source: mediastreamer2/src/audiostream.c @ 1445:07590391fa90

Last change on this file since 1445:07590391fa90 was 1445:07590391fa90, checked in by Nikita Kozlov <nikita@…>, 11 months ago

enable statistics before creating aec filter
File size: 25.0 KB
Line 
1/*
2mediastreamer2 library - modular sound and video processing and streaming
3Copyright (C) 2006  Simon MORLAT (simon.morlat@linphone.org)
4
5This program is free software; you can redistribute it and/or
6modify it under the terms of the GNU General Public License
7as published by the Free Software Foundation; either version 2
8of the License, or (at your option) any later version.
9
10This program is distributed in the hope that it will be useful,
11but WITHOUT ANY WARRANTY; without even the implied warranty of
12MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
13GNU General Public License for more details.
14
15You should have received a copy of the GNU General Public License
16along with this program; if not, write to the Free Software
17Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
18*/
19
20
21#ifdef HAVE_CONFIG_H
22#include "mediastreamer-config.h"
23#endif
24
25#include "mediastreamer2/mediastream.h"
26
27#include "mediastreamer2/dtmfgen.h"
28#include "mediastreamer2/mssndcard.h"
29#include "mediastreamer2/msrtp.h"
30#include "mediastreamer2/msfileplayer.h"
31#include "mediastreamer2/msfilerec.h"
32#include "mediastreamer2/msvolume.h"
33#include "mediastreamer2/msequalizer.h"
34
35#ifdef INET6
36        #include <sys/types.h>
37#ifndef WIN32
38        #include <sys/socket.h>
39        #include <netdb.h>
40#endif
41#endif
42
43
44#define MAX_RTP_SIZE    1500
45
46
47/* this code is not part of the library itself, it is part of the mediastream program */
48void audio_stream_free(AudioStream *stream)
49{
50        if (stream->session!=NULL) {
51                rtp_session_unregister_event_queue(stream->session,stream->evq);
52                rtp_session_destroy(stream->session);
53        }
54        if (stream->evq) ortp_ev_queue_destroy(stream->evq);
55        if (stream->rtpsend!=NULL) ms_filter_destroy(stream->rtpsend);
56        if (stream->rtprecv!=NULL) ms_filter_destroy(stream->rtprecv);
57        if (stream->soundread!=NULL) ms_filter_destroy(stream->soundread);
58        if (stream->soundwrite!=NULL) ms_filter_destroy(stream->soundwrite);
59        if (stream->encoder!=NULL) ms_filter_destroy(stream->encoder);
60        if (stream->decoder!=NULL) ms_filter_destroy(stream->decoder);
61        if (stream->dtmfgen!=NULL) ms_filter_destroy(stream->dtmfgen);
62        if (stream->ec!=NULL)   ms_filter_destroy(stream->ec);
63        if (stream->volrecv!=NULL) ms_filter_destroy(stream->volrecv);
64        if (stream->volsend!=NULL) ms_filter_destroy(stream->volsend);
65        if (stream->equalizer!=NULL) ms_filter_destroy(stream->equalizer);
66        if (stream->ticker!=NULL) ms_ticker_destroy(stream->ticker);
67        if (stream->read_resampler!=NULL) ms_filter_destroy(stream->read_resampler);
68        if (stream->write_resampler!=NULL) ms_filter_destroy(stream->write_resampler);
69        if (stream->dtmfgen_rtp!=NULL) ms_filter_destroy(stream->dtmfgen_rtp);
70        ms_free(stream);
71}
72
73static int dtmf_tab[16]={'0','1','2','3','4','5','6','7','8','9','*','#','A','B','C','D'};
74
75static void on_dtmf_received(RtpSession *s, int dtmf, void * user_data)
76{
77        AudioStream *stream=(AudioStream*)user_data;
78        if (dtmf>15){
79                ms_warning("Unsupported telephone-event type.");
80                return;
81        }
82        ms_message("Receiving dtmf %c.",dtmf_tab[dtmf]);
83        if (stream->dtmfgen!=NULL && stream->play_dtmfs){
84                ms_filter_call_method(stream->dtmfgen,MS_DTMF_GEN_PUT,&dtmf_tab[dtmf]);
