Merge branch 'master' of git.linphone.org:mediastreamer2

build/win32native/Makefile.am

-EXTRA_DIST=alldescs.h mediastreamer2.vcproj mediastream.vcproj videodisplay.vcproj mediastreamer2.def 
+EXTRA_DIST=alldescs.h mediastreamer2.vcproj  videodisplay.vcproj mediastreamer2.def 
 

include/mediastreamer2/dtmfgen.h

@@ -24,6 +24,12 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
 
 #define MS_DTMF_GEN_PUT		MS_FILTER_METHOD(MS_DTMF_GEN_ID,0,const char)
 
+#define MS_DTMF_GEN_PLAY		MS_FILTER_METHOD(MS_DTMF_GEN_ID,0,const char) /*alias to put*/
+/**Start playing a dtmf */
+#define MS_DTMF_GEN_START		MS_FILTER_METHOD(MS_DTMF_GEN_ID,1,const char)
+/**Stop currently played dtmf*/
+#define MS_DTMF_GEN_STOP		MS_FILTER_METHOD_NO_ARG(MS_DTMF_GEN_ID,2)
+
 extern MSFilterDesc ms_dtmf_gen_desc;
 
 #endif

include/mediastreamer2/mediastream.h

@@ -53,6 +53,7 @@ struct _AudioStream
 	MSFilter *rtprecv;
 	MSFilter *rtpsend;
 	MSFilter *dtmfgen;
+	MSFilter *dtmfgen_rtp;
 	MSFilter *ec;/*echo canceler*/
 	MSFilter *volsend,*volrecv; /*MSVolumes*/
 	MSFilter *read_resampler;
@@ -64,6 +65,7 @@ struct _AudioStream
 	int ec_tail_len; /*milliseconds*/
 	int ec_delay;	/*milliseconds*/
 	int ec_framesize; /* number of fft points */
+	OrtpEvQueue *evq;
 	bool_t play_dtmfs;
 	bool_t use_gc;
 	bool_t use_agc;
@@ -81,6 +83,7 @@ struct _RingStream
 {
 	MSTicker *ticker;
 	MSFilter *source;
+	MSFilter *gendtmf;
 	MSFilter *sndwrite;
 };
 
@@ -99,6 +102,10 @@ int audio_stream_start_with_files (AudioStream * stream, RtpProfile * prof,
 					    int pt, int jitt_comp,
 					    const char * infile,  const char * outfile);
 
+int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char *remip,int remport,
+	int rem_rtcp_port, int payload,int jitt_comp, const char *infile, const char *outfile,
+	MSSndCard *playcard, MSSndCard *captcard, bool_t use_ec);
+
 void audio_stream_play(AudioStream *st, const char *name);
 void audio_stream_record(AudioStream *st, const char *name);
 
@@ -163,6 +170,12 @@ void audio_stream_get_local_rtp_stats(AudioStream *stream, rtp_stats_t *stats);
 
 typedef void (*VideoStreamRenderCallback)(void *user_pointer, const MSPicture *local_view, const MSPicture *remote_view);
 
+typedef enum _VideoStreamDir{
+	VideoStreamSendRecv,
+	VideoStreamSendOnly,
+	VideoStreamRecvOnly
+}VideoStreamDir;
+
 struct _VideoStream
 {
 	MSTicker *ticker;
@@ -182,17 +195,23 @@ struct _VideoStream
 	VideoStreamRenderCallback rendercb;
 	void *render_pointer;
 	char *display_name;
+	VideoStreamDir dir;
 	bool_t adapt_bitrate;
 };
 
 typedef struct _VideoStream VideoStream;
 
+
+
 VideoStream *video_stream_new(int locport, bool_t use_ipv6);
+void video_stream_set_direction(VideoStream *vs, VideoStreamDir dir);
 void video_stream_enable_adaptive_bitrate_control(VideoStream *s, bool_t yesno);
 void video_stream_set_render_callback(VideoStream *s, VideoStreamRenderCallback cb, void *user_pointer);
 void video_stream_set_display_filter_name(VideoStream *s, const char *fname);
 int video_stream_start(VideoStream * stream, RtpProfile *profile, const char *remip, int remport, int rem_rtcp_port,
 		int payload, int jitt_comp, MSWebCam *device);
+
+
 void video_stream_set_relay_session_id(VideoStream *stream, const char *relay_session_id);
 void video_stream_set_rtcp_information(VideoStream *st, const char *cname, const char *tool);
 /*function to call periodically to handle various events */
@@ -203,17 +222,17 @@ void video_stream_set_sent_video_size(VideoStream *stream, MSVideoSize vsize);
 void video_stream_enable_self_view(VideoStream *stream, bool_t val);
 unsigned long video_stream_get_native_window_id(VideoStream *stream);
 
