#include <sys/time.h>
#include "asterisk/endian.h"
#include "asterisk/linkedlists.h"
Go to the source code of this file.
Data Structures | |
struct | ast_codec_pref |
struct | ast_format_list |
Definition of supported media formats (codecs). More... | |
struct | ast_frame |
Data structure associated with a single frame of data. More... | |
struct | ast_option_header |
struct | oprmode |
AST_Smoother | |
#define | ast_smoother_feed(s, f) __ast_smoother_feed(s, f, 0) |
#define | ast_smoother_feed_be(s, f) __ast_smoother_feed(s, f, 0) |
#define | ast_smoother_feed_le(s, f) __ast_smoother_feed(s, f, 1) |
int | __ast_smoother_feed (struct ast_smoother *s, struct ast_frame *f, int swap) |
void | ast_smoother_free (struct ast_smoother *s) |
int | ast_smoother_get_flags (struct ast_smoother *smoother) |
ast_smoother * | ast_smoother_new (int bytes) |
ast_frame * | ast_smoother_read (struct ast_smoother *s) |
void | ast_smoother_reconfigure (struct ast_smoother *s, int bytes) |
Reconfigure an existing smoother to output a different number of bytes per frame. | |
void | ast_smoother_reset (struct ast_smoother *s, int bytes) |
void | ast_smoother_set_flags (struct ast_smoother *smoother, int flags) |
int | ast_smoother_test_flag (struct ast_smoother *s, int flag) |
Defines | |
#define | AST_FORMAT_ADPCM (1 << 5) |
#define | AST_FORMAT_ALAW (1 << 3) |
#define | AST_FORMAT_AUDIO_MASK ((1 << 16)-1) |
#define | AST_FORMAT_AUDIO_UNDEFINED ((1 << 13) | (1 << 14)) |
#define | AST_FORMAT_G722 (1 << 12) |
#define | AST_FORMAT_G723_1 (1 << 0) |
#define | AST_FORMAT_G726 (1 << 11) |
#define | AST_FORMAT_G726_AAL2 (1 << 4) |
#define | AST_FORMAT_G729A (1 << 8) |
#define | AST_FORMAT_GSM (1 << 1) |
#define | AST_FORMAT_H261 (1 << 18) |
#define | AST_FORMAT_H263 (1 << 19) |
#define | AST_FORMAT_H263_PLUS (1 << 20) |
#define | AST_FORMAT_H264 (1 << 21) |
#define | AST_FORMAT_ILBC (1 << 10) |
#define | AST_FORMAT_JPEG (1 << 16) |
#define | AST_FORMAT_LPC10 (1 << 7) |
#define | AST_FORMAT_MP4_VIDEO (1 << 22) |
#define | AST_FORMAT_PNG (1 << 17) |
#define | AST_FORMAT_SLINEAR (1 << 6) |
#define | AST_FORMAT_SLINEAR16 (1 << 15) |
#define | AST_FORMAT_SPEEX (1 << 9) |
#define | AST_FORMAT_T140 (1 << 25) |
#define | AST_FORMAT_TEXT_MASK (((1 << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK)) |
#define | AST_FORMAT_ULAW (1 << 2) |
#define | AST_FORMAT_VIDEO_MASK (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) |
#define | ast_frame_byteswap_be(fr) do { ; } while(0) |
#define | ast_frame_byteswap_le(fr) do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data, __f->data, __f->samples); } while(0) |
#define | AST_FRAME_DTMF AST_FRAME_DTMF_END |
#define | AST_FRAME_SET_BUFFER(fr, _base, _ofs, _datalen) |
#define | ast_frfree(fr) ast_frame_free(fr, 1) |
#define | AST_FRIENDLY_OFFSET 64 |
Offset into a frame's data buffer. | |
#define | AST_HTML_BEGIN 4 |
#define | AST_HTML_DATA 2 |
#define | AST_HTML_END 8 |
#define | AST_HTML_LDCOMPLETE 16 |
#define | AST_HTML_LINKREJECT 20 |
#define | AST_HTML_LINKURL 18 |
#define | AST_HTML_NOSUPPORT 17 |
#define | AST_HTML_UNLINK 19 |
#define | AST_HTML_URL 1 |
#define | AST_MALLOCD_DATA (1 << 1) |
#define | AST_MALLOCD_HDR (1 << 0) |
#define | AST_MALLOCD_SRC (1 << 2) |
#define | AST_MIN_OFFSET 32 |
#define | AST_MODEM_T38 1 |
#define | AST_MODEM_V150 2 |
#define | AST_OPTION_AUDIO_MODE 4 |
#define | AST_OPTION_ECHOCAN 8 |
#define | AST_OPTION_FLAG_ACCEPT 1 |
#define | AST_OPTION_FLAG_ANSWER 5 |
#define | AST_OPTION_FLAG_QUERY 4 |
#define | AST_OPTION_FLAG_REJECT 2 |
#define | AST_OPTION_FLAG_REQUEST 0 |
#define | AST_OPTION_FLAG_WTF 6 |
#define | AST_OPTION_OPRMODE 7 |
#define | AST_OPTION_RELAXDTMF 3 |
#define | AST_OPTION_RXGAIN 6 |
#define | AST_OPTION_T38_STATE 10 |
#define | AST_OPTION_TDD 2 |
#define | AST_OPTION_TONE_VERIFY 1 |
#define | AST_OPTION_TXGAIN 5 |
#define | AST_SMOOTHER_FLAG_BE (1 << 1) |
#define | AST_SMOOTHER_FLAG_G729 (1 << 0) |
Enumerations | |
enum | { AST_FRFLAG_HAS_TIMING_INFO = (1 << 0), AST_FRFLAG_FROM_TRANSLATOR = (1 << 1), AST_FRFLAG_FROM_DSP = (1 << 2), AST_FRFLAG_FROM_FILESTREAM = (1 << 3) } |
enum | ast_control_frame_type { AST_CONTROL_HANGUP = 1, AST_CONTROL_RING = 2, AST_CONTROL_RINGING = 3, AST_CONTROL_ANSWER = 4, AST_CONTROL_BUSY = 5, AST_CONTROL_TAKEOFFHOOK = 6, AST_CONTROL_OFFHOOK = 7, AST_CONTROL_CONGESTION = 8, AST_CONTROL_FLASH = 9, AST_CONTROL_WINK = 10, AST_CONTROL_OPTION = 11, AST_CONTROL_RADIO_KEY = 12, AST_CONTROL_RADIO_UNKEY = 13, AST_CONTROL_PROGRESS = 14, AST_CONTROL_PROCEEDING = 15, AST_CONTROL_HOLD = 16, AST_CONTROL_UNHOLD = 17, AST_CONTROL_VIDUPDATE = 18, AST_CONTROL_T38 = 19, AST_CONTROL_SRCUPDATE = 20 } |
enum | ast_control_t38 { AST_T38_REQUEST_NEGOTIATE = 1, AST_T38_REQUEST_TERMINATE, AST_T38_NEGOTIATED, AST_T38_TERMINATED, AST_T38_REFUSED } |
enum | ast_frame_type { AST_FRAME_DTMF_END = 1, AST_FRAME_VOICE, AST_FRAME_VIDEO, AST_FRAME_CONTROL, AST_FRAME_NULL, AST_FRAME_IAX, AST_FRAME_TEXT, AST_FRAME_IMAGE, AST_FRAME_HTML, AST_FRAME_CNG, AST_FRAME_MODEM, AST_FRAME_DTMF_BEGIN } |
Frame types. More... | |
Functions | |
char * | ast_codec2str (int codec) |
Get a name from a format Gets a name from a format. | |
int | ast_codec_choose (struct ast_codec_pref *pref, int formats, int find_best) |
Select the best audio format according to preference list from supplied options. If "find_best" is non-zero then if nothing is found, the "Best" format of the format list is selected, otherwise 0 is returned. | |
int | ast_codec_get_len (int format, int samples) |
Returns the number of bytes for the number of samples of the given format. | |
int | ast_codec_get_samples (struct ast_frame *f) |
Returns the number of samples contained in the frame. | |
static int | ast_codec_interp_len (int format) |
Gets duration in ms of interpolation frame for a format. | |
int | ast_codec_pref_append (struct ast_codec_pref *pref, int format) |
Append a audio codec to a preference list, removing it first if it was already there. | |
void | ast_codec_pref_convert (struct ast_codec_pref *pref, char *buf, size_t size, int right) |
Shift an audio codec preference list up or down 65 bytes so that it becomes an ASCII string. | |
ast_format_list | ast_codec_pref_getsize (struct ast_codec_pref *pref, int format) |
Get packet size for codec. | |
int | ast_codec_pref_index (struct ast_codec_pref *pref, int index) |
Codec located at a particular place in the preference index. | |
void | ast_codec_pref_init (struct ast_codec_pref *pref) |
Initialize an audio codec preference to "no preference". | |
void | ast_codec_pref_prepend (struct ast_codec_pref *pref, int format, int only_if_existing) |
Prepend an audio codec to a preference list, removing it first if it was already there. | |
void | ast_codec_pref_remove (struct ast_codec_pref *pref, int format) |
Remove audio a codec from a preference list. | |
int | ast_codec_pref_setsize (struct ast_codec_pref *pref, int format, int framems) |
Set packet size for codec. | |
int | ast_codec_pref_string (struct ast_codec_pref *pref, char *buf, size_t size) |
Dump audio codec preference list into a string. | |
static force_inline int | ast_format_rate (int format) |
Get the sample rate for a given format. | |
int | ast_frame_adjust_volume (struct ast_frame *f, int adjustment) |
Adjusts the volume of the audio samples contained in a frame. | |
void | ast_frame_dump (const char *name, struct ast_frame *f, char *prefix) |
ast_frame * | ast_frame_enqueue (struct ast_frame *head, struct ast_frame *f, int maxlen, int dupe) |
Appends a frame to the end of a list of frames, truncating the maximum length of the list. | |
void | ast_frame_free (struct ast_frame *fr, int cache) |
Requests a frame to be allocated Frees a frame 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. | |
int | ast_parse_allow_disallow (struct ast_codec_pref *pref, int *mask, const char *list, int allowing) |
Parse an "allow" or "deny" line in a channel or device configuration and update the capabilities mask and pref if provided. Video codecs are not added to codec preference lists, since we can not transcode. | |
void | ast_swapcopy_samples (void *dst, const void *src, int samples) |
Variables | |
ast_frame | ast_null_frame |
Definition in file frame.h.