85        }
86}
87
88bool_t ms_is_ipv6(const char *remote){
89        bool_t ret=FALSE;
90#ifdef INET6
91        struct addrinfo hints, *res0;
92
93        int err;
94        memset(&hints, 0, sizeof(hints));
95        hints.ai_family = PF_UNSPEC;
96        hints.ai_socktype = SOCK_DGRAM;
97        err = getaddrinfo(remote,"8000", &hints, &res0);
98        if (err!=0) {
99                ms_warning ("get_local_addr_for: %s", gai_strerror(err));
100                return FALSE;
101        }
102        ret=(res0->ai_addr->sa_family==AF_INET6);
103        freeaddrinfo(res0);
104#endif
105        return ret;
106}
107
108static void audio_stream_configure_resampler(MSFilter *resampler,MSFilter *from,MSFilter *to) {
109        int from_rate=0, to_rate=0;
110        ms_filter_call_method(from,MS_FILTER_GET_SAMPLE_RATE,&from_rate);
111        ms_filter_call_method(to,MS_FILTER_GET_SAMPLE_RATE,&to_rate);
112        ms_filter_call_method(resampler,MS_FILTER_SET_SAMPLE_RATE,&from_rate);
113        ms_filter_call_method(resampler,MS_FILTER_SET_OUTPUT_SAMPLE_RATE,&to_rate);
114        ms_message("configuring %s-->%s from rate[%i] to rate [%i]",
115                   from->desc->name, to->desc->name, from_rate,to_rate);
116}
117
118RtpSession * create_duplex_rtpsession( int locport, bool_t ipv6){
119        RtpSession *rtpr;
120        rtpr=rtp_session_new(RTP_SESSION_SENDRECV);
121        rtp_session_set_recv_buf_size(rtpr,MAX_RTP_SIZE);
122        rtp_session_set_scheduling_mode(rtpr,0);
123        rtp_session_set_blocking_mode(rtpr,0);
124        rtp_session_enable_adaptive_jitter_compensation(rtpr,TRUE);
125        rtp_session_set_symmetric_rtp(rtpr,TRUE);
126#ifndef HAVE_CSL
127        rtp_session_set_local_addr(rtpr,ipv6 ? "::" : "0.0.0.0",locport);
128#endif
129        rtp_session_signal_connect(rtpr,"timestamp_jump",(RtpCallback)rtp_session_resync,(long)NULL);
130        rtp_session_signal_connect(rtpr,"ssrc_changed",(RtpCallback)rtp_session_resync,(long)NULL);
131        rtp_session_set_ssrc_changed_threshold(rtpr,0);
132        return rtpr;
133}
134
135#if defined(_WIN32_WCE)
136time_t
137ms_time (time_t *t)
138{
139    DWORD timemillis = GetTickCount();
140        if (timemillis>0)
141        {
142                if (t!=NULL)
143                        *t = timemillis/1000;
144        }
145        return timemillis/1000;
146}
147#endif
148
149bool_t audio_stream_alive(AudioStream * stream, int timeout){
150        RtpSession *session=stream->session;
151        const rtp_stats_t *stats=rtp_session_get_stats(session);
152        if (stats->recv!=0){
153                if (stream->evq){
154                        OrtpEvent *ev=ortp_ev_queue_get(stream->evq);
155                        if (ev!=NULL){
156                                if (ortp_event_get_type(ev)==ORTP_EVENT_RTCP_PACKET_RECEIVED){
157                                        stream->last_packet_time=ms_time(NULL);
158                                }
159                                ortp_event_destroy(ev);
160                        }
161                }
162                if (stats->recv!=stream->last_packet_count){
163                        stream->last_packet_count=stats->recv;
164                        stream->last_packet_time=ms_time(NULL);
165                }else{
166                        if (ms_time(NULL)-stream->last_packet_time>timeout){
167                                /* more than timeout seconds of inactivity*/
168                                return FALSE;
169                        }
170                }
171        }
172        return TRUE;
173}
174
175/*this function must be called from the MSTicker thread:
176it replaces one filter by another one.
177This is a dirty hack that works anyway.