+/*provided for compatibility, use video_stream_set_direction() instead */
+int video_stream_recv_only_start(VideoStream *videostream, RtpProfile *profile, const char *addr, int port, int used_pt, int jitt_comp);
+int video_stream_send_only_start(VideoStream *videostream,
+				RtpProfile *profile, const char *addr, int port, int rtcp_port, 
+				int used_pt, int  jitt_comp, MSWebCam *device);
+void video_stream_recv_only_stop(VideoStream *vs);
+void video_stream_send_only_stop(VideoStream *vs);
 
 VideoStream * video_preview_start(MSWebCam *device, MSVideoSize vsize);
 void video_preview_stop(VideoStream *stream);
 
-int video_stream_recv_only_start(VideoStream * stream, RtpProfile *profile, const char *remip, int remport, int payload, int jitt_comp);
-int video_stream_send_only_start(VideoStream * stream, RtpProfile *profile, const char *remip, int remport,
-		int rem_rtcp_port, int payload, int jitt_comp, MSWebCam *device);
-void video_stream_recv_only_stop(VideoStream *stream);
-void video_stream_send_only_stop(VideoStream *stream);
-
-
 bool_t ms_is_ipv6(const char *address);
 
 #ifdef __cplusplus

src/audiostream.c

@@ -47,7 +47,11 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
 /* this code is not part of the library itself, it is part of the mediastream program */
 void audio_stream_free(AudioStream *stream)
 {
-	if (stream->session!=NULL) rtp_session_destroy(stream->session);
+	if (stream->session!=NULL) {
+		rtp_session_unregister_event_queue(stream->session,stream->evq);
+		rtp_session_destroy(stream->session);
+	}
+	if (stream->evq) ortp_ev_queue_destroy(stream->evq);
 	if (stream->rtpsend!=NULL) ms_filter_destroy(stream->rtpsend);
 	if (stream->rtprecv!=NULL) ms_filter_destroy(stream->rtprecv);
 	if (stream->soundread!=NULL) ms_filter_destroy(stream->soundread);
@@ -62,6 +66,7 @@ void audio_stream_free(AudioStream *stream)
 	if (stream->ticker!=NULL) ms_ticker_destroy(stream->ticker);
 	if (stream->read_resampler!=NULL) ms_filter_destroy(stream->read_resampler);
 	if (stream->write_resampler!=NULL) ms_filter_destroy(stream->write_resampler);
+	if (stream->dtmfgen_rtp!=NULL) ms_filter_destroy(stream->dtmfgen_rtp);
 	ms_free(stream);
 }
 
@@ -142,6 +147,15 @@ bool_t audio_stream_alive(AudioStream * stream, int timeout){
 	RtpSession *session=stream->session;
 	const rtp_stats_t *stats=rtp_session_get_stats(session);
 	if (stats->recv!=0){
+		if (stream->evq){
+			OrtpEvent *ev=ortp_ev_queue_get(stream->evq);
+			if (ev!=NULL){
+				if (ortp_event_get_type(ev)==ORTP_EVENT_RTCP_PACKET_RECEIVED){
+					stream->last_packet_time=ms_time(NULL);
+				}
+				ortp_event_destroy(ev);
+			}
+		}
 		if (stats->recv!=stream->last_packet_count){
 			stream->last_packet_count=stats->recv;
 			stream->last_packet_time=ms_time(NULL);
@@ -193,7 +207,6 @@ static void payload_type_changed(RtpSession *session, unsigned long data){
 	audio_stream_change_decoder(stream,pt);
 }
 