#define AST_FORMAT_ADPCM (1 << 5) |
ADPCM (IMA)
Definition at line 255 of file frame.h.
Referenced by adpcmtolin_sample(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), convertcap(), vox_read(), and vox_write().
#define AST_FORMAT_ALAW (1 << 3) |
Raw A-law data (G.711)
Definition at line 251 of file frame.h.
Referenced by alawtolin_sample(), alawtoulaw_sample(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), cb_events(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_new(), dahdi_read(), dahdi_write(), find_transcoders(), is_encoder(), misdn_read(), misdn_set_opt_exec(), oh323_rtp_read(), pcm_seek(), pcm_write(), read_config(), and start_rtp().
#define AST_FORMAT_AUDIO_MASK ((1 << 16)-1) |
Maximum audio mask
Definition at line 275 of file frame.h.
Referenced by add_sdp(), ast_best_codec(), ast_channel_make_compatible_helper(), ast_codec_choose(), ast_filehelper(), ast_openstream_full(), ast_openvstream(), ast_parse_allow_disallow(), ast_playstream(), ast_request(), ast_rtp_read(), ast_translate_available_formats(), ast_translator_best_choice(), ast_writestream(), begin_dial_channel(), filestream_destructor(), func_channel_read(), generator_force(), gtalk_rtp_read(), jingle_rtp_read(), oh323_request(), phone_read(), process_sdp(), set_format(), sip_call(), sip_request_call(), sip_rtp_read(), sip_write(), skinny_request(), transmit_connect_with_sdp(), and transmit_modify_with_sdp().
#define AST_FORMAT_AUDIO_UNDEFINED ((1 << 13) | (1 << 14)) |
#define AST_FORMAT_G722 (1 << 12) |
G.722
Definition at line 269 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_rtp_write(), ast_slinfactory_feed(), au_seek(), convertcap(), g722tolin16_sample(), g722tolin_sample(), and pcm_read().
#define AST_FORMAT_G723_1 (1 << 0) |
G.723.1 compression
Definition at line 245 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_write(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_destroy(), dahdi_translate(), g723_read(), g723_write(), load_module(), phone_request(), phone_setup(), phone_write(), register_translator(), and start_rtp().
#define AST_FORMAT_G726 (1 << 11) |
ADPCM (G.726, 32kbps, RFC3551 codeword packing)
Definition at line 267 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_set_rtpmap_type(), g726_read(), g726_write(), and g726tolin_sample().
#define AST_FORMAT_G726_AAL2 (1 << 4) |
ADPCM (G.726, 32kbps, AAL2 codeword packing)
Definition at line 253 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_rtp_lookup_mime_subtype(), ast_rtp_set_rtpmap_type(), codec_ast2skinny(), codec_skinny2ast(), and setup_rtp_connection().
#define AST_FORMAT_G729A (1 << 8) |
G.729A audio
Definition at line 261 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), codec_ast2skinny(), codec_skinny2ast(), convertcap(), dahdi_destroy(), dahdi_translate(), g729_read(), g729_write(), load_module(), phone_request(), phone_setup(), phone_write(), and start_rtp().
#define AST_FORMAT_GSM (1 << 1) |
GSM compression
Definition at line 247 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), convertcap(), gsm_read(), gsm_write(), gsmtolin_sample(), wav_read(), and wav_write().
#define AST_FORMAT_H261 (1 << 18) |
H.261 Video
Definition at line 281 of file frame.h.
Referenced by codec_ast2skinny(), codec_skinny2ast(), and h261_encap().
#define AST_FORMAT_H263 (1 << 19) |
H.263 Video
Definition at line 283 of file frame.h.
Referenced by codec_ast2skinny(), codec_skinny2ast(), h263_encap(), h263_read(), and h263_write().
#define AST_FORMAT_H263_PLUS (1 << 20) |
#define AST_FORMAT_H264 (1 << 21) |
H.264 Video
Definition at line 287 of file frame.h.
Referenced by h264_encap(), h264_read(), and h264_write().
#define AST_FORMAT_ILBC (1 << 10) |
iLBC Free Compression
Definition at line 265 of file frame.h.
Referenced by add_codec_to_sdp(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_codec_interp_len(), convertcap(), ilbc_read(), ilbc_write(), and ilbctolin_sample().
#define AST_FORMAT_JPEG (1 << 16) |
JPEG Images
Definition at line 277 of file frame.h.
Referenced by jpeg_read_image(), and jpeg_write_image().
#define AST_FORMAT_LPC10 (1 << 7) |
LPC10, 180 samples/frame
Definition at line 259 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_samples(), and lpc10tolin_sample().
#define AST_FORMAT_MP4_VIDEO (1 << 22) |
#define AST_FORMAT_PNG (1 << 17) |
#define AST_FORMAT_SLINEAR (1 << 6) |
Raw 16-bit Signed Linear (8000 Hz) PCM
Definition at line 257 of file frame.h.
Referenced by __ast_play_and_record(), __ast_register_translator(), action_originate(), agent_new(), alsa_new(), alsa_read(), alsa_request(), ast_audiohook_read_frame(), ast_best_codec(), ast_channel_make_compatible_helper(), ast_channel_start_silence_generator(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_call_progress(), ast_dsp_digitdetect(), ast_dsp_process(), ast_dsp_silence(), ast_frame_adjust_volume(), ast_frame_slinear_sum(), ast_rtp_read(), ast_slinfactory_feed(), ast_speech_new(), attempt_reconnect(), audio_audiohook_write_list(), audiohook_read_frame_both(), audiohook_read_frame_single(), background_detect_exec(), build_conf(), chanspy_exec(), conf_run(), dahdi_read(), dahdi_translate(), dahdi_write(), dictate_exec(), do_waiting(), eagi_exec(), extenspy_exec(), fax_generator_generate(), find_transcoders(), function_ilink(), handle_jack_audio(), handle_recordfile(), handle_speechcreate(), handle_speechrecognize(), iax_frame_wrap(), ices_exec(), init_outgoing(), is_encoder(), isAnsweringMachine(), jack_hook_callback(), linear_alloc(), linear_generator(), lintoadpcm_sample(), lintoalaw_sample(), lintog722_sample(), lintog726_sample(), lintogsm_sample(), lintoilbc_sample(), lintolpc10_sample(), lintospeex_sample(), lintoulaw_sample(), load_module(), load_moh_classes(), local_ast_moh_start(), measurenoise(), misdn_set_opt_exec(), mixmonitor_thread(), moh_class_malloc(), mp3_exec(), nbs_request(), nbs_xwrite(), NBScat_exec(), ogg_vorbis_read(), ogg_vorbis_write(), oh323_rtp_read(), orig_app(), orig_exten(), oss_new(), oss_read(), oss_request(), parkandannounce_exec(), phone_new(), phone_read(), phone_request(), phone_setup(), phone_write(), playtones_alloc(), read_config(), rpt(), rpt_call(), rpt_tele_thread(), send_waveform_to_channel(), silence_generator_generate(), slin8_to_slin16_sample(), slinear_read(), slinear_write(), socket_process(), speech_background(), spy_generate(), tonepair_alloc(), transmit_audio(), usbradio_new(), usbradio_read(), usbradio_request(), wav_read(), and wav_write().
#define AST_FORMAT_SLINEAR16 (1 << 15) |
Raw 16-bit Signed Linear (16000 Hz) PCM
Definition at line 273 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_format_rate(), ast_slinfactory_feed(), console_new(), lin16tog722_sample(), slin16_to_slin8_sample(), slinear_read(), slinear_write(), and stream_monitor().
#define AST_FORMAT_SPEEX (1 << 9) |
SpeeX Free Compression
Definition at line 263 of file frame.h.
Referenced by ast_best_codec(), ast_codec_get_samples(), ast_rtp_write(), convertcap(), and speextolin_sample().
#define AST_FORMAT_T140 (1 << 25) |
T.140 Text format - ITU T.140, RFC 4351
Definition at line 292 of file frame.h.
Referenced by ast_write().
#define AST_FORMAT_TEXT_MASK (((1 << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK)) |
Definition at line 293 of file frame.h.
Referenced by add_sdp(), ast_request(), check_peer_ok(), sip_new(), and sip_rtp_read().
#define AST_FORMAT_ULAW (1 << 2) |
Raw mu-law data (G.711)
Definition at line 249 of file frame.h.
Referenced by __adsi_transmit_messages(), _ast_adsi_transmit_message_full(), adsi_careful_send(), alarmreceiver_exec(), ast_best_codec(), ast_codec_get_len(), ast_codec_get_samples(), ast_dsp_process(), calc_energy(), codec_ast2skinny(), codec_skinny2ast(), conf_run(), convertcap(), dahdi_new(), dahdi_read(), dahdi_translate(), dahdi_write(), find_transcoders(), is_encoder(), load_module(), milliwatt_generate(), oh323_rtp_read(), old_milliwatt_exec(), phone_request(), phone_setup(), phone_write(), pri_dchannel(), send_tone_burst(), start_rtp(), ulawtoalaw_sample(), and ulawtolin_sample().
#define AST_FORMAT_VIDEO_MASK (((1 << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) |
Definition at line 290 of file frame.h.
Referenced by add_sdp(), ast_openvstream(), ast_request(), ast_rtp_read(), ast_translate_available_formats(), check_peer_ok(), check_user_ok(), create_addr_from_peer(), func_channel_read(), gtalk_new(), gtalk_rtp_read(), jingle_new(), jingle_rtp_read(), sip_new(), and sip_rtp_read().
#define ast_frame_byteswap_be | ( | fr | ) | do { ; } while(0) |
#define ast_frame_byteswap_le | ( | fr | ) | do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data, __f->data, __f->samples); } while(0) |
#define AST_FRAME_DTMF AST_FRAME_DTMF_END |
Definition at line 124 of file frame.h.
