(ftype, "Text"); 00904 strcpy(subclass, "N/A"); 00905 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00906 break; 00907 case AST_FRAME_IMAGE: 00908 strcpy(ftype, "Image"); 00909 snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass)); 00910 break; 00911 case AST_FRAME_HTML: 00912 strcpy(ftype, "HTML"); 00913 switch(f->subclass) { 00914 case AST_HTML_URL: 00915 strcpy(subclass, "URL"); 00916 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00917 break; 00918 case AST_HTML_DATA: 00919 strcpy(subclass, "Data"); 00920 break; 00921 case AST_HTML_BEGIN: 00922 strcpy(subclass, "Begin"); 00923 break; 00924 case AST_HTML_END: 00925 strcpy(subclass, "End"); 00926 break; 00927 case AST_HTML_LDCOMPLETE: 00928 strcpy(subclass, "Load Complete"); 00929 break; 00930 case AST_HTML_NOSUPPORT: 00931 strcpy(subclass, "No Support"); 00932 break; 00933 case AST_HTML_LINKURL: 00934 strcpy(subclass, "Link URL"); 00935 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00936 break; 00937 case AST_HTML_UNLINK: 00938 strcpy(subclass, "Unlink"); 00939 break; 00940 case AST_HTML_LINKREJECT: 00941 strcpy(subclass, "Link Reject"); 00942 break; 00943 default: 00944 snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass); 00945 break; 00946 } 00947 break; 00948 case AST_FRAME_MODEM: 00949 strcpy(ftype, "Modem"); 00950 switch (f->subclass) { 00951 case AST_MODEM_T38: 00952 strcpy(subclass, "T.38"); 00953 break; 00954 case AST_MODEM_V150: 00955 strcpy(subclass, "V.150"); 00956 break; 00957 default: 00958 snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass); 00959 break; 00960 } 00961 break; 00962 default: 00963 snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype); 00964 } 00965 if (!ast_strlen_zero(moreinfo)) 00966 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n", 00967 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00968 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00969 f->frametype, 00970 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00971 f->subclass, 00972 term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)), 00973 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00974 else 00975 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n", 00976 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00977 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00978 f->frametype, 00979 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00980 f->subclass, 00981 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00982 }
struct ast_frame* ast_frame_enqueue | ( | struct ast_frame * | head, | |
struct ast_frame * | f, | |||
int | maxlen, | |||
int | dupe | |||
) |
Appends a frame to the end of a list of frames, truncating the maximum length of the list.
void ast_frame_free | ( | struct ast_frame * | fr, | |
int | cache | |||
) |
Requests a frame to be allocated Frees a frame.
fr | Frame to free | |
cache | Whether to consider this frame for frame caching |
Definition at line 354 of file frame.c.
References ast_dsp_frame_freed(), ast_filestream_frame_freed(), AST_FRFLAG_FROM_DSP, AST_FRFLAG_FROM_FILESTREAM, AST_FRFLAG_FROM_TRANSLATOR, AST_LIST_INSERT_HEAD, AST_LIST_LOCK, AST_LIST_REMOVE, AST_LIST_UNLOCK, AST_MALLOCD_DATA, AST_MALLOCD_HDR, AST_MALLOCD_SRC, ast_test_flag, ast_threadstorage_get(), ast_translate_frame_freed(), ast_frame::data, frame_cache, FRAME_CACHE_MAX_SIZE, frames, free, ast_frame::mallocd, ast_frame::offset, and ast_frame::src.
Referenced by mixmonitor_thread().