178It would be interesting to have something that does the job
179simplier within the MSTicker api
180*/
181void audio_stream_change_decoder(AudioStream *stream, int payload){
182        RtpSession *session=stream->session;
183        RtpProfile *prof=rtp_session_get_profile(session);
184        PayloadType *pt=rtp_profile_get_payload(prof,payload);
185        if (pt!=NULL){
186                MSFilter *dec=ms_filter_create_decoder(pt->mime_type);
187                if (dec!=NULL){
188                        ms_filter_unlink(stream->rtprecv, 0, stream->decoder, 0);
189                        ms_filter_unlink(stream->decoder,0,stream->dtmfgen,0);
190                        ms_filter_postprocess(stream->decoder);
191                        ms_filter_destroy(stream->decoder);
192                        stream->decoder=dec;
193                        if (pt->recv_fmtp!=NULL)
194                                ms_filter_call_method(stream->decoder,MS_FILTER_ADD_FMTP,(void*)pt->recv_fmtp);
195                        ms_filter_link (stream->rtprecv, 0, stream->decoder, 0);
196                        ms_filter_link (stream->decoder,0 , stream->dtmfgen, 0);
197                        ms_filter_preprocess(stream->decoder,stream->ticker);
198
199                }else{
200                        ms_warning("No decoder found for %s",pt->mime_type);
201                }
202        }else{
203                ms_warning("No payload defined with number %i",payload);
204        }
205}
206
207static void payload_type_changed(RtpSession *session, unsigned long data){
208        AudioStream *stream=(AudioStream*)data;
209        int pt=rtp_session_get_recv_payload_type(stream->session);
210        audio_stream_change_decoder(stream,pt);
211}
212
213int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char *remip,int remport,
214        int rem_rtcp_port, int payload,int jitt_comp, const char *infile, const char *outfile,
215        MSSndCard *playcard, MSSndCard *captcard, bool_t use_ec)
216{
217        RtpSession *rtps=stream->session;
218        PayloadType *pt;
219        int tmp;
220        MSConnectionHelper h;
221        int sample_rate;
222
223        rtp_session_set_profile(rtps,profile);
224        if (remport>0) rtp_session_set_remote_addr_full(rtps,remip,remport,rem_rtcp_port);
225        rtp_session_set_payload_type(rtps,payload);
226        rtp_session_set_jitter_compensation(rtps,jitt_comp);
227
228        if (remport>0)
229                ms_filter_call_method(stream->rtpsend,MS_RTP_SEND_SET_SESSION,rtps);
230        stream->rtprecv=ms_filter_new(MS_RTP_RECV_ID);
231        ms_filter_call_method(stream->rtprecv,MS_RTP_RECV_SET_SESSION,rtps);
232        stream->session=rtps;
233
234        stream->dtmfgen=ms_filter_new(MS_DTMF_GEN_ID);
235        rtp_session_signal_connect(rtps,"telephone-event",(RtpCallback)on_dtmf_received,(unsigned long)stream);
236        rtp_session_signal_connect(rtps,"payload_type_changed",(RtpCallback)payload_type_changed,(unsigned long)stream);
237        /* creates the local part */
238        if (captcard!=NULL) stream->soundread=ms_snd_card_create_reader(captcard);
239        else {
240                stream->soundread=ms_filter_new(MS_FILE_PLAYER_ID);
241                stream->read_resampler=ms_filter_new(MS_RESAMPLE_ID);
242                if (infile!=NULL) audio_stream_play(stream,infile);
243        }
244        if (playcard!=NULL) stream->soundwrite=ms_snd_card_create_writer(playcard);
245        else {
246                stream->soundwrite=ms_filter_new(MS_FILE_REC_ID);
247                if (outfile!=NULL) audio_stream_record(stream,outfile);
248        }
249
250        /* creates the couple of encoder/decoder */
251        pt=rtp_profile_get_payload(profile,payload);
252        if (pt==NULL){
253                ms_error("audiostream.c: undefined payload type.");
254                return -1;
255        }
256        if (rtp_profile_get_payload_from_mime (profile,"telephone-event")==NULL
257            && ( strcasecmp(pt->mime_type,"pcmu")==0 || strcasecmp(pt->mime_type,"pcma")==0)){
258                /*if no telephone-event payload is usable and pcma or pcmu is used, we will generate