Referenced by __adsi_transmit_messages(), __ast_play_and_record(), action_dahdidialoffhook(), agent_ack_sleep(), ast_audiohook_write_list(), ast_bridge_call(), ast_dsp_process(), ast_feature_request_and_dial(), ast_jb_put(), background_detect_exec(), cb_events(), channel_spy(), cli_console_dial(), conf_exec(), conf_run(), console_dial(), dahdi_bridge(), dahdi_read(), dictate_exec(), disa_exec(), do_immediate_setup(), echo_exec(), eivr_comm(), gtalk_handle_dtmf(), handle_recordfile(), handle_request(), handle_request_info(), handle_speechrecognize(), jingle_handle_dtmf(), keypad_digit(), mgcp_rtp_read(), misdn_bridge(), mp3_exec(), NBScat_exec(), oh323_rtp_read(), phone_exception(), process_ast_dsp(), receive_dtmf_digits(), rpt(), rpt_call(), send_waveform_to_channel(), sip_rtp_read(), speech_background(), ss_thread(), transmit_audio(), unistim_do_senddigit(), unistim_senddigit_end(), volume_callback(), and wait_for_winner().
#define AST_FRAME_SET_BUFFER | ( | fr, | |||
_base, | |||||
_ofs, | |||||
_datalen | ) |
Value:
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 186 of file frame.h.
Referenced by fax_generator_generate(), g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), ilbc_read(), ogg_vorbis_read(), pcm_read(), slinear_read(), t38_tx_packet_handler(), vox_read(), and wav_read().
#define ast_frfree | ( | fr | ) | ast_frame_free(fr, 1) |
Definition at line 434 of file frame.h.
Referenced by __adsi_transmit_messages(), __ast_answer(), __ast_play_and_record(), __ast_queue_frame(), __ast_read(), __ast_request_and_dial(), adsi_careful_send(), agent_ack_sleep(), agent_read(), ast_audiohook_read_frame(), ast_autoservice_stop(), ast_bridge_call(), ast_channel_free(), ast_dsp_process(), ast_feature_request_and_dial(), ast_jb_destroy(), ast_jb_put(), ast_readaudio_callback(), ast_readvideo_callback(), ast_recvtext(), ast_rtp_write(), ast_safe_sleep_conditional(), ast_send_image(), ast_slinfactory_destroy(), ast_slinfactory_feed(), ast_slinfactory_flush(), ast_slinfactory_read(), ast_tonepair(), ast_translate(), ast_udptl_bridge(), ast_waitfordigit_full(), ast_write(), ast_writestream(), async_wait(), audio_audiohook_write_list(), autoservice_run(), background_detect_exec(), bridge_native_loop(), bridge_p2p_loop(), builtin_atxfer(), calc_cost(), channel_spy(), check_goto_on_transfer(), conf_exec(), conf_flush(), conf_free(), conf_run(), create_jb(), dahdi_bridge(), dictate_exec(), disa_exec(), do_idle_thread(), do_parking_thread(), do_waiting(), echo_exec(), eivr_comm(), find_cache(), gen_generate(), handle_cli_file_convert(), handle_invite_replaces(), handle_recordfile(), handle_speechrecognize(), iax_park_thread(), ices_exec(), isAnsweringMachine(), jb_empty_and_reset_adaptive(), jb_empty_and_reset_fixed(), jb_get_and_deliver(), launch_asyncagi(), masq_park_call(), measurenoise(), moh_files_generator(), monitor_dial(), mp3_exec(), NBScat_exec(), receive_dtmf_digits(), recordthread(), rpt(), run_agi(), send_tone_burst(), send_waveform_to_channel(), sendurl_exec(), speech_background(), spy_generate(), ss_thread(), transmit_audio(), transmit_t38(), wait_for_answer(), wait_for_hangup(), wait_for_winner(), waitforring_exec(), and waitstream_core().
#define AST_FRIENDLY_OFFSET 64 |
Offset into a frame's data buffer.
By providing some "empty" space prior to the actual data of an ast_frame, this gives any consumer of the frame ample space to prepend other necessary information without having to create a new buffer.
As an example, RTP can use the data from an ast_frame and simply prepend the RTP header information into the space provided by AST_FRIENDLY_OFFSET instead of having to create a new buffer with the necessary space allocated.
Definition at line 207 of file frame.h.
Referenced by __get_from_jb(), alsa_read(), ast_frdup(), ast_frisolate(), ast_prod(), ast_rtcp_read(), ast_rtp_read(), ast_smoother_read(), ast_trans_frameout(), ast_udptl_read(), conf_run(), dahdi_decoder_frameout(), dahdi_encoder_frameout(), dahdi_read(), fax_generator_generate(), g723_read(), g726_read(), g729_read(), gsm_read(), h263_read(), h264_read(), iax_frame_wrap(), ilbc_read(), jb_get_and_deliver(), linear_generator(), milliwatt_generate(), moh_generate(), mohalloc(), mp3_exec(), NBScat_exec(), newpvt(), ogg_vorbis_read(), oss_read(), pcm_read(), phone_read(), process_rfc3389(), send_tone_burst(), send_waveform_to_channel(), slinear_read(), sms_generate(), usbradio_read(), vox_read(), and wav_read().
#define AST_HTML_BEGIN 4 |
#define AST_HTML_DATA 2 |
#define AST_HTML_END 8 |
#define AST_HTML_LDCOMPLETE 16 |
Load is complete
Definition at line 233 of file frame.h.
Referenced by ast_frame_dump(), and sendurl_exec().
#define AST_HTML_LINKREJECT 20 |
#define AST_HTML_LINKURL 18 |
#define AST_HTML_NOSUPPORT 17 |
Peer is unable to support HTML
Definition at line 235 of file frame.h.
Referenced by ast_frame_dump(), and sendurl_exec().
#define AST_HTML_UNLINK 19 |
#define AST_HTML_URL 1 |
Sending a URL
Definition at line 225 of file frame.h.
Referenced by ast_channel_sendurl(), ast_frame_dump(), and sip_sendhtml().
#define AST_MALLOCD_DATA (1 << 1) |
Need the data be free'd?
Definition at line 213 of file frame.h.
Referenced by __frame_free(), ast_frisolate(), and create_video_frame().
#define AST_MALLOCD_HDR (1 << 0) |
Need the header be free'd?
Definition at line 211 of file frame.h.
Referenced by __frame_free(), ast_frame_header_new(), ast_frdup(), ast_frisolate(), and create_video_frame().
#define AST_MALLOCD_SRC (1 << 2) |
Need the source be free'd? (haha!)
Definition at line 215 of file frame.h.
Referenced by __frame_free(), and ast_frisolate().
#define AST_MIN_OFFSET 32 |
#define AST_MODEM_T38 1 |
T.38 Fax-over-IP
Definition at line 219 of file frame.h.
Referenced by ast_frame_dump(), t38_tx_packet_handler(), transmit_t38(), and udptl_rx_packet().
#define AST_MODEM_V150 2 |
#define AST_OPTION_AUDIO_MODE 4 |
Set (or clear) Audio (Not-Clear) Mode
Definition at line 348 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 370 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 345 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 364 of file frame.h.
Referenced by dahdi_setoption(), func_channel_write(), iax2_setoption(), play_record_review(), reset_volumes(), set_talk_volume(), and vm_forwardoptions().
#define AST_OPTION_T38_STATE 10 |
Definition at line 376 of file frame.h.
Referenced by ast_channel_get_t38_state(), and sip_queryoption().
#define AST_OPTION_TDD 2 |
Put a compatible channel into TDD (TTY for the hearing-impared) mode
Definition at line 342 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 339 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 356 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 326 of file frame.h.
Referenced by __ast_smoother_feed(), ast_smoother_read(), and smoother_frame_feed().
anonymous enum |
Definition at line 126 of file frame.h.
00126 { 00127 /*! This frame contains valid timing information */ 00128 AST_FRFLAG_HAS_TIMING_INFO = (1 << 0), 00129 /*! This frame came from a translator and is still the original frame. 00130 * The translator can not be free'd if the frame inside of it still has 00131 * this flag set. */ 00132 AST_FRFLAG_FROM_TRANSLATOR = (1 << 1), 00133 /*! This frame came from a dsp and is still the original frame. 00134 * The dsp cannot be free'd if the frame inside of it still has 00135 * this flag set. */ 00136 AST_FRFLAG_FROM_DSP = (1 << 2), 00137 /*! This frame came from a filestream and is still the original frame. 00138 * The filestream cannot be free'd if the frame inside of it still has 00139 * this flag set. */ 00140 AST_FRFLAG_FROM_FILESTREAM = (1 << 3), 00141 };
Definition at line 295 of file frame.h.
00295 { 00296 AST_CONTROL_HANGUP = 1, /*!< Other end has hungup */ 00297 AST_CONTROL_RING = 2, /*!< Local ring */ 00298 AST_CONTROL_RINGING = 3, /*!< Remote end is ringing */ 00299 AST_CONTROL_ANSWER = 4, /*!< Remote end has answered */ 00300 AST_CONTROL_BUSY = 5, /*!< Remote end is busy */ 00301 AST_CONTROL_TAKEOFFHOOK = 6, /*!< Make it go off hook */ 00302 AST_CONTROL_OFFHOOK = 7, /*!< Line is off hook */ 00303 AST_CONTROL_CONGESTION = 8, /*!< Congestion (circuits busy) */ 00304 AST_CONTROL_FLASH = 9, /*!< Flash hook */ 00305 AST_CONTROL_WINK = 10, /*!< Wink */ 00306 AST_CONTROL_OPTION = 11, /*!< Set a low-level option */ 00307 AST_CONTROL_RADIO_KEY = 12, /*!< Key Radio */ 00308 AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */ 00309 AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */ 00310 AST_CONTROL_PROCEEDING = 15, /*!< Indicate CALL PROCEEDING */ 00311 AST_CONTROL_HOLD = 16, /*!< Indicate call is placed on hold */ 00312 AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold */ 00313 AST_CONTROL_VIDUPDATE = 18, /*!< Indicate video frame update */ 00314 AST_CONTROL_T38 = 19, /*!< T38 state change request/notification */ 00315 AST_CONTROL_SRCUPDATE = 20, /*!< Indicate source of media has changed */ 00316 };
enum ast_control_t38 |
Definition at line 318 of file frame.h.