00355 { 00356 if (ast_test_flag(fr, AST_FRFLAG_FROM_TRANSLATOR)) { 00357 ast_translate_frame_freed(fr); 00358 } else if (ast_test_flag(fr, AST_FRFLAG_FROM_DSP)) { 00359 ast_dsp_frame_freed(fr); 00360 } else if (ast_test_flag(fr, AST_FRFLAG_FROM_FILESTREAM)) { 00361 ast_filestream_frame_freed(fr); 00362 } 00363 00364 if (!fr->mallocd) 00365 return; 00366 00367 #if !defined(LOW_MEMORY) 00368 if (cache && fr->mallocd == AST_MALLOCD_HDR) { 00369 /* Cool, only the header is malloc'd, let's just cache those for now 00370 * to keep things simple... */ 00371 struct ast_frame_cache *frames; 00372 00373 if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames))) 00374 && frames->size < FRAME_CACHE_MAX_SIZE) { 00375 AST_LIST_INSERT_HEAD(&frames->list, fr, frame_list); 00376 frames->size++; 00377 return; 00378 } 00379 } 00380 #endif 00381 00382 if (fr->mallocd & AST_MALLOCD_DATA) { 00383 if (fr->data) 00384 free(fr->data - fr->offset); 00385 } 00386 if (fr->mallocd & AST_MALLOCD_SRC) { 00387 if (fr->src) 00388 free((char *)fr->src); 00389 } 00390 if (fr->mallocd & AST_MALLOCD_HDR) { 00391 #ifdef TRACE_FRAMES 00392 AST_LIST_LOCK(&headerlist); 00393 headers--; 00394 AST_LIST_REMOVE(&headerlist, fr, frame_list); 00395 AST_LIST_UNLOCK(&headerlist); 00396 #endif 00397 free(fr); 00398 } 00399 }
Sums two frames of audio samples.
f1 | The first frame (which will contain the result) | |
f2 | The second frame |
Definition at line 1631 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.
01632 { 01633 int count; 01634 short *data1, *data2; 01635 01636 if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR)) 01637 return -1; 01638 01639 if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR)) 01640 return -1; 01641 01642 if (f1->samples != f2->samples) 01643 return -1; 01644 01645 for (count = 0, data1 = f1->data, data2 = f2->data; 01646 count < f1->samples; 01647 count++, data1++, data2++) 01648 ast_slinear_saturated_add(data1, data2); 01649 01650 return 0; 01651 }
Copies a frame.
fr | frame to copy Duplicates a frame -- should only rarely be used, typically frisolate is good enough |
Definition at line 465 of file frame.c.
References ast_calloc_cache, ast_copy_flags, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, AST_LIST_REMOVE_CURRENT, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, AST_MALLOCD_HDR, ast_threadstorage_get(), ast_frame::data, ast_frame::datalen, ast_frame::delivery, f, frame_cache, frames, ast_frame::frametype, ast_frame::len, len(), ast_frame::mallocd, ast_frame::mallocd_hdr_len, ast_frame::offset, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, and ast_frame::ts.
Referenced by __ast_queue_frame(), ast_jb_put(), ast_rtp_write(), ast_slinfactory_feed(), audiohook_read_frame_single(), autoservice_run(), recordthread(), and rpt().
00466 { 00467 struct ast_frame *out = NULL; 00468 int len, srclen = 0; 00469 void *buf = NULL; 00470 00471 #if !defined(LOW_MEMORY) 00472 struct ast_frame_cache *frames; 00473 #endif 00474 00475 /* Start with standard stuff */ 00476 len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00477 /* If we have a source, add space for it */ 00478 /* 00479 * XXX Watch out here - if we receive a src which is not terminated 00480 * properly, we can be easily attacked. Should limit the size we deal with. 00481 */ 00482 if (f->src) 00483 srclen = strlen(f->src); 00484 if (srclen > 0) 00485 len += srclen + 1; 00486 00487 #if !defined(LOW_MEMORY) 00488 if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) { 00489 AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) { 00490 if (out->mallocd_hdr_len >= len) { 00491 size_t mallocd_len = out->mallocd_hdr_len; 00492 AST_LIST_REMOVE_CURRENT(&frames->list, frame_list); 00493 memset(out, 0, sizeof(*out)); 00494 out->mallocd_hdr_len = mallocd_len; 00495 buf = out; 00496 frames->size--; 00497 break; 00498 } 00499 } 00500 AST_LIST_TRAVERSE_SAFE_END 00501 } 00502 #endif 00503 00504 if (!buf) { 00505 if (!(buf = ast_calloc_cache(1, len))) 00506 return NULL; 00507 out = buf; 00508 out->mallocd_hdr_len = len; 00509 } 00510 00511 out->frametype = f->frametype; 00512 out->subclass = f->subclass; 00513 out->datalen = f->datalen; 00514 out->samples = f->samples; 00515 out->delivery = f->delivery; 00516 /* Set us as having malloc'd header only, so it will eventually 00517 get freed. */ 00518 out->mallocd = AST_MALLOCD_HDR; 00519 out->offset = AST_FRIENDLY_OFFSET; 00520 if (out->datalen) { 00521 out->data = buf + sizeof(*out) + AST_FRIENDLY_OFFSET; 00522 memcpy(out->data, f->data, out->datalen); 00523 } 00524 if (srclen > 0) { 00525 /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */ 00526 char *src; 00527 out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00528 src = (char *) out->src; 00529 /* Must have space since we allocated for it */ 00530 strcpy(src, f->src); 00531 } 00532 ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO); 00533 out->ts = f->ts; 00534 out->len = f->len; 00535 out->seqno = f->seqno; 00536 return out; 00537 }
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 406 of file frame.c.