259                  inband dtmf*/
260                stream->dtmfgen_rtp=ms_filter_new (MS_DTMF_GEN_ID);
261        }
262       
263        if (ms_filter_call_method(stream->rtpsend,MS_FILTER_GET_SAMPLE_RATE,&sample_rate)!=0){
264                ms_error("Sample rate is unknown for RTP side !");
265                return -1;
266        }
267       
268        stream->encoder=ms_filter_create_encoder(pt->mime_type);
269        stream->decoder=ms_filter_create_decoder(pt->mime_type);
270        if ((stream->encoder==NULL) || (stream->decoder==NULL)){
271                /* big problem: we have not a registered codec for this payload...*/
272                ms_error("mediastream.c: No decoder available for payload %i.",payload);
273                return -1;
274        }
275
276        stream->volsend=ms_filter_new(MS_VOLUME_ID);
277        stream->volrecv=ms_filter_new(MS_VOLUME_ID);
278        audio_stream_enable_echo_limiter(stream,stream->el_type);
279        audio_stream_enable_noise_gate(stream,stream->use_ng);
280
281        if (stream->use_agc){
282                int tmp=1;
283                if (stream->volsend==NULL)
284                        stream->volsend=ms_filter_new(MS_VOLUME_ID);
285                ms_filter_call_method(stream->volsend,MS_VOLUME_ENABLE_AGC,&tmp);
286        }
287
288        /* give the sound filters some properties */
289        if (ms_filter_call_method(stream->soundread,MS_FILTER_SET_SAMPLE_RATE,&sample_rate) != 0) {
290                /* need to add resampler*/
291                if (stream->read_resampler == NULL) stream->read_resampler=ms_filter_new(MS_RESAMPLE_ID);
292        }
293
294        if (ms_filter_call_method(stream->soundwrite,MS_FILTER_SET_SAMPLE_RATE,&sample_rate) != 0) {
295                /* need to add resampler*/
296                if (stream->write_resampler == NULL) stream->write_resampler=ms_filter_new(MS_RESAMPLE_ID);
297        }
298
299        tmp=1;
300        ms_filter_call_method(stream->soundwrite,MS_FILTER_SET_NCHANNELS, &tmp);
301
302        /*configure the echo canceller if required */
303        if (!use_ec) {
304                ms_filter_destroy(stream->ec);
305                stream->ec=NULL;
306        }
307        if (stream->ec){
308                ms_filter_call_method(stream->ec,MS_FILTER_SET_SAMPLE_RATE,&sample_rate);
309        }
310
311        /* give the encoder/decoder some parameters*/
312        ms_filter_call_method(stream->encoder,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate);
313        ms_message("Payload's bitrate is %i",pt->normal_bitrate);
314        if (pt->normal_bitrate>0){
315                ms_message("Setting audio encoder network bitrate to %i",pt->normal_bitrate);
316                ms_filter_call_method(stream->encoder,MS_FILTER_SET_BITRATE,&pt->normal_bitrate);
317        }
318        ms_filter_call_method(stream->decoder,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate);
319
320        if (pt->send_fmtp!=NULL) ms_filter_call_method(stream->encoder,MS_FILTER_ADD_FMTP, (void*)pt->send_fmtp);
321        if (pt->recv_fmtp!=NULL) ms_filter_call_method(stream->decoder,MS_FILTER_ADD_FMTP,(void*)pt->recv_fmtp);
322
323        /*create the equalizer*/
324        stream->equalizer=ms_filter_new(MS_EQUALIZER_ID);
325        tmp=stream->eq_active;
326        ms_filter_call_method(stream->equalizer,MS_EQUALIZER_SET_ACTIVE,&tmp);
327        /*configure resampler if needed*/
328        if (stream->read_resampler){
329                audio_stream_configure_resampler(stream->read_resampler,stream->soundread,stream->rtpsend);
330        }
331
332        if (stream->write_resampler){
333                audio_stream_configure_resampler(stream->write_resampler,stream->rtprecv,stream->soundwrite);
334        }
335        /* and then connect all */
336        /* tip: draw yourself the picture if you don't understand */
337
338        /*sending graph*/
339        ms_connection_helper_start(&h);
340        ms_connection_helper_link(&h,stream->soundread,-1,0);
341        if (stream->read_resampler)
342                ms_connection_helper_link(&h,stream->read_resampler,0,0);
343        if (stream->ec)