00318 { 00319 AST_T38_REQUEST_NEGOTIATE = 1, /*!< Request T38 on a channel (voice to fax) */ 00320 AST_T38_REQUEST_TERMINATE, /*!< Terminate T38 on a channel (fax to voice) */ 00321 AST_T38_NEGOTIATED, /*!< T38 negotiated (fax mode) */ 00322 AST_T38_TERMINATED, /*!< T38 terminated (back to voice) */ 00323 AST_T38_REFUSED /*!< T38 refused for some reason (usually rejected by remote end) */ 00324 };
enum ast_frame_type |
Frame types.
Definition at line 97 of file frame.h.
00097 { 00098 /*! DTMF end event, subclass is the digit */ 00099 AST_FRAME_DTMF_END = 1, 00100 /*! Voice data, subclass is AST_FORMAT_* */ 00101 AST_FRAME_VOICE, 00102 /*! Video frame, maybe?? :) */ 00103 AST_FRAME_VIDEO, 00104 /*! A control frame, subclass is AST_CONTROL_* */ 00105 AST_FRAME_CONTROL, 00106 /*! An empty, useless frame */ 00107 AST_FRAME_NULL, 00108 /*! Inter Asterisk Exchange private frame type */ 00109 AST_FRAME_IAX, 00110 /*! Text messages */ 00111 AST_FRAME_TEXT, 00112 /*! Image Frames */ 00113 AST_FRAME_IMAGE, 00114 /*! HTML Frame */ 00115 AST_FRAME_HTML, 00116 /*! Comfort Noise frame (subclass is level of CNG in -dBov), 00117 body may include zero or more 8-bit quantization coefficients */ 00118 AST_FRAME_CNG, 00119 /*! Modem-over-IP data streams */ 00120 AST_FRAME_MODEM, 00121 /*! DTMF begin event, subclass is the digit */ 00122 AST_FRAME_DTMF_BEGIN, 00123 };
int __ast_smoother_feed | ( | struct ast_smoother * | s, | |
struct ast_frame * | f, | |||
int | swap | |||
) |
Definition at line 199 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.
00200 { 00201 if (f->frametype != AST_FRAME_VOICE) { 00202 ast_log(LOG_WARNING, "Huh? Can't smooth a non-voice frame!\n"); 00203 return -1; 00204 } 00205 if (!s->format) { 00206 s->format = f->subclass; 00207 s->samplesperbyte = (float)f->samples / (float)f->datalen; 00208 } else if (s->format != f->subclass) { 00209 ast_log(LOG_WARNING, "Smoother was working on %d format frames, now trying to feed %d?\n", s->format, f->subclass); 00210 return -1; 00211 } 00212 if (s->len + f->datalen > SMOOTHER_SIZE) { 00213 ast_log(LOG_WARNING, "Out of smoother space\n"); 00214 return -1; 00215 } 00216 if (((f->datalen == s->size) || 00217 ((f->datalen < 10) && (s->flags & AST_SMOOTHER_FLAG_G729))) && 00218 !s->opt && 00219 !s->len && 00220 (f->offset >= AST_MIN_OFFSET)) { 00221 /* Optimize by sending the frame we just got 00222 on the next read, thus eliminating the douple 00223 copy */ 00224 if (swap) 00225 ast_swapcopy_samples(f->data, f->data, f->samples); 00226 s->opt = f; 00227 s->opt_needs_swap = swap ? 1 : 0; 00228 return 0; 00229 } 00230 00231 return smoother_frame_feed(s, f, swap); 00232 }
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 646 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, and ast_format_list::desc.
Referenced by moh_alloc(), show_codec_n(), and show_codecs().
00647 { 00648 int x; 00649 char *ret = "unknown"; 00650 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00651 if (AST_FORMAT_LIST[x].bits == codec) { 00652 ret = AST_FORMAT_LIST[x].desc; 00653 break; 00654 } 00655 } 00656 return ret; 00657 }
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 1204 of file frame.c.
References ARRAY_LEN, ast_best_codec(), ast_debug, AST_FORMAT_AUDIO_MASK, AST_FORMAT_LIST, ast_format_list::bits, and ast_codec_pref::order.
Referenced by __oh323_new(), gtalk_new(), jingle_new(), process_sdp(), sip_new(), and socket_process().
01205 { 01206 int x, ret = 0, slot; 01207 01208 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01209 slot = pref->order[x]; 01210 01211 if (!slot) 01212 break; 01213 if (formats & AST_FORMAT_LIST[slot-1].bits) { 01214 ret = AST_FORMAT_LIST[slot-1].bits; 01215 break; 01216 } 01217 } 01218 if (ret & AST_FORMAT_AUDIO_MASK) 01219 return ret; 01220 01221 ast_debug(4, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec"); 01222 01223 return find_best ? ast_best_codec(formats) : 0; 01224 }
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 1468 of file frame.c.
References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G722, AST_FORMAT_G723_1, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_ILBC, AST_FORMAT_SLINEAR, AST_FORMAT_SLINEAR16, AST_FORMAT_ULAW, ast_getformatname(), ast_log(), len(), and LOG_WARNING.
Referenced by moh_generate(), and monmp3thread().
01469 { 01470 int len = 0; 01471 01472 /* XXX Still need speex, g723, and lpc10 XXX */ 01473 switch(format) { 01474 case AST_FORMAT_G723_1: 01475 len = (samples / 240) * 20; 01476 break; 01477 case AST_FORMAT_ILBC: 01478 len = (samples / 240) * 50; 01479 break; 01480 case AST_FORMAT_GSM: 01481 len = (samples / 160) * 33; 01482 break; 01483 case AST_FORMAT_G729A: 01484 len = samples / 8; 01485 break; 01486 case AST_FORMAT_SLINEAR: 01487 case AST_FORMAT_SLINEAR16: 01488 len = samples * 2; 01489 break; 01490 case AST_FORMAT_ULAW: 01491 case AST_FORMAT_ALAW: 01492 len = samples; 01493 break; 01494 case AST_FORMAT_G722: 01495 case AST_FORMAT_ADPCM: 01496 case AST_FORMAT_G726: 01497 case AST_FORMAT_G726_AAL2: 01498 len = samples / 2; 01499 break; 01500 default: 01501 ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format)); 01502 } 01503 01504 return len; 01505 }
int ast_codec_get_samples | ( | struct ast_frame * | f | ) |
Returns the number of samples contained in the frame.
Definition at line 1424 of file frame.c.
References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G722, AST_FORMAT_G723_1, AST_FORMAT_G726, AST_FORMAT_G726_AAL2, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_ILBC, AST_FORMAT_LPC10, AST_FORMAT_SLINEAR, AST_FORMAT_SLINEAR16, AST_FORMAT_SPEEX, AST_FORMAT_ULAW, ast_getformatname(), ast_log(), f, g723_samples(), LOG_WARNING, and speex_samples().
Referenced by ast_rtp_read(), isAnsweringMachine(), moh_generate(), schedule_delivery(), socket_process(), and socket_process_meta().
01425 { 01426 int samples=0; 01427 switch(f->subclass) { 01428 case AST_FORMAT_SPEEX: 01429 samples = speex_samples(f->data, f->datalen); 01430 break; 01431 case AST_FORMAT_G723_1: 01432 samples = g723_samples(f->data, f->datalen); 01433 break; 01434 case AST_FORMAT_ILBC: 01435 samples = 240 * (f->datalen / 50); 01436 break; 01437 case AST_FORMAT_GSM: 01438 samples = 160 * (f->datalen / 33); 01439 break; 01440 case AST_FORMAT_G729A: 01441 samples = f->datalen * 8; 01442 break; 01443 case AST_FORMAT_SLINEAR: 01444 case AST_FORMAT_SLINEAR16: 01445 samples = f->datalen / 2; 01446 break; 01447 case AST_FORMAT_LPC10: 01448 /* assumes that the RTP packet contains one LPC10 frame */ 01449 samples = 22 * 8; 01450 samples += (((char *)(f->data))[7] & 0x1) * 8; 01451 break; 01452 case AST_FORMAT_ULAW: 01453 case AST_FORMAT_ALAW: 01454 samples = f->datalen; 01455 break; 01456 case AST_FORMAT_G722: 01457 case AST_FORMAT_ADPCM: 01458 case AST_FORMAT_G726: 01459 case AST_FORMAT_G726_AAL2: 01460 samples = f->datalen * 2; 01461 break; 01462 default: 01463 ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass)); 01464 } 01465 return samples; 01466 }
static int ast_codec_interp_len | ( | int | format | ) | [inline, static] |
Gets duration in ms of interpolation frame for a format.
Definition at line 625 of file frame.h.
References AST_FORMAT_ILBC.
Referenced by __get_from_jb(), and jb_get_and_deliver().
00626 { 00627 return (format == AST_FORMAT_ILBC) ? 30 : 20; 00628 }
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 1063 of file frame.c.
References ARRAY_LEN, ast_codec_pref_remove(), AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow().
01064 { 01065 int x, newindex = 0; 01066 01067 ast_codec_pref_remove(pref, format); 01068 01069 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01070 if (AST_FORMAT_LIST[x].bits == format) { 01071 newindex = x + 1; 01072 break; 01073 } 01074 } 01075 01076 if (newindex) { 01077 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01078 if (!pref->order[x]) { 01079 pref->order[x] = newindex; 01080 break; 01081 } 01082 } 01083 } 01084 01085 return x; 01086 }
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 965 of file frame.c.
References ast_codec_pref::order.
Referenced by check_access(), create_addr(), dump_prefs(), and socket_process().