-
 int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char *remip,int remport,
 	int rem_rtcp_port, int payload,int jitt_comp, const char *infile, const char *outfile,
 	MSSndCard *playcard, MSSndCard *captcard, bool_t use_ec)
@@ -202,6 +215,7 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
 	PayloadType *pt;
 	int tmp;
 	MSConnectionHelper h;
+	int sample_rate;
 
 	rtp_session_set_profile(rtps,profile);
 	if (remport>0) rtp_session_set_remote_addr_full(rtps,remip,remport,rem_rtcp_port);
@@ -217,7 +231,7 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
 	stream->dtmfgen=ms_filter_new(MS_DTMF_GEN_ID);
 	rtp_session_signal_connect(rtps,"telephone-event",(RtpCallback)on_dtmf_received,(unsigned long)stream);
 	rtp_session_signal_connect(rtps,"payload_type_changed",(RtpCallback)payload_type_changed,(unsigned long)stream);
-
+	rtp_session_signal_connect(rtps,"payload_type_changed",(RtpCallback)payload_type_changed,(unsigned long)stream);
 	/* creates the local part */
 	if (captcard!=NULL) stream->soundread=ms_snd_card_create_reader(captcard);
 	else {
@@ -237,6 +251,18 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
 		ms_error("audiostream.c: undefined payload type.");
 		return -1;
 	}
+	if (rtp_profile_get_payload_from_mime (profile,"telephone-event")==NULL
+	    && ( strcasecmp(pt->mime_type,"pcmu")==0 || strcasecmp(pt->mime_type,"pcma")==0)){
+		/*if no telephone-event payload is usable and pcma or pcmu is used, we will generate
+		  inband dtmf*/
+		stream->dtmfgen_rtp=ms_filter_new (MS_DTMF_GEN_ID);
+	}
+	
+	if (ms_filter_call_method(stream->rtpsend,MS_FILTER_GET_SAMPLE_RATE,&sample_rate)!=0){
+		ms_error("Sample rate is unknown for RTP side !");
+		return -1;
+	}
+	
 	stream->encoder=ms_filter_create_encoder(pt->mime_type);
 	stream->decoder=ms_filter_create_decoder(pt->mime_type);
 	if ((stream->encoder==NULL) || (stream->decoder==NULL)){
@@ -270,12 +296,12 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
 	}
 
 	/* give the sound filters some properties */
-	if (ms_filter_call_method(stream->soundread,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate) != 0) {
+	if (ms_filter_call_method(stream->soundread,MS_FILTER_SET_SAMPLE_RATE,&sample_rate) != 0) {
 		/* need to add resampler*/
 		if (stream->read_resampler == NULL) stream->read_resampler=ms_filter_new(MS_RESAMPLE_ID);
 	}
 
-	if (ms_filter_call_method(stream->soundwrite,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate) != 0) {
+	if (ms_filter_call_method(stream->soundwrite,MS_FILTER_SET_SAMPLE_RATE,&sample_rate) != 0) {
 		/* need to add resampler*/
 		if (stream->write_resampler == NULL) stream->write_resampler=ms_filter_new(MS_RESAMPLE_ID);
 	}
@@ -286,7 +312,7 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
 	/*configure the echo canceller if required */
 	if (use_ec) {
 		stream->ec=ms_filter_new(MS_SPEEX_EC_ID);
-		ms_filter_call_method(stream->ec,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate);
+		ms_filter_call_method(stream->ec,MS_FILTER_SET_SAMPLE_RATE,&sample_rate);
 		if (stream->ec_tail_len!=0)
 			ms_filter_call_method(stream->ec,MS_ECHO_CANCELLER_SET_TAIL_LENGTH,&stream->ec_tail_len);
 		if (stream->ec_delay!=0){
@@ -339,6 +365,8 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
 		ms_connection_helper_link(&h,stream->ec,1,1);
 	if (stream->volsend)
 		ms_connection_helper_link(&h,stream->volsend,0,0);
+	if (stream->dtmfgen_rtp)
+		ms_connection_helper_link(&h,stream->dtmfgen_rtp,0,0);
 	ms_connection_helper_link(&h,stream->encoder,0,0);
 	ms_connection_helper_link(&h,stream->rtpsend,0,-1);
 