344                ms_connection_helper_link(&h,stream->ec,1,1);
345        if (stream->volsend)
346                ms_connection_helper_link(&h,stream->volsend,0,0);
347        if (stream->dtmfgen_rtp)
348                ms_connection_helper_link(&h,stream->dtmfgen_rtp,0,0);
349        ms_connection_helper_link(&h,stream->encoder,0,0);
350        ms_connection_helper_link(&h,stream->rtpsend,0,-1);
351
352        /*receiving graph*/
353        ms_connection_helper_start(&h);
354        ms_connection_helper_link(&h,stream->rtprecv,-1,0);
355        ms_connection_helper_link(&h,stream->decoder,0,0);
356        ms_connection_helper_link(&h,stream->dtmfgen,0,0);
357        if (stream->equalizer)
358                ms_connection_helper_link(&h,stream->equalizer,0,0);
359        if (stream->volrecv)
360                ms_connection_helper_link(&h,stream->volrecv,0,0);
361        if (stream->ec)
362                ms_connection_helper_link(&h,stream->ec,0,0);
363        if (stream->write_resampler)
364                ms_connection_helper_link(&h,stream->write_resampler,0,0);
365        ms_connection_helper_link(&h,stream->soundwrite,0,-1);
366
367        /* create ticker */
368        stream->ticker=ms_ticker_new();
369        ms_ticker_set_name(stream->ticker,"Audio MSTicker");
370        ms_ticker_attach(stream->ticker,stream->soundread);
371        ms_ticker_attach(stream->ticker,stream->rtprecv);
372
373        return 0;
374}
375
376
377int audio_stream_start_with_files(AudioStream *stream, RtpProfile *prof,const char *remip, int remport,
378        int rem_rtcp_port, int pt,int jitt_comp, const char *infile, const char * outfile)
379{
380        return audio_stream_start_full(stream,prof,remip,remport,rem_rtcp_port,pt,jitt_comp,infile,outfile,NULL,NULL,FALSE);
381}
382
383AudioStream * audio_stream_start(RtpProfile *prof,int locport,const char *remip,int remport,int profile,int jitt_comp,bool_t use_ec)
384{
385        MSSndCard *sndcard_playback;
386        MSSndCard *sndcard_capture;
387        AudioStream *stream;
388        sndcard_capture=ms_snd_card_manager_get_default_capture_card(ms_snd_card_manager_get());
389        sndcard_playback=ms_snd_card_manager_get_default_playback_card(ms_snd_card_manager_get());
390        if (sndcard_capture==NULL || sndcard_playback==NULL)
391                return NULL;
392        stream=audio_stream_new(locport, ms_is_ipv6(remip));
393        if (audio_stream_start_full(stream,prof,remip,remport,remport+1,profile,jitt_comp,NULL,NULL,sndcard_playback,sndcard_capture,use_ec)==0) return stream;
394        audio_stream_free(stream);
395        return NULL;
396}
397
398AudioStream *audio_stream_start_with_sndcards(RtpProfile *prof,int locport,const char *remip,int remport,int profile,int jitt_comp,MSSndCard *playcard, MSSndCard *captcard, bool_t use_ec)
399{
400        AudioStream *stream;
401        if (playcard==NULL) {
402                ms_error("No playback card.");
403                return NULL;
404        }
405        if (captcard==NULL) {
406                ms_error("No capture card.");
407                return NULL;
408        }
409        stream=audio_stream_new(locport, ms_is_ipv6(remip));
410        if (audio_stream_start_full(stream,prof,remip,remport,remport+1,profile,jitt_comp,NULL,NULL,playcard,captcard,use_ec)==0) return stream;
411        audio_stream_free(stream);
412        return NULL;
413}
414
415void audio_stream_set_rtcp_information(AudioStream *st, const char *cname, const char *tool){
416        if (st->session!=NULL){
417                rtp_session_set_source_description(st->session,cname,NULL,NULL,NULL,NULL,tool , "This is free software (GPL) !");
418        }
419}
420
421void audio_stream_play(AudioStream *st, const char *name){
422        if (ms_filter_get_id(st->soundread)==MS_FILE_PLAYER_ID){
423                ms_filter_call_method_noarg(st->soundread,MS_FILE_PLAYER_CLOSE);
424                ms_filter_call_method(st->soundread,MS_FILE_PLAYER_OPEN,(void*)name);
425                if (st->read_resampler){
426                        audio_stream_configure_resampler(st->read_resampler,st->soundread,st->rtpsend);
427                }
428                ms_filter_call_method_noarg(st->soundread,MS_FILE_PLAYER_START);