00966 { 00967 int x, differential = (int) 'A', mem; 00968 char *from, *to; 00969 00970 if (right) { 00971 from = pref->order; 00972 to = buf; 00973 mem = size; 00974 } else { 00975 to = pref->order; 00976 from = buf; 00977 mem = 32; 00978 } 00979 00980 memset(to, 0, mem); 00981 for (x = 0; x < 32 ; x++) { 00982 if (!from[x]) 00983 break; 00984 to[x] = right ? (from[x] + differential) : (from[x] - differential); 00985 } 00986 }
struct ast_format_list ast_codec_pref_getsize | ( | struct ast_codec_pref * | pref, | |
int | format | |||
) |
Get packet size for codec.
Definition at line 1165 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_format_list::bits, ast_format_list::cur_ms, ast_format_list::def_ms, format, ast_format_list::inc_ms, ast_format_list::max_ms, and ast_format_list::min_ms.
Referenced by add_codec_to_sdp(), ast_rtp_bridge(), ast_rtp_codec_setpref(), ast_rtp_write(), handle_open_receive_channel_ack_message(), skinny_set_rtp_peer(), and transmit_connect().
01166 { 01167 int x, index = -1, framems = 0; 01168 struct ast_format_list fmt = { 0, }; 01169 01170 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01171 if (AST_FORMAT_LIST[x].bits == format) { 01172 fmt = AST_FORMAT_LIST[x]; 01173 index = x; 01174 break; 01175 } 01176 } 01177 01178 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01179 if (pref->order[x] == (index + 1)) { 01180 framems = pref->framing[x]; 01181 break; 01182 } 01183 } 01184 01185 /* size validation */ 01186 if (!framems) 01187 framems = AST_FORMAT_LIST[index].def_ms; 01188 01189 if (AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */ 01190 framems -= framems % AST_FORMAT_LIST[index].inc_ms; 01191 01192 if (framems < AST_FORMAT_LIST[index].min_ms) 01193 framems = AST_FORMAT_LIST[index].min_ms; 01194 01195 if (framems > AST_FORMAT_LIST[index].max_ms) 01196 framems = AST_FORMAT_LIST[index].max_ms; 01197 01198 fmt.cur_ms = framems; 01199 01200 return fmt; 01201 }
int ast_codec_pref_index | ( | struct ast_codec_pref * | pref, | |
int | index | |||
) |
Codec located at a particular place in the preference index.
Definition at line 1023 of file frame.c.
References AST_FORMAT_LIST, ast_format_list::bits, and ast_codec_pref::order.
Referenced by _sip_show_peer(), add_sdp(), ast_codec_pref_string(), function_iaxpeer(), function_sippeer(), gtalk_invite(), handle_cli_iax2_show_peer(), jingle_accept_call(), print_codec_to_cli(), and socket_process().
01024 { 01025 int slot = 0; 01026 01027 01028 if ((index >= 0) && (index < sizeof(pref->order))) { 01029 slot = pref->order[index]; 01030 } 01031 01032 return slot ? AST_FORMAT_LIST[slot-1].bits : 0; 01033 }
void ast_codec_pref_init | ( | struct ast_codec_pref * | pref | ) |
void ast_codec_pref_prepend | ( | struct ast_codec_pref * | pref, | |
int | format, | |||
int | only_if_existing | |||
) |
Prepend an audio codec to a preference list, removing it first if it was already there.
Definition at line 1089 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_codec_pref::framing, and ast_codec_pref::order.
Referenced by create_addr().
01090 { 01091 int x, newindex = 0; 01092 01093 /* First step is to get the codecs "index number" */ 01094 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01095 if (AST_FORMAT_LIST[x].bits == format) { 01096 newindex = x + 1; 01097 break; 01098 } 01099 } 01100 /* Done if its unknown */ 01101 if (!newindex) 01102 return; 01103 01104 /* Now find any existing occurrence, or the end */ 01105 for (x = 0; x < 32; x++) { 01106 if (!pref->order[x] || pref->order[x] == newindex) 01107 break; 01108 } 01109 01110 if (only_if_existing && !pref->order[x]) 01111 return; 01112 01113 /* Move down to make space to insert - either all the way to the end, 01114 or as far as the existing location (which will be overwritten) */ 01115 for (; x > 0; x--) { 01116 pref->order[x] = pref->order[x - 1]; 01117 pref->framing[x] = pref->framing[x - 1]; 01118 } 01119 01120 /* And insert the new entry */ 01121 pref->order[0] = newindex; 01122 pref->framing[0] = 0; /* ? */ 01123 }
void ast_codec_pref_remove | ( | struct ast_codec_pref * | pref, | |
int | format | |||
) |
Remove audio a codec from a preference list.
Definition at line 1036 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, and ast_codec_pref::order.
Referenced by ast_codec_pref_append(), and ast_parse_allow_disallow().
01037 { 01038 struct ast_codec_pref oldorder; 01039 int x, y = 0; 01040 int slot; 01041 int size; 01042 01043 if (!pref->order[0]) 01044 return; 01045 01046 memcpy(&oldorder, pref, sizeof(oldorder)); 01047 memset(pref, 0, sizeof(*pref)); 01048 01049 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01050 slot = oldorder.order[x]; 01051 size = oldorder.framing[x]; 01052 if (! slot) 01053 break; 01054 if (AST_FORMAT_LIST[slot-1].bits != format) { 01055 pref->order[y] = slot; 01056 pref->framing[y++] = size; 01057 } 01058 } 01059 01060 }
int ast_codec_pref_setsize | ( | struct ast_codec_pref * | pref, | |
int | format, | |||
int | framems | |||
) |
Set packet size for codec.
Definition at line 1126 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_format_list::def_ms, ast_codec_pref::framing, ast_format_list::inc_ms, ast_format_list::max_ms, ast_format_list::min_ms, and ast_codec_pref::order.
Referenced by ast_parse_allow_disallow(), and process_sdp().
01127 { 01128 int x, index = -1; 01129 01130 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01131 if (AST_FORMAT_LIST[x].bits == format) { 01132 index = x; 01133 break; 01134 } 01135 } 01136 01137 if (index < 0) 01138 return -1; 01139 01140 /* size validation */ 01141 if (!framems) 01142 framems = AST_FORMAT_LIST[index].def_ms; 01143 01144 if (AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */ 01145 framems -= framems % AST_FORMAT_LIST[index].inc_ms; 01146 01147 if (framems < AST_FORMAT_LIST[index].min_ms) 01148 framems = AST_FORMAT_LIST[index].min_ms; 01149 01150 if (framems > AST_FORMAT_LIST[index].max_ms) 01151 framems = AST_FORMAT_LIST[index].max_ms; 01152 01153 01154 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 01155 if (pref->order[x] == (index + 1)) { 01156 pref->framing[x] = framems; 01157 break; 01158 } 01159 } 01160 01161 return x; 01162 }
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 988 of file frame.c.
References ast_codec_pref_index(), and ast_getformatname().
Referenced by dump_prefs(), and socket_process().
00989 { 00990 int x, codec; 00991 size_t total_len, slen; 00992 char *formatname; 00993 00994 memset(buf,0,size); 00995 total_len = size; 00996 buf[0] = '('; 00997 total_len--; 00998 for(x = 0; x < 32 ; x++) { 00999 if (total_len <= 0) 01000 break; 01001 if (!(codec = ast_codec_pref_index(pref,x))) 01002 break; 01003 if ((formatname = ast_getformatname(codec))) { 01004 slen = strlen(formatname); 01005 if (slen > total_len) 01006 break; 01007 strncat(buf, formatname, total_len - 1); /* safe */ 01008 total_len -= slen; 01009 } 01010 if (total_len && x < 31 && ast_codec_pref_index(pref , x + 1)) { 01011 strncat(buf, "|", total_len - 1); /* safe */ 01012 total_len--; 01013 } 01014 } 01015 if (total_len) { 01016 strncat(buf, ")", total_len - 1); /* safe */ 01017 total_len--; 01018 } 01019 01020 return size - total_len; 01021 }
static force_inline int ast_format_rate | ( | int | format | ) | [static] |
Get the sample rate for a given format.
Definition at line 652 of file frame.h.
References AST_FORMAT_G722, and AST_FORMAT_SLINEAR16.
Referenced by ast_read_generator_actions(), ast_readaudio_callback(), ast_readvideo_callback(), ast_rtp_read(), ast_smoother_read(), ast_translate(), calc_cost(), and generator_force().
00653 { 00654 if (format == AST_FORMAT_G722 || format == AST_FORMAT_SLINEAR16) 00655 return 16000; 00656 00657 return 8000; 00658 }
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 1507 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(), conf_run(), and volume_callback().
01508 { 01509 int count; 01510 short *fdata = f->data; 01511 short adjust_value = abs(adjustment); 01512 01513 if ((f->frametype != AST_FRAME_VOICE) || (f->subclass != AST_FORMAT_SLINEAR)) 01514 return -1; 01515 01516 if (!adjustment) 01517 return 0; 01518 01519 for (count = 0; count < f->samples; count++) { 01520 if (adjustment > 0) { 01521 ast_slinear_saturated_multiply(&fdata[count], &adjust_value); 01522 } else if (adjustment < 0) { 01523 ast_slinear_saturated_divide(&fdata[count], &adjust_value); 01524 } 01525 } 01526 01527 return 0; 01528 }
void ast_frame_dump | ( | const char * | name, | |
struct ast_frame * | f, | |||
char * | prefix | |||
) |
Dump a frame for debugging purposes
Definition at line 748 of file frame.c.