References ast_clear_flag, ast_copy_flags, ast_frame_header_new(), AST_FRFLAG_FROM_DSP, AST_FRFLAG_FROM_TRANSLATOR, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, ast_malloc, AST_MALLOCD_DATA, AST_MALLOCD_HDR, AST_MALLOCD_SRC, ast_strdup, ast_test_flag, ast_frame::data, ast_frame::datalen, ast_frame::frametype, free, ast_frame::len, ast_frame::mallocd, ast_frame::offset, ast_frame::samples, ast_frame::seqno, ast_frame::src, ast_frame::subclass, and ast_frame::ts.
Referenced by jpeg_read_image().
00407 { 00408 struct ast_frame *out; 00409 void *newdata; 00410 00411 ast_clear_flag(fr, AST_FRFLAG_FROM_TRANSLATOR); 00412 ast_clear_flag(fr, AST_FRFLAG_FROM_DSP); 00413 00414 if (!(fr->mallocd & AST_MALLOCD_HDR)) { 00415 /* Allocate a new header if needed */ 00416 if (!(out = ast_frame_header_new())) 00417 return NULL; 00418 out->frametype = fr->frametype; 00419 out->subclass = fr->subclass; 00420 out->datalen = fr->datalen; 00421 out->samples = fr->samples; 00422 out->offset = fr->offset; 00423 out->data = fr->data; 00424 /* Copy the timing data */ 00425 ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO); 00426 if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) { 00427 out->ts = fr->ts; 00428 out->len = fr->len; 00429 out->seqno = fr->seqno; 00430 } 00431 } else 00432 out = fr; 00433 00434 if (!(fr->mallocd & AST_MALLOCD_SRC)) { 00435 if (fr->src) { 00436 if (!(out->src = ast_strdup(fr->src))) { 00437 if (out != fr) 00438 free(out); 00439 return NULL; 00440 } 00441 } 00442 } else 00443 out->src = fr->src; 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 if (out != fr) 00450 free(out); 00451 return NULL; 00452 } 00453 newdata += AST_FRIENDLY_OFFSET; 00454 out->offset = AST_FRIENDLY_OFFSET; 00455 out->datalen = fr->datalen; 00456 memcpy(newdata, fr->data, fr->datalen); 00457 out->data = newdata; 00458 } 00459 00460 out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA; 00461 00462 return out; 00463 }
struct ast_format_list* ast_get_format_list | ( | size_t * | size | ) |
Definition at line 555 of file frame.c.
References AST_FORMAT_LIST.
00556 { 00557 *size = (sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0])); 00558 return AST_FORMAT_LIST; 00559 }
struct ast_format_list* ast_get_format_list_index | ( | int | index | ) |
Definition at line 550 of file frame.c.
References AST_FORMAT_LIST.
00551 { 00552 return &AST_FORMAT_LIST[index]; 00553 }
int ast_getformatbyname | ( | const char * | name | ) |
Gets a format from a name.
name | string of format |
Definition at line 621 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().
00622 { 00623 int x, all, format = 0; 00624 00625 all = strcasecmp(name, "all") ? 0 : 1; 00626 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00627 if(AST_FORMAT_LIST[x].visible && (all || 00628 !strcasecmp(AST_FORMAT_LIST[x].name,name) || 00629 !strcasecmp(AST_FORMAT_LIST[x].name,ast_expand_codec_alias(name)))) { 00630 format |= AST_FORMAT_LIST[x].bits; 00631 if(!all) 00632 break; 00633 } 00634 } 00635 00636 return format; 00637 }
char* ast_getformatname | ( | int | format | ) |
Get the name of a format.
format | id of format |
Definition at line 561 of file frame.c.