@@ -441,6 +469,8 @@ AudioStream *audio_stream_new(int locport, bool_t ipv6){
 	AudioStream *stream=(AudioStream *)ms_new0(AudioStream,1);
 	stream->session=create_duplex_rtpsession(locport,ipv6);
 	stream->rtpsend=ms_filter_new(MS_RTP_SEND_ID);
+	stream->evq=ortp_ev_queue_new();
+	rtp_session_register_event_queue(stream->session,stream->evq);
 	stream->play_dtmfs=TRUE;
 	stream->use_gc=FALSE;
 	stream->use_agc=FALSE;
@@ -558,32 +588,48 @@ RingStream * ring_start_with_cb(const char *file,int interval,MSSndCard *sndcard
 	int tmp;
 	stream=(RingStream *)ms_new0(RingStream,1);
 	stream->source=ms_filter_new(MS_FILE_PLAYER_ID);
-	if (ms_filter_call_method(stream->source,MS_FILE_PLAYER_OPEN,(void*)file)<0){
-		ms_filter_destroy(stream->source);
-		ms_free(stream);
-		return NULL;
-	}
+	if (file)
+		ms_filter_call_method(stream->source,MS_FILE_PLAYER_OPEN,(void*)file);
+	
 	ms_filter_call_method(stream->source,MS_FILE_PLAYER_LOOP,&interval);
 	ms_filter_call_method_noarg(stream->source,MS_FILE_PLAYER_START);
 	if (func!=NULL)
 		ms_filter_set_notify_callback(stream->source,func,user_data);
+	stream->gendtmf=ms_filter_new(MS_DTMF_GEN_ID);
+	
+	
 	stream->sndwrite=ms_snd_card_create_writer(sndcard);
 	ms_filter_call_method(stream->source,MS_FILTER_GET_SAMPLE_RATE,&tmp);
+	ms_filter_call_method(stream->gendtmf,MS_FILTER_SET_SAMPLE_RATE,&tmp);
 	ms_filter_call_method(stream->sndwrite,MS_FILTER_SET_SAMPLE_RATE,&tmp);
 	ms_filter_call_method(stream->source,MS_FILTER_GET_NCHANNELS,&tmp);
+	ms_filter_call_method(stream->gendtmf,MS_FILTER_SET_NCHANNELS,&tmp);
 	ms_filter_call_method(stream->sndwrite,MS_FILTER_SET_NCHANNELS,&tmp);
 	stream->ticker=ms_ticker_new();
 	ms_ticker_set_name(stream->ticker,"Audio (ring) MSTicker");
-	ms_filter_link(stream->source,0,stream->sndwrite,0);
+	ms_filter_link(stream->source,0,stream->gendtmf,0);
+	ms_filter_link(stream->gendtmf,0,stream->sndwrite,0);
 	ms_ticker_attach(stream->ticker,stream->source);
 	return stream;
 }
 
+void ring_play_dtmf(RingStream *stream, char dtmf, int duration_ms){
+	if (duration_ms>0)
+		ms_filter_call_method(stream->gendtmf, MS_DTMF_GEN_PLAY, &dtmf);
+	else ms_filter_call_method(stream->gendtmf, MS_DTMF_GEN_START, &dtmf);
+}
+
+void ring_stop_dtmf(RingStream *stream){
+	ms_filter_call_method_noarg(stream->gendtmf, MS_DTMF_GEN_STOP);
+}
+
 void ring_stop(RingStream *stream){
 	ms_ticker_detach(stream->ticker,stream->source);
-	ms_filter_unlink(stream->source,0,stream->sndwrite,0);
+	ms_filter_unlink(stream->source,0,stream->gendtmf,0);
+	ms_filter_unlink(stream->gendtmf,0,stream->sndwrite,0);
 	ms_ticker_destroy(stream->ticker);
 	ms_filter_destroy(stream->source);
+	ms_filter_destroy(stream->gendtmf);
 	ms_filter_destroy(stream->sndwrite);
 	ms_free(stream);
 #ifdef _WIN32_WCE
@@ -595,10 +641,10 @@ void ring_stop(RingStream *stream){
 
 int audio_stream_send_dtmf(AudioStream *stream, char dtmf)
 {
-	if (stream->rtpsend)
+	if (stream->dtmfgen_rtp)
+		ms_filter_call_method(stream->dtmfgen_rtp,MS_DTMF_GEN_PLAY,&dtmf);
+	else if (stream->rtpsend)
 		ms_filter_call_method(stream->rtpsend,MS_RTP_SEND_SEND_DTMF,&dtmf);
-	if (stream->dtmfgen)
-		ms_filter_call_method(stream->dtmfgen,MS_DTMF_GEN_PUT,&dtmf);
 	return 0;
 }
 

src/dtmfgen.c

@@ -18,6 +18,7 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
 */
 
 #include "mediastreamer2/dtmfgen.h"
+#include "mediastreamer2/msticker.h"
 