References AST_CONTROL_ANSWER, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_FLASH, AST_CONTROL_HANGUP, AST_CONTROL_HOLD, AST_CONTROL_OFFHOOK, AST_CONTROL_OPTION, AST_CONTROL_RADIO_KEY, AST_CONTROL_RADIO_UNKEY, AST_CONTROL_RING, AST_CONTROL_RINGING, AST_CONTROL_T38, AST_CONTROL_TAKEOFFHOOK, AST_CONTROL_UNHOLD, AST_CONTROL_WINK, ast_copy_string(), AST_FRAME_CONTROL, AST_FRAME_DTMF_BEGIN, AST_FRAME_DTMF_END, AST_FRAME_HTML, AST_FRAME_IAX, AST_FRAME_IMAGE, AST_FRAME_MODEM, AST_FRAME_NULL, AST_FRAME_TEXT, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_getformatname(), AST_HTML_BEGIN, AST_HTML_DATA, AST_HTML_END, AST_HTML_LDCOMPLETE, AST_HTML_LINKREJECT, AST_HTML_LINKURL, AST_HTML_NOSUPPORT, AST_HTML_UNLINK, AST_HTML_URL, AST_MODEM_T38, AST_MODEM_V150, ast_strlen_zero(), AST_T38_NEGOTIATED, AST_T38_REFUSED, AST_T38_REQUEST_NEGOTIATE, AST_T38_REQUEST_TERMINATE, AST_T38_TERMINATED, ast_verbose(), COLOR_BLACK, COLOR_BRCYAN, COLOR_BRGREEN, COLOR_BRMAGENTA, COLOR_BRRED, COLOR_YELLOW, f, and term_color().
Referenced by __ast_read(), and ast_write().
00749 { 00750 const char noname[] = "unknown"; 00751 char ftype[40] = "Unknown Frametype"; 00752 char cft[80]; 00753 char subclass[40] = "Unknown Subclass"; 00754 char csub[80]; 00755 char moreinfo[40] = ""; 00756 char cn[60]; 00757 char cp[40]; 00758 char cmn[40]; 00759 const char *message = "Unknown"; 00760 00761 if (!name) 00762 name = noname; 00763 00764 00765 if (!f) { 00766 ast_verbose("%s [ %s (NULL) ] [%s]\n", 00767 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00768 term_color(cft, "HANGUP", COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00769 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00770 return; 00771 } 00772 /* XXX We should probably print one each of voice and video when the format changes XXX */ 00773 if (f->frametype == AST_FRAME_VOICE) 00774 return; 00775 if (f->frametype == AST_FRAME_VIDEO) 00776 return; 00777 switch(f->frametype) { 00778 case AST_FRAME_DTMF_BEGIN: 00779 strcpy(ftype, "DTMF Begin"); 00780 subclass[0] = f->subclass; 00781 subclass[1] = '\0'; 00782 break; 00783 case AST_FRAME_DTMF_END: 00784 strcpy(ftype, "DTMF End"); 00785 subclass[0] = f->subclass; 00786 subclass[1] = '\0'; 00787 break; 00788 case AST_FRAME_CONTROL: 00789 strcpy(ftype, "Control"); 00790 switch(f->subclass) { 00791 case AST_CONTROL_HANGUP: 00792 strcpy(subclass, "Hangup"); 00793 break; 00794 case AST_CONTROL_RING: 00795 strcpy(subclass, "Ring"); 00796 break; 00797 case AST_CONTROL_RINGING: 00798 strcpy(subclass, "Ringing"); 00799 break; 00800 case AST_CONTROL_ANSWER: 00801 strcpy(subclass, "Answer"); 00802 break; 00803 case AST_CONTROL_BUSY: 00804 strcpy(subclass, "Busy"); 00805 break; 00806 case AST_CONTROL_TAKEOFFHOOK: 00807 strcpy(subclass, "Take Off Hook"); 00808 break; 00809 case AST_CONTROL_OFFHOOK: 00810 strcpy(subclass, "Line Off Hook"); 00811 break; 00812 case AST_CONTROL_CONGESTION: 00813 strcpy(subclass, "Congestion"); 00814 break; 00815 case AST_CONTROL_FLASH: 00816 strcpy(subclass, "Flash"); 00817 break; 00818 case AST_CONTROL_WINK: 00819 strcpy(subclass, "Wink"); 00820 break; 00821 case AST_CONTROL_OPTION: 00822 strcpy(subclass, "Option"); 00823 break; 00824 case AST_CONTROL_RADIO_KEY: 00825 strcpy(subclass, "Key Radio"); 00826 break; 00827 case AST_CONTROL_RADIO_UNKEY: 00828 strcpy(subclass, "Unkey Radio"); 00829 break; 00830 case AST_CONTROL_HOLD: 00831 strcpy(subclass, "Hold"); 00832 break; 00833 case AST_CONTROL_UNHOLD: 00834 strcpy(subclass, "Unhold"); 00835 break; 00836 case AST_CONTROL_T38: 00837 if (f->datalen != sizeof(enum ast_control_t38)) { 00838 message = "Invalid"; 00839 } else { 00840 enum ast_control_t38 state = *((enum ast_control_t38 *) f->data); 00841 if (state == AST_T38_REQUEST_NEGOTIATE) 00842 message = "Negotiation Requested"; 00843 else if (state == AST_T38_REQUEST_TERMINATE) 00844 message = "Negotiation Request Terminated"; 00845 else if (state == AST_T38_NEGOTIATED) 00846 message = "Negotiated"; 00847 else if (state == AST_T38_TERMINATED) 00848 message = "Terminated"; 00849 else if (state == AST_T38_REFUSED) 00850 message = "Refused"; 00851 } 00852 snprintf(subclass, sizeof(subclass), "T38/%s", message); 00853 break; 00854 case -1: 00855 strcpy(subclass, "Stop generators"); 00856 break; 00857 default: 00858 snprintf(subclass, sizeof(subclass), "Unknown control '%d'", f->subclass); 00859 } 00860 break; 00861 case AST_FRAME_NULL: 00862 strcpy(ftype, "Null Frame"); 00863 strcpy(subclass, "N/A"); 00864 break; 00865 case AST_FRAME_IAX: 00866 /* Should never happen */ 00867 strcpy(ftype, "IAX Specific"); 00868 snprintf(subclass, sizeof(subclass), "IAX Frametype %d", f->subclass); 00869 break; 00870 case AST_FRAME_TEXT: 00871 strcpy(ftype, "Text"); 00872 strcpy(subclass, "N/A"); 00873 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00874 break; 00875 case AST_FRAME_IMAGE: 00876 strcpy(ftype, "Image"); 00877 snprintf(subclass, sizeof(subclass), "Image format %s\n", ast_getformatname(f->subclass)); 00878 break; 00879 case AST_FRAME_HTML: 00880 strcpy(ftype, "HTML"); 00881 switch(f->subclass) { 00882 case AST_HTML_URL: 00883 strcpy(subclass, "URL"); 00884 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00885 break; 00886 case AST_HTML_DATA: 00887 strcpy(subclass, "Data"); 00888 break; 00889 case AST_HTML_BEGIN: 00890 strcpy(subclass, "Begin"); 00891 break; 00892 case AST_HTML_END: 00893 strcpy(subclass, "End"); 00894 break; 00895 case AST_HTML_LDCOMPLETE: 00896 strcpy(subclass, "Load Complete"); 00897 break; 00898 case AST_HTML_NOSUPPORT: 00899 strcpy(subclass, "No Support"); 00900 break; 00901 case AST_HTML_LINKURL: 00902 strcpy(subclass, "Link URL"); 00903 ast_copy_string(moreinfo, f->data, sizeof(moreinfo)); 00904 break; 00905 case AST_HTML_UNLINK: 00906 strcpy(subclass, "Unlink"); 00907 break; 00908 case AST_HTML_LINKREJECT: 00909 strcpy(subclass, "Link Reject"); 00910 break; 00911 default: 00912 snprintf(subclass, sizeof(subclass), "Unknown HTML frame '%d'\n", f->subclass); 00913 break; 00914 } 00915 break; 00916 case AST_FRAME_MODEM: 00917 strcpy(ftype, "Modem"); 00918 switch (f->subclass) { 00919 case AST_MODEM_T38: 00920 strcpy(subclass, "T.38"); 00921 break; 00922 case AST_MODEM_V150: 00923 strcpy(subclass, "V.150"); 00924 break; 00925 default: 00926 snprintf(subclass, sizeof(subclass), "Unknown MODEM frame '%d'\n", f->subclass); 00927 break; 00928 } 00929 break; 00930 default: 00931 snprintf(ftype, sizeof(ftype), "Unknown Frametype '%d'", f->frametype); 00932 } 00933 if (!ast_strlen_zero(moreinfo)) 00934 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) '%s' ] [%s]\n", 00935 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00936 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00937 f->frametype, 00938 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00939 f->subclass, 00940 term_color(cmn, moreinfo, COLOR_BRGREEN, COLOR_BLACK, sizeof(cmn)), 00941 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00942 else 00943 ast_verbose("%s [ TYPE: %s (%d) SUBCLASS: %s (%d) ] [%s]\n", 00944 term_color(cp, prefix, COLOR_BRMAGENTA, COLOR_BLACK, sizeof(cp)), 00945 term_color(cft, ftype, COLOR_BRRED, COLOR_BLACK, sizeof(cft)), 00946 f->frametype, 00947 term_color(csub, subclass, COLOR_BRCYAN, COLOR_BLACK, sizeof(csub)), 00948 f->subclass, 00949 term_color(cn, name, COLOR_YELLOW, COLOR_BLACK, sizeof(cn))); 00950 }
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 373 of file frame.c.
References __frame_free(), AST_LIST_NEXT, ast_frame::frame_list, and ast_frame::next.
Referenced by mixmonitor_thread().
00374 { 00375 struct ast_frame *next; 00376 00377 for (next = AST_LIST_NEXT(frame, frame_list); 00378 frame; 00379 frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) { 00380 __frame_free(frame, cache); 00381 } 00382 }
Sums two frames of audio samples.
f1 | The first frame (which will contain the result) | |
f2 | The second frame |
Definition at line 1530 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.
01531 { 01532 int count; 01533 short *data1, *data2; 01534 01535 if ((f1->frametype != AST_FRAME_VOICE) || (f1->subclass != AST_FORMAT_SLINEAR)) 01536 return -1; 01537 01538 if ((f2->frametype != AST_FRAME_VOICE) || (f2->subclass != AST_FORMAT_SLINEAR)) 01539 return -1; 01540 01541 if (f1->samples != f2->samples) 01542 return -1; 01543 01544 for (count = 0, data1 = f1->data, data2 = f2->data; 01545 count < f1->samples; 01546 count++, data1++, data2++) 01547 ast_slinear_saturated_add(data1, data2); 01548 01549 return 0; 01550 }
Copies a frame.
fr | frame to copy Duplicates a frame -- should only rarely be used, typically frisolate is good enough |
Definition at line 470 of file frame.c.