References AST_FORMAT_LIST, ast_format_list::bits, name, and ast_format_list::visible.
Referenced by __ast_play_and_record(), __ast_read(), __ast_register_translator(), __login_exec(), _sip_show_peer(), add_codec_to_answer(), add_codec_to_sdp(), agent_call(), ast_codec_get_len(), ast_codec_get_samples(), ast_codec_pref_string(), ast_dsp_process(), ast_frame_dump(), ast_openvstream(), ast_rtp_write(), ast_slinfactory_feed(), ast_streamfile(), ast_translator_build_path(), ast_unregister_translator(), ast_writestream(), background_detect_exec(), dahdi_read(), do_waiting(), eagi_exec(), func_channel_read(), function_iaxpeer(), function_sippeer(), gtalk_show_channels(), iax2_request(), iax2_show_channels(), iax2_show_peer(), iax_show_provisioning(), moh_classes_show(), moh_release(), oh323_rtp_read(), phone_setup(), print_codec_to_cli(), rebuild_matrix(), register_translator(), set_format(), set_peer_capabilities(), show_codecs(), show_codecs_deprecated(), show_file_formats(), show_file_formats_deprecated(), show_image_formats(), show_image_formats_deprecated(), show_translation(), show_translation_deprecated(), sip_request_call(), sip_rtp_read(), and socket_process().
00562 { 00563 int x; 00564 char *ret = "unknown"; 00565 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00566 if(AST_FORMAT_LIST[x].visible && AST_FORMAT_LIST[x].bits == format) { 00567 ret = AST_FORMAT_LIST[x].name; 00568 break; 00569 } 00570 } 00571 return ret; 00572 }
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 574 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().
00575 { 00576 int x; 00577 unsigned len; 00578 char *start, *end = buf; 00579 00580 if (!size) 00581 return buf; 00582 snprintf(end, size, "0x%x (", format); 00583 len = strlen(end); 00584 end += len; 00585 size -= len; 00586 start = end; 00587 for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) { 00588 if (AST_FORMAT_LIST[x].visible && (AST_FORMAT_LIST[x].bits & format)) { 00589 snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name); 00590 len = strlen(end); 00591 end += len; 00592 size -= len; 00593 } 00594 } 00595 if (start == end) 00596 snprintf(start, size, "nothing)"); 00597 else if (size > 1) 00598 *(end -1) = ')'; 00599 return buf; 00600 }
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 1337 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().
01338 { 01339 char *parse = NULL, *this = NULL, *psize = NULL; 01340 int format = 0, framems = 0; 01341 01342 parse = ast_strdupa(list); 01343 while ((this = strsep(&parse, ","))) { 01344 framems = 0; 01345 if ((psize = strrchr(this, ':'))) { 01346 *psize++ = '\0'; 01347 if (option_debug) 01348 ast_log(LOG_DEBUG,"Packetization for codec: %s is %s\n", this, psize); 01349 framems = atoi(psize); 01350 if (framems < 0) 01351 framems = 0; 01352 } 01353 if (!(format = ast_getformatbyname(this))) { 01354 ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this); 01355 continue; 01356 } 01357 01358 if (mask) { 01359 if (allowing) 01360 *mask |= format; 01361 else 01362 *mask &= ~format; 01363 } 01364 01365 /* Set up a preference list for audio. Do not include video in preferences 01366 since we can not transcode video and have to use whatever is offered 01367 */ 01368 if (pref && (format & AST_FORMAT_AUDIO_MASK)) { 01369 if (strcasecmp(this, "all")) { 01370 if (allowing) { 01371 ast_codec_pref_append(pref, format); 01372 ast_codec_pref_setsize(pref, format, framems); 01373 } 01374 else 01375 ast_codec_pref_remove(pref, format); 01376 } else if (!allowing) { 01377 memset(pref, 0, sizeof(*pref)); 01378 } 01379 } 01380 } 01381 }
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 191 of file frame.c.
References ast_malloc, ast_smoother_reset(), and s.
Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().