 
 #include <math.h>
@@ -26,12 +27,17 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
 #define M_PI       3.14159265358979323846
 #endif
 
+#define NO_SAMPLES_THRESHOLD 100 /*ms*/
+#define TRAILLING_SILENCE 500 /*ms*/
+
 struct DtmfGenState{
 	int rate;
 	int dur;
 	int pos;
 	float highfreq;
 	float lowfreq;
+	int nosamples_time;
+	int silence;
 	char dtmf;
 };
 
@@ -43,6 +49,8 @@ static void dtmfgen_init(MSFilter *f){
 	s->dur=s->rate/10;
 	s->pos=0;
 	s->dtmf=0;
+	s->nosamples_time=0;
+	s->silence=0;
 	f->data=s;
 }
 
@@ -123,45 +131,94 @@ static int dtmfgen_put(MSFilter *f, void *arg){
 			ms_warning("Not a dtmf key.");
 			return -1;
 	}
+	ms_filter_lock(f);
 	s->lowfreq=s->lowfreq/s->rate;
 	s->highfreq=s->highfreq/s->rate;
-
+	s->dur=s->rate/10; /*100 ms duration */
+	s->silence=0;
 	s->dtmf=dtmf[0];
+	ms_filter_unlock(f);
+	return 0;
+}
+
+static int dtmfgen_start(MSFilter *f, void *arg){
+	if (dtmfgen_put(f,arg)==0){
+		DtmfGenState *s=(DtmfGenState*)f->data;
+		s->dur=5*s->rate;
+		return 0;
+	}
+	return -1;
+}
+
+static int dtmfgen_stop(MSFilter *f, void *arg){
+	DtmfGenState *s=(DtmfGenState*)f->data;
+	s->dtmf=0;
 	return 0;
 }
 
 static int dtmfgen_set_rate(MSFilter *f, void *arg){
 	DtmfGenState *s=(DtmfGenState*)f->data;
 	s->rate=*((int*)arg);
-	s->dur=s->rate/10;
+	
 	return 0;
 }
 
+
+static void write_dtmf(DtmfGenState *s , int16_t *sample, int nsamples){
+	int i;
+	for (i=0;i<nsamples && s->pos<s->dur;i++,s->pos++){
+		sample[i]=(int16_t)(10000.0*sin(2*M_PI*(float)s->pos*s->lowfreq));
+		sample[i]+=(int16_t)(10000.0*sin(2*M_PI*(float)s->pos*s->highfreq));
+	}
+	if (s->pos==s->dur){
+		s->pos=0;
+		s->dtmf=0;
+		s->silence=TRAILLING_SILENCE;
+	}
+}
+
 static void dtmfgen_process(MSFilter *f){
 	mblk_t *m;
 	DtmfGenState *s=(DtmfGenState*)f->data;
+	int nsamples;
 
-	while((m=ms_queue_get(f->inputs[0]))!=NULL){
-		if (s->dtmf!=0){
-			int nsamples=(m->b_wptr-m->b_rptr)/2;
-			int i;
-			int16_t *sample=(int16_t*)m->b_rptr;
-			for (i=0;i<nsamples && s->pos<s->dur;i++,s->pos++){
-				sample[i]=(int16_t)(10000.0*sin(2*M_PI*(float)s->pos*s->lowfreq));
-				sample[i]+=(int16_t)(10000.0*sin(2*M_PI*(float)s->pos*s->highfreq));
+	ms_filter_lock(f);
+	if (ms_queue_empty(f->inputs[0])){
+		s->nosamples_time+=f->ticker->interval;
+		if ((s->dtmf!=0 || s->silence!=0) && s->nosamples_time>NO_SAMPLES_THRESHOLD){
+			/*after 100 ms without stream we decide to generate our own sample
+			 instead of writing into incoming stream samples*/
+			nsamples=(f->ticker->interval*s->rate)/1000;
+			m=allocb(nsamples*2,0);
+			if (s->silence==0){
+				write_dtmf(s,(int16_t*)m->b_wptr,nsamples);
+			}else{
+				memset(m->b_wptr,0,nsamples*2);
+				s->silence-=f->ticker->interval;
+				if (s->silence<0) s->silence=0;
 			}
-			if (s->pos==s->dur){
-				s->pos=0;
-				s->dtmf=0;
+			m->b_wptr+=nsamples*2;
+			ms_queue_put(f->outputs[0],m);
+		}
+	}else{
+		s->nosamples_time=0;
+		s->silence=0;
+		while((m=ms_queue_get(f->inputs[0]))!=NULL){
+			if (s->dtmf!=0){
+				nsamples=(m->b_wptr-m->b_rptr)/2;
+				write_dtmf(s, (int16_t*)m->b_rptr,nsamples);
 			}
+			ms_queue_put(f->outputs[0],m);
 		}
-		ms_queue_put(f->outputs[0],m);
 	}
+	ms_filter_unlock(f);
 }
 