References ast_calloc_cache, ast_copy_flags, AST_FRFLAG_HAS_TIMING_INFO, AST_FRIENDLY_OFFSET, AST_LIST_REMOVE_CURRENT, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, AST_MALLOCD_HDR, ast_threadstorage_get(), buf, ast_frame::data, ast_frame::datalen, ast_frame::delivery, f, frame_cache, frames, ast_frame::frametype, ast_frame::len, len(), ast_frame::mallocd, ast_frame::mallocd_hdr_len, ast_frame::offset, ast_frame::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 transmit_audio().
00471 { 00472 struct ast_frame *out = NULL; 00473 int len, srclen = 0; 00474 void *buf = NULL; 00475 00476 #if !defined(LOW_MEMORY) 00477 struct ast_frame_cache *frames; 00478 #endif 00479 00480 /* Start with standard stuff */ 00481 len = sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00482 /* If we have a source, add space for it */ 00483 /* 00484 * XXX Watch out here - if we receive a src which is not terminated 00485 * properly, we can be easily attacked. Should limit the size we deal with. 00486 */ 00487 if (f->src) 00488 srclen = strlen(f->src); 00489 if (srclen > 0) 00490 len += srclen + 1; 00491 00492 #if !defined(LOW_MEMORY) 00493 if ((frames = ast_threadstorage_get(&frame_cache, sizeof(*frames)))) { 00494 AST_LIST_TRAVERSE_SAFE_BEGIN(&frames->list, out, frame_list) { 00495 if (out->mallocd_hdr_len >= len) { 00496 size_t mallocd_len = out->mallocd_hdr_len; 00497 00498 AST_LIST_REMOVE_CURRENT(frame_list); 00499 memset(out, 0, sizeof(*out)); 00500 out->mallocd_hdr_len = mallocd_len; 00501 buf = out; 00502 frames->size--; 00503 break; 00504 } 00505 } 00506 AST_LIST_TRAVERSE_SAFE_END; 00507 } 00508 #endif 00509 00510 if (!buf) { 00511 if (!(buf = ast_calloc_cache(1, len))) 00512 return NULL; 00513 out = buf; 00514 out->mallocd_hdr_len = len; 00515 } 00516 00517 out->frametype = f->frametype; 00518 out->subclass = f->subclass; 00519 out->datalen = f->datalen; 00520 out->samples = f->samples; 00521 out->delivery = f->delivery; 00522 /* Set us as having malloc'd header only, so it will eventually 00523 get freed. */ 00524 out->mallocd = AST_MALLOCD_HDR; 00525 out->offset = AST_FRIENDLY_OFFSET; 00526 if (out->datalen) { 00527 out->data = buf + sizeof(*out) + AST_FRIENDLY_OFFSET; 00528 memcpy(out->data, f->data, out->datalen); 00529 } 00530 if (srclen > 0) { 00531 /* This may seem a little strange, but it's to avoid a gcc (4.2.4) compiler warning */ 00532 char *src; 00533 out->src = buf + sizeof(*out) + AST_FRIENDLY_OFFSET + f->datalen; 00534 src = (char *) out->src; 00535 /* Must have space since we allocated for it */ 00536 strcpy(src, f->src); 00537 } 00538 ast_copy_flags(out, f, AST_FRFLAG_HAS_TIMING_INFO); 00539 out->ts = f->ts; 00540 out->len = f->len; 00541 out->seqno = f->seqno; 00542 return out; 00543 }
Makes a frame independent of any static storage.
fr | frame to act upon Take a frame, and if it's not been malloc'd, make a malloc'd copy and if the data hasn't been malloced then make the data malloc'd. If you need to store frames, say for queueing, then you should call this function. |
Definition at line 389 of file frame.c.
References ast_clear_flag, ast_copy_flags, ast_frame_header_new(), ast_frdup(), ast_free, 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, 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_answer(), ast_slinfactory_feed(), autoservice_run(), and jpeg_read_image().
00390 { 00391 struct ast_frame *out; 00392 void *newdata; 00393 00394 /* if none of the existing frame is malloc'd, let ast_frdup() do it 00395 since it is more efficient 00396 */ 00397 if (fr->mallocd == 0) { 00398 return ast_frdup(fr); 00399 } 00400 00401 /* if everything is already malloc'd, we are done */ 00402 if ((fr->mallocd & (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) == 00403 (AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA)) { 00404 return fr; 00405 } 00406 00407 if (!(fr->mallocd & AST_MALLOCD_HDR)) { 00408 /* Allocate a new header if needed */ 00409 if (!(out = ast_frame_header_new())) { 00410 return NULL; 00411 } 00412 out->frametype = fr->frametype; 00413 out->subclass = fr->subclass; 00414 out->datalen = fr->datalen; 00415 out->samples = fr->samples; 00416 out->offset = fr->offset; 00417 /* Copy the timing data */ 00418 ast_copy_flags(out, fr, AST_FRFLAG_HAS_TIMING_INFO); 00419 if (ast_test_flag(fr, AST_FRFLAG_HAS_TIMING_INFO)) { 00420 out->ts = fr->ts; 00421 out->len = fr->len; 00422 out->seqno = fr->seqno; 00423 } 00424 } else { 00425 ast_clear_flag(fr, AST_FRFLAG_FROM_TRANSLATOR); 00426 ast_clear_flag(fr, AST_FRFLAG_FROM_DSP); 00427 ast_clear_flag(fr, AST_FRFLAG_FROM_FILESTREAM); 00428 out = fr; 00429 } 00430 00431 if (!(fr->mallocd & AST_MALLOCD_SRC) && fr->src) { 00432 if (!(out->src = ast_strdup(fr->src))) { 00433 if (out != fr) { 00434 ast_free(out); 00435 } 00436 return NULL; 00437 } 00438 } else { 00439 out->src = fr->src; 00440 fr->src = NULL; 00441 fr->mallocd &= ~AST_MALLOCD_SRC; 00442 } 00443 00444 if (!(fr->mallocd & AST_MALLOCD_DATA)) { 00445 if (!(newdata = ast_malloc(fr->datalen + AST_FRIENDLY_OFFSET))) { 00446 if (out->src != fr->src) { 00447 ast_free((void *) out->src); 00448 } 00449 if (out != fr) { 00450 ast_free(out); 00451 } 00452 return NULL; 00453 } 00454 newdata += AST_FRIENDLY_OFFSET; 00455 out->offset = AST_FRIENDLY_OFFSET; 00456 out->datalen = fr->datalen; 00457 memcpy(newdata, fr->data, fr->datalen); 00458 out->data = newdata; 00459 } else { 00460 out->data = fr->data; 00461 memset(&fr->data, 0, sizeof(fr->data)); 00462 fr->mallocd &= ~AST_MALLOCD_DATA; 00463 } 00464 00465 out->mallocd = AST_MALLOCD_HDR | AST_MALLOCD_SRC | AST_MALLOCD_DATA; 00466 00467 return out; 00468 }
struct ast_format_list* ast_get_format_list | ( | size_t * | size | ) |
Definition at line 561 of file frame.c.
References ARRAY_LEN, and AST_FORMAT_LIST.
00562 { 00563 *size = ARRAY_LEN(AST_FORMAT_LIST); 00564 return AST_FORMAT_LIST; 00565 }
struct ast_format_list* ast_get_format_list_index | ( | int | index | ) |
Definition at line 556 of file frame.c.
References AST_FORMAT_LIST.
00557 { 00558 return &AST_FORMAT_LIST[index]; 00559 }
int ast_getformatbyname | ( | const char * | name | ) |
Gets a format from a name.
name | string of format |
Definition at line 628 of file frame.c.
References ARRAY_LEN, ast_expand_codec_alias(), AST_FORMAT_LIST, ast_format_list::bits, and format.
Referenced by ast_parse_allow_disallow(), iax_template_parse(), load_moh_classes(), local_ast_moh_start(), reload_config(), and try_suggested_sip_codec().
00629 { 00630 int x, all, format = 0; 00631 00632 all = strcasecmp(name, "all") ? 0 : 1; 00633 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00634 if (all || 00635 !strcasecmp(AST_FORMAT_LIST[x].name,name) || 00636 !strcasecmp(AST_FORMAT_LIST[x].name,ast_expand_codec_alias(name))) { 00637 format |= AST_FORMAT_LIST[x].bits; 00638 if (!all) 00639 break; 00640 } 00641 } 00642 00643 return format; 00644 }
char* ast_getformatname | ( | int | format | ) |
Get the name of a format.
format | id of format |
Definition at line 567 of file frame.c.
References ARRAY_LEN, AST_FORMAT_LIST, ast_format_list::bits, and ast_format_list::name.
Referenced by __ast_play_and_record(), __ast_read(), __ast_register_translator(), _sip_show_peer(), add_codec_to_answer(), add_codec_to_sdp(), add_tcodec_to_sdp(), add_vcodec_to_sdp(), agent_call(), ast_codec_get_len(), ast_codec_get_samples(), ast_codec_pref_string(), ast_dsp_process(), ast_frame_dump(), ast_openvstream(), ast_rtp_write(), ast_slinfactory_feed(), ast_streamfile(), ast_translator_build_path(), ast_unregister_translator(), ast_writestream(), background_detect_exec(), dahdi_read(), do_waiting(), eagi_exec(), func_channel_read(), function_iaxpeer(), function_sippeer(), gtalk_show_channels(), handle_cli_core_show_file_formats(), handle_cli_core_show_translation(), handle_cli_iax2_show_channels(), handle_cli_iax2_show_peer(), handle_cli_moh_show_classes(), handle_core_show_image_formats(), iax2_request(), iax_show_provisioning(), jingle_show_channels(), login_exec(), moh_release(), oh323_rtp_read(), phone_setup(), print_codec_to_cli(), rebuild_matrix(), register_translator(), set_format(), set_local_capabilities(), set_peer_capabilities(), show_codecs(), sip_request_call(), sip_rtp_read(), socket_process(), start_rtp(), unistim_request(), unistim_rtp_read(), and unistim_write().