429        }else{
430                ms_error("Cannot play file: the stream hasn't been started with"
431                " audio_stream_start_with_files");
432        }
433}
434
435void audio_stream_record(AudioStream *st, const char *name){
436        if (ms_filter_get_id(st->soundwrite)==MS_FILE_REC_ID){
437                ms_filter_call_method_noarg(st->soundwrite,MS_FILE_REC_CLOSE);
438                ms_filter_call_method(st->soundwrite,MS_FILE_REC_OPEN,(void*)name);
439                ms_filter_call_method_noarg(st->soundwrite,MS_FILE_REC_START);
440        }else{
441                ms_error("Cannot record file: the stream hasn't been started with"
442                " audio_stream_start_with_files");
443        }
444}
445
446
447AudioStream *audio_stream_new(int locport, bool_t ipv6){
448        AudioStream *stream=(AudioStream *)ms_new0(AudioStream,1);
449        ms_filter_enable_statistics(TRUE);
450        ms_filter_reset_statistics();
451        MSFilterDesc *ec_desc=ms_filter_lookup_by_name("MSOslec");
452
453        stream->session=create_duplex_rtpsession(locport,ipv6);
454        /*some filters are created right now to allow configuration by the application before start() */
455        stream->rtpsend=ms_filter_new(MS_RTP_SEND_ID);
456       
457        if (ec_desc!=NULL)
458                stream->ec=ms_filter_new_from_desc(ec_desc);
459        else
460                stream->ec=ms_filter_new(MS_SPEEX_EC_ID);
461
462        stream->evq=ortp_ev_queue_new();
463        rtp_session_register_event_queue(stream->session,stream->evq);
464        stream->play_dtmfs=TRUE;
465        stream->use_gc=FALSE;
466        stream->use_agc=FALSE;
467        stream->use_ng=FALSE;
468        return stream;
469}
470
471void audio_stream_play_received_dtmfs(AudioStream *st, bool_t yesno){
472        st->play_dtmfs=yesno;
473}
474
475int audio_stream_start_now(AudioStream *stream, RtpProfile * prof,  const char *remip, int remport, int rem_rtcp_port, int payload_type, int jitt_comp, MSSndCard *playcard, MSSndCard *captcard, bool_t use_ec){
476        return audio_stream_start_full(stream,prof,remip,remport,rem_rtcp_port,
477                payload_type,jitt_comp,NULL,NULL,playcard,captcard,use_ec);
478}
479
480void audio_stream_set_relay_session_id(AudioStream *stream, const char *id){
481        ms_filter_call_method(stream->rtpsend, MS_RTP_SEND_SET_RELAY_SESSION_ID,(void*)id);
482}
483
484void audio_stream_set_echo_canceller_params(AudioStream *stream, int tail_len_ms, int delay_ms, int framesize){
485        if (tail_len_ms!=0)
486                ms_filter_call_method(stream->ec,MS_ECHO_CANCELLER_SET_TAIL_LENGTH,&tail_len_ms);
487        if (delay_ms!=0){
488                ms_filter_call_method(stream->ec,MS_ECHO_CANCELLER_SET_DELAY,&delay_ms);
489        }
490        if (framesize!=0)
491                ms_filter_call_method(stream->ec,MS_ECHO_CANCELLER_SET_FRAMESIZE,&framesize);
492}
493
494void audio_stream_enable_echo_limiter(AudioStream *stream, EchoLimiterType type){
495        stream->el_type=type;
496        if (stream->volsend){
497                bool_t enable_noise_gate = stream->el_type==ELControlFull;
498                ms_filter_call_method(stream->volrecv,MS_VOLUME_ENABLE_NOISE_GATE,&enable_noise_gate);
499                ms_filter_call_method(stream->volsend,MS_VOLUME_SET_PEER,type!=ELInactive?stream->volrecv:NULL);
500        } else {
501                ms_warning("cannot set echo limiter to mode [%i] because no volume send",type);
502        }
503}
504
505void audio_stream_enable_gain_control(AudioStream *stream, bool_t val){
506        stream->use_gc=val;
507}
508
509void audio_stream_enable_automatic_gain_control(AudioStream *stream, bool_t val){
510        stream->use_agc=val;
511}
512
513void audio_stream_enable_noise_gate(AudioStream *stream, bool_t val){
514        stream->use_ng=val;
515        if (stream->volsend){
516                ms_filter_call_method(stream->volsend,MS_VOLUME_ENABLE_NOISE_GATE,&val);
517        } else {
518                ms_warning("cannot set noise gate mode to [%i] because no volume send",val);
519        }
520
521
522}
523