00192 { 00193 struct ast_smoother *s; 00194 if (size < 1) 00195 return NULL; 00196 if ((s = ast_malloc(sizeof(*s)))) 00197 ast_smoother_reset(s, size); 00198 return s; 00199 }
struct ast_frame* ast_smoother_read | ( | struct ast_smoother * | s | ) |
Definition at line 251 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().
00252 { 00253 struct ast_frame *opt; 00254 int len; 00255 00256 /* IF we have an optimization frame, send it */ 00257 if (s->opt) { 00258 if (s->opt->offset < AST_FRIENDLY_OFFSET) 00259 ast_log(LOG_WARNING, "Returning a frame of inappropriate offset (%d).\n", 00260 s->opt->offset); 00261 opt = s->opt; 00262 s->opt = NULL; 00263 return opt; 00264 } 00265 00266 /* Make sure we have enough data */ 00267 if (s->len < s->size) { 00268 /* Or, if this is a G.729 frame with VAD on it, send it immediately anyway */ 00269 if (!((s->flags & AST_SMOOTHER_FLAG_G729) && (s->size % 10))) 00270 return NULL; 00271 } 00272 len = s->size; 00273 if (len > s->len) 00274 len = s->len; 00275 /* Make frame */ 00276 s->f.frametype = AST_FRAME_VOICE; 00277 s->f.subclass = s->format; 00278 s->f.data = s->framedata + AST_FRIENDLY_OFFSET; 00279 s->f.offset = AST_FRIENDLY_OFFSET; 00280 s->f.datalen = len; 00281 /* Samples will be improper given VAD, but with VAD the concept really doesn't even exist */ 00282 s->f.samples = len * s->samplesperbyte; /* XXX rounding */ 00283 s->f.delivery = s->delivery; 00284 /* Fill Data */ 00285 memcpy(s->f.data, s->data, len); 00286 s->len -= len; 00287 /* Move remaining data to the front if applicable */ 00288 if (s->len) { 00289 /* In principle this should all be fine because if we are sending 00290 G.729 VAD, the next timestamp will take over anyawy */ 00291 memmove(s->data, s->data + len, s->len); 00292 if (!ast_tvzero(s->delivery)) { 00293 /* If we have delivery time, increment it, otherwise, leave it at 0 */ 00294 s->delivery = ast_tvadd(s->delivery, ast_samp2tv(s->f.samples, 8000)); 00295 } 00296 } 00297 /* Return frame */ 00298 return &s->f; 00299 }
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 169 of file frame.c.
References s, and smoother_frame_feed().
Referenced by ast_rtp_codec_setpref().
00170 { 00171 /* if there is no change, then nothing to do */ 00172 if (s->size == bytes) { 00173 return; 00174 } 00175 /* set the new desired output size */ 00176 s->size = bytes; 00177 /* if there is no 'optimized' frame in the smoother, 00178 * then there is nothing left to do 00179 */ 00180 if (!s->opt) { 00181 return; 00182 } 00183 /* there is an 'optimized' frame here at the old size, 00184 * but it must now be put into the buffer so the data 00185 * can be extracted at the new size 00186 */ 00187 smoother_frame_feed(s, s->opt, s->opt_needs_swap); 00188 s->opt = NULL; 00189 }
void ast_smoother_reset | ( | struct ast_smoother * | s, | |
int | bytes | |||
) |
Definition at line 163 of file frame.c.
References s.
Referenced by ast_smoother_new().
00164 { 00165 memset(s, 0, sizeof(*s)); 00166 s->size = bytes; 00167 }
void ast_smoother_set_flags | ( | struct ast_smoother * | smoother, | |
int | flags | |||
) |
Definition at line 206 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 211 of file frame.c.
References s.
Referenced by ast_rtp_write().
00212 { 00213 return (s->flags & flag); 00214 }
void ast_swapcopy_samples | ( | void * | dst, | |
const void * | src, | |||
int | samples | |||
) |
Definition at line 539 of file frame.c.
Referenced by __ast_smoother_feed(), iax_frame_wrap(), phone_write_buf(), and smoother_frame_feed().
00540 { 00541 int i; 00542 unsigned short *dst_s = dst; 00543 const unsigned short *src_s = src; 00544 00545 for (i = 0; i < samples; i++) 00546 dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8); 00547 }
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 139 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().