00568 { 00569 int x; 00570 char *ret = "unknown"; 00571 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00572 if (AST_FORMAT_LIST[x].bits == format) { 00573 ret = AST_FORMAT_LIST[x].name; 00574 break; 00575 } 00576 } 00577 return ret; 00578 }
char* ast_getformatname_multiple | ( | char * | buf, | |
size_t | size, | |||
int | format | |||
) |
Get the names of a set of formats.
buf | a buffer for the output string | |
size | size of buf (bytes) | |
format | the format (combined IDs of codecs) Prints a list of readable codec names corresponding to "format". ex: for format=AST_FORMAT_GSM|AST_FORMAT_SPEEX|AST_FORMAT_ILBC it will return "0x602 (GSM|SPEEX|ILBC)" |
Definition at line 580 of file frame.c.
References ARRAY_LEN, ast_copy_string(), AST_FORMAT_LIST, ast_format_list::bits, len(), and name.
Referenced by __ast_read(), _sip_show_peer(), add_sdp(), ast_streamfile(), function_iaxpeer(), function_sippeer(), gtalk_is_answered(), gtalk_newcall(), handle_cli_iax2_show_peer(), handle_showchan(), handle_skinny_show_line(), process_sdp(), serialize_showchan(), set_format(), show_channels_cb(), sip_new(), sip_request_call(), sip_show_channel(), sip_show_settings(), and sip_write().
00581 { 00582 int x; 00583 unsigned len; 00584 char *start, *end = buf; 00585 00586 if (!size) 00587 return buf; 00588 snprintf(end, size, "0x%x (", format); 00589 len = strlen(end); 00590 end += len; 00591 size -= len; 00592 start = end; 00593 for (x = 0; x < ARRAY_LEN(AST_FORMAT_LIST); x++) { 00594 if (AST_FORMAT_LIST[x].bits & format) { 00595 snprintf(end, size,"%s|",AST_FORMAT_LIST[x].name); 00596 len = strlen(end); 00597 end += len; 00598 size -= len; 00599 } 00600 } 00601 if (start == end) 00602 ast_copy_string(start, "nothing)", size); 00603 else if (size > 1) 00604 *(end -1) = ')'; 00605 return buf; 00606 }
int ast_parse_allow_disallow | ( | struct ast_codec_pref * | pref, | |
int * | mask, | |||
const char * | list, | |||
int | allowing | |||
) |
Parse an "allow" or "deny" line in a channel or device configuration and update the capabilities mask and pref if provided. Video codecs are not added to codec preference lists, since we can not transcode.
Definition at line 1226 of file frame.c.
References ast_codec_pref_append(), ast_codec_pref_remove(), ast_codec_pref_setsize(), ast_debug, AST_FORMAT_AUDIO_MASK, ast_getformatbyname(), ast_log(), ast_strdupa, format, LOG_WARNING, parse(), and strsep().
Referenced by action_originate(), apply_outgoing(), build_device(), build_peer(), build_user(), gtalk_create_member(), gtalk_load_config(), jingle_create_member(), jingle_load_config(), reload_config(), set_config(), and update_common_options().
01227 { 01228 int errors = 0; 01229 char *parse = NULL, *this = NULL, *psize = NULL; 01230 int format = 0, framems = 0; 01231 01232 parse = ast_strdupa(list); 01233 while ((this = strsep(&parse, ","))) { 01234 framems = 0; 01235 if ((psize = strrchr(this, ':'))) { 01236 *psize++ = '\0'; 01237 ast_debug(1, "Packetization for codec: %s is %s\n", this, psize); 01238 framems = atoi(psize); 01239 if (framems < 0) { 01240 framems = 0; 01241 errors++; 01242 ast_log(LOG_WARNING, "Bad packetization value for codec %s\n", this); 01243 } 01244 } 01245 if (!(format = ast_getformatbyname(this))) { 01246 ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this); 01247 errors++; 01248 continue; 01249 } 01250 01251 if (mask) { 01252 if (allowing) 01253 *mask |= format; 01254 else 01255 *mask &= ~format; 01256 } 01257 01258 /* Set up a preference list for audio. Do not include video in preferences 01259 since we can not transcode video and have to use whatever is offered 01260 */ 01261 if (pref && (format & AST_FORMAT_AUDIO_MASK)) { 01262 if (strcasecmp(this, "all")) { 01263 if (allowing) { 01264 ast_codec_pref_append(pref, format); 01265 ast_codec_pref_setsize(pref, format, framems); 01266 } 01267 else 01268 ast_codec_pref_remove(pref, format); 01269 } else if (!allowing) { 01270 memset(pref, 0, sizeof(*pref)); 01271 } 01272 } 01273 } 01274 return errors; 01275 }
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 174 of file frame.c.
References ast_malloc, ast_smoother_reset(), and s.
Referenced by ast_rtp_codec_setpref(), and ast_rtp_write().
00175 { 00176 struct ast_smoother *s; 00177 if (size < 1) 00178 return NULL; 00179 if ((s = ast_malloc(sizeof(*s)))) 00180 ast_smoother_reset(s, size); 00181 return s; 00182 }
struct ast_frame* ast_smoother_read | ( | struct ast_smoother * | s | ) |
Definition at line 234 of file frame.c.
References ast_format_rate(), AST_FRAME_VOICE, AST_FRIENDLY_OFFSET, ast_log(), ast_samp2tv(), AST_SMOOTHER_FLAG_G729, ast_tvadd(), ast_tvzero(), len(), LOG_WARNING, and s.
Referenced by ast_rtp_write().
00235 { 00236 struct ast_frame *opt; 00237 int len; 00238 00239 /* IF we have an optimization frame, send it */ 00240 if (s->opt) { 00241 if (s->opt->offset < AST_FRIENDLY_OFFSET) 00242 ast_log(LOG_WARNING, "Returning a frame of inappropriate offset (%d).\n", 00243 s->opt->offset); 00244 opt = s->opt; 00245 s->opt = NULL; 00246 return opt; 00247 } 00248 00249 /* Make sure we have enough data */ 00250 if (s->len < s->size) { 00251 /* Or, if this is a G.729 frame with VAD on it, send it immediately anyway */ 00252 if (!((s->flags & AST_SMOOTHER_FLAG_G729) && (s->len % 10))) 00253 return NULL; 00254 } 00255 len = s->size; 00256 if (len > s->len) 00257 len = s->len; 00258 /* Make frame */ 00259 s->f.frametype = AST_FRAME_VOICE; 00260 s->f.subclass = s->format; 00261 s->f.data = s->framedata + AST_FRIENDLY_OFFSET; 00262 s->f.offset = AST_FRIENDLY_OFFSET; 00263 s->f.datalen = len; 00264 /* Samples will be improper given VAD, but with VAD the concept really doesn't even exist */ 00265 s->f.samples = len * s->samplesperbyte; /* XXX rounding */ 00266 s->f.delivery = s->delivery; 00267 /* Fill Data */ 00268 memcpy(s->f.data, s->data, len); 00269 s->len -= len; 00270 /* Move remaining data to the front if applicable */ 00271 if (s->len) { 00272 /* In principle this should all be fine because if we are sending 00273 G.729 VAD, the next timestamp will take over anyawy */ 00274 memmove(s->data, s->data + len, s->len); 00275 if (!ast_tvzero(s->delivery)) { 00276 /* If we have delivery time, increment it, otherwise, leave it at 0 */ 00277 s->delivery = ast_tvadd(s->delivery, ast_samp2tv(s->f.samples, ast_format_rate(s->format))); 00278 } 00279 } 00280 /* Return frame */ 00281 return &s->f; 00282 }
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 152 of file frame.c.
References s, and smoother_frame_feed().
Referenced by ast_rtp_codec_setpref().
00153 { 00154 /* if there is no change, then nothing to do */ 00155 if (s->size == bytes) { 00156 return; 00157 } 00158 /* set the new desired output size */ 00159 s->size = bytes; 00160 /* if there is no 'optimized' frame in the smoother, 00161 * then there is nothing left to do 00162 */ 00163 if (!s->opt) { 00164 return; 00165 } 00166 /* there is an 'optimized' frame here at the old size, 00167 * but it must now be put into the buffer so the data 00168 * can be extracted at the new size 00169 */ 00170 smoother_frame_feed(s, s->opt, s->opt_needs_swap); 00171 s->opt = NULL; 00172 }
void ast_smoother_reset | ( | struct ast_smoother * | s, | |
int | bytes | |||
) |
Definition at line 146 of file frame.c.
References s.
Referenced by ast_smoother_new().
00147 { 00148 memset(s, 0, sizeof(*s)); 00149 s->size = bytes; 00150 }
void ast_smoother_set_flags | ( | struct ast_smoother * | smoother, | |
int | flags | |||
) |
Definition at line 189 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 194 of file frame.c.
References s.
Referenced by ast_rtp_write().
00195 { 00196 return (s->flags & flag); 00197 }
void ast_swapcopy_samples | ( | void * | dst, | |
const void * | src, | |||
int | samples | |||
) |
Definition at line 545 of file frame.c.
Referenced by __ast_smoother_feed(), iax_frame_wrap(), phone_write_buf(), and smoother_frame_feed().
00546 { 00547 int i; 00548 unsigned short *dst_s = dst; 00549 const unsigned short *src_s = src; 00550 00551 for (i = 0; i < samples; i++) 00552 dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8); 00553 }
struct ast_frame ast_null_frame |
Queueing a null frame is fairly common, so we declare a global null frame object for this purpose instead of having to declare one on the stack
Definition at line 122 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(), console_read(), features_read(), gtalk_rtp_read(), handle_request_invite(), handle_response_invite(), iax2_read(), jingle_rtp_read(), local_read(), mgcp_rtp_read(), oh323_read(), oh323_rtp_read(), process_rfc2833(), process_sdp(), send_dtmf(), sip_read(), sip_rtp_read(), skinny_rtp_read(), unistim_rtp_read(), and wakeup_sub().