524void audio_stream_set_mic_gain(AudioStream *stream, float gain){
525        if (stream->volsend){
526                ms_filter_call_method(stream->volsend,MS_VOLUME_SET_GAIN,&gain);
527        }else ms_warning("Could not apply gain: gain control wasn't activated. "
528                        "Use audio_stream_enable_gain_control() before starting the stream.");
529}
530
531void audio_stream_enable_equalizer(AudioStream *stream, bool_t enabled){
532        stream->eq_active=enabled;
533        if (stream->equalizer){
534                int tmp=enabled;
535                ms_filter_call_method(stream->equalizer,MS_EQUALIZER_SET_ACTIVE,&tmp);
536        }
537}
538
539void audio_stream_equalizer_set_gain(AudioStream *stream, int frequency, float gain, int freq_width){
540        if (stream->equalizer){
541                MSEqualizerGain d;
542                d.frequency=frequency;
543                d.gain=gain;
544                d.width=freq_width;
545                ms_filter_call_method(stream->equalizer,MS_EQUALIZER_SET_GAIN,&d);
546        }
547}
548
549void audio_stream_stop(AudioStream * stream)
550{
551        if (stream->ticker){
552                MSConnectionHelper h;
553                ms_ticker_detach(stream->ticker,stream->soundread);
554                ms_ticker_detach(stream->ticker,stream->rtprecv);
555
556                rtp_stats_display(rtp_session_get_stats(stream->session),"Audio session's RTP statistics");
557
558                /*dismantle the outgoing graph*/
559                ms_connection_helper_start(&h);
560                ms_connection_helper_unlink(&h,stream->soundread,-1,0);
561                if (stream->read_resampler!=NULL)
562                        ms_connection_helper_unlink(&h,stream->read_resampler,0,0);
563                if (stream->ec!=NULL)
564                        ms_connection_helper_unlink(&h,stream->ec,1,1);
565                if (stream->volsend!=NULL)
566                        ms_connection_helper_unlink(&h,stream->volsend,0,0);
567                if (stream->dtmfgen_rtp)
568                        ms_connection_helper_unlink(&h,stream->dtmfgen_rtp,0,0);
569                ms_connection_helper_unlink(&h,stream->encoder,0,0);
570                ms_connection_helper_unlink(&h,stream->rtpsend,0,-1);
571
572                /*dismantle the receiving graph*/
573                ms_connection_helper_start(&h);
574                ms_connection_helper_unlink(&h,stream->rtprecv,-1,0);
575                ms_connection_helper_unlink(&h,stream->decoder,0,0);
576                ms_connection_helper_unlink(&h,stream->dtmfgen,0,0);
577                if (stream->equalizer)
578                        ms_connection_helper_unlink(&h,stream->equalizer,0,0);
579                if (stream->volrecv!=NULL)
580                        ms_connection_helper_unlink(&h,stream->volrecv,0,0);
581                if (stream->ec!=NULL)
582                        ms_connection_helper_unlink(&h,stream->ec,0,0);
583                if (stream->write_resampler!=NULL)
584                        ms_connection_helper_unlink(&h,stream->write_resampler,0,0);
585                ms_connection_helper_unlink(&h,stream->soundwrite,0,-1);
586
587        }
588        audio_stream_free(stream);
589        ms_filter_log_statistics();
590}
591
592RingStream * ring_start(const char *file, int interval, MSSndCard *sndcard){
593   return ring_start_with_cb(file,interval,sndcard,NULL,NULL);
594}
595
596RingStream * ring_start_with_cb(const char *file,int interval,MSSndCard *sndcard, MSFilterNotifyFunc func,void * user_data)
597{
598        RingStream *stream;
599        int tmp;
600        int srcrate,dstrate;
601        MSConnectionHelper h;
602
603        stream=(RingStream *)ms_new0(RingStream,1);
604        stream->source=ms_filter_new(MS_FILE_PLAYER_ID);
605        if (file)
606                ms_filter_call_method(stream->source,MS_FILE_PLAYER_OPEN,(void*)file);
607       
608        ms_filter_call_method(stream->source,MS_FILE_PLAYER_LOOP,&interval);
609        ms_filter_call_method_noarg(stream->source,MS_FILE_PLAYER_START);
610        if (func!=NULL)
611                ms_filter_set_notify_callback(stream->source,func,user_data);
612        stream->gendtmf=ms_filter_new(MS_DTMF_GEN_ID);
613       
614       
615        stream->sndwrite=ms_snd_card_create_writer(sndcard);
616        ms_filter_call_method(stream->source,MS_FILTER_GET_SAMPLE_RATE,&srcrate);
617        ms_filter_call_method(stream->gendtmf,MS_FILTER_SET_SAMPLE_RATE,&srcrate);
618        ms_filter_call_method(stream->sndwrite,MS_FILTER_SET_SAMPLE_RATE,&srcrate);
619        ms_filter_call_method(stream->sndwrite,MS_FILTER_GET_SAMPLE_RATE,&dstrate);
620        if (srcrate!=dstrate){
621                stream->write_resampler=ms_filter_new(MS_RESAMPLE_ID);
622                ms_filter_call_method(stream->write_resampler,MS_FILTER_SET_SAMPLE_RATE,&srcrate);
623                ms_filter_call_method(stream->write_resampler,MS_FILTER_SET_OUTPUT_SAMPLE_RATE,&dstrate);
624                ms_message("configuring resampler from rate[%i] to rate [%i]", srcrate,dstrate);
625        }
626        ms_filter_call_method(stream->source,MS_FILTER_GET_NCHANNELS,&tmp);
627        ms_filter_call_method(stream->gendtmf,MS_FILTER_SET_NCHANNELS,&tmp);
628        ms_filter_call_method(stream->sndwrite,MS_FILTER_SET_NCHANNELS,&tmp);
629
630        stream->volume = ms_filter_new(MS_VOLUME_ID);
631       
632        stream->ticker=ms_ticker_new();
633       
634        ms_ticker_set_name(stream->ticker,"Audio (ring) MSTicker");
635
636        ms_connection_helper_start(&h);
637        ms_connection_helper_link(&h,stream->source,-1,0);
638        ms_connection_helper_link(&h,stream->volume,0,0);
639        ms_connection_helper_link(&h,stream->gendtmf,0,0);
640        if (stream->write_resampler)
641                ms_connection_helper_link(&h,stream->write_resampler,0,0);
642        ms_connection_helper_link(&h,stream->sndwrite,0,-1);
643        ms_ticker_attach(stream->ticker,stream->source);
644       
645        return stream;
646}
647
648void ring_play_dtmf(RingStream *stream, char dtmf, int duration_ms){
649        if (duration_ms>0)
650                ms_filter_call_method(stream->gendtmf, MS_DTMF_GEN_PLAY, &dtmf);
651        else ms_filter_call_method(stream->gendtmf, MS_DTMF_GEN_START, &dtmf);
652}
653
654void ring_stop_dtmf(RingStream *stream){
655        ms_filter_call_method_noarg(stream->gendtmf, MS_DTMF_GEN_STOP);
656}
657
658void ring_stop(RingStream *stream){
659        MSConnectionHelper h;
660        ms_ticker_detach(stream->ticker,stream->source);
661
662        ms_connection_helper_start(&h);
663        ms_connection_helper_unlink(&h,stream->source,-1,0);
664        ms_connection_helper_unlink(&h,stream->gendtmf,0,0);
665        if (stream->write_resampler)
666                ms_connection_helper_unlink(&h,stream->write_resampler,0,0);
667        ms_connection_helper_unlink(&h,stream->sndwrite,0,-1);
668
669        ms_ticker_destroy(stream->ticker);
670        ms_filter_destroy(stream->source);
671        ms_filter_destroy(stream->volume);
672        ms_filter_destroy(stream->gendtmf);
673        ms_filter_destroy(stream->sndwrite);
674        ms_free(stream);
675#ifdef _WIN32_WCE
676        ms_warning("Sleeping a bit after closing the audio device...");
677        ms_sleep(1);
678#endif
679}
680
681
682int audio_stream_send_dtmf(AudioStream *stream, char dtmf)
683{
684        if (stream->dtmfgen_rtp)
685                ms_filter_call_method(stream->dtmfgen_rtp,MS_DTMF_GEN_PLAY,&dtmf);
686        else if (stream->rtpsend)
687                ms_filter_call_method(stream->rtpsend,MS_RTP_SEND_SEND_DTMF,&dtmf);
688        return 0;
689}
690
691void audio_stream_get_local_rtp_stats(AudioStream *stream, rtp_stats_t *lstats){
692        if (stream->session){
693                const rtp_stats_t *stats=rtp_session_get_stats(stream->session);
694                memcpy(lstats,stats,sizeof(*stats));
695        }else memset(lstats,0,sizeof(rtp_stats_t));
696}
697
698
699void audio_stream_mute_rtp(AudioStream *stream, bool_t val) 
700{
701  if (stream->rtpsend){
702    if (val)
703      ms_filter_call_method(stream->rtpsend,MS_RTP_SEND_MUTE_MIC,&val);
704    else
705      ms_filter_call_method(stream->rtpsend,MS_RTP_SEND_UNMUTE_MIC,&val);
706  }
707}
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