 MSFilterMethod dtmfgen_methods[]={
 	{	MS_FILTER_SET_SAMPLE_RATE	,	dtmfgen_set_rate	},
-	{	MS_DTMF_GEN_PUT			,	dtmfgen_put		},
+	{	MS_DTMF_GEN_PLAY			,	dtmfgen_put		},
+	{  MS_DTMF_GEN_START		,   dtmfgen_start },
+	{  MS_DTMF_GEN_STOP		, 	dtmfgen_stop },
 	{	0				,	NULL			}
 };
 
@@ -180,7 +237,8 @@ MSFilterDesc ms_dtmf_gen_desc={
     dtmfgen_process,
 	NULL,
     dtmfgen_uninit,
-	dtmfgen_methods
+	dtmfgen_methods,
+	MS_FILTER_IS_PUMP
 };
 
 #else
@@ -195,7 +253,8 @@ MSFilterDesc ms_dtmf_gen_desc={
 	.init=dtmfgen_init,
 	.process=dtmfgen_process,
 	.uninit=dtmfgen_uninit,
-	.methods=dtmfgen_methods
+	.methods=dtmfgen_methods,
+	.flags=MS_FILTER_IS_PUMP
 };
 
 #endif

src/msrtp.c

@@ -30,6 +30,8 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
 struct SenderData {
 	RtpSession *session;
 	uint32_t tsoff;
+	uint32_t last_ts;
+	int64_t last_sent_time;
 	uint32_t skip_until;
 	int rate;
 	char dtmf;
@@ -59,6 +61,8 @@ static void sender_init(MSFilter * f)
 	d->mute_mic=FALSE;
 	d->relay_session_id_size=0;
 	d->last_rsi_time=0;
+	d->last_sent_time=-1;
+	d->last_ts=0;
 	f->data = d;
 }
 
@@ -100,9 +104,7 @@ static int sender_set_session(MSFilter * f, void *arg)
 		rtp_profile_get_payload(rtp_session_get_profile(s),
 								rtp_session_get_send_payload_type(s));
 	if (pt != NULL) {
-		if (strcasecmp("g722", pt->mime_type)==0 )
-			d->rate=8000;
-		else d->rate = pt->clock_rate;
+		d->rate = pt->clock_rate;
 	} else {
 		ms_warning("Sending undefined payload type ?");
 	}
@@ -137,7 +139,22 @@ static int sender_set_relay_session_id(MSFilter *f, void*arg){
 
 static int sender_get_sr(MSFilter *f, void *arg){
 	SenderData *d = (SenderData *) f->data;
-	*(int*)arg=d->rate;
+	PayloadType *pt;
+	if (d->session==NULL) {
+		ms_warning("Could not obtain sample rate, session is not set.");
+		return -1;
+	}
+	pt=rtp_profile_get_payload(rtp_session_get_profile(d->session),
+									rtp_session_get_recv_payload_type(d->session));
+	if (pt != NULL) {
+		if (strcasecmp(pt->mime_type,"G722")==0)
+			*(int*)arg=16000;
+		else
+			*(int*)arg=pt->clock_rate;
+	}else{
+		ms_warning("MSRtpSend: Could not obtain sample rate, payload type is unknown.");
+		return -1;
+	}
 	return 0;
 }
 
@@ -146,32 +163,26 @@ static uint32_t get_cur_timestamp(MSFilter * f, uint32_t packet_ts)
 {
 	SenderData *d = (SenderData *) f->data;
 	uint32_t curts = (uint32_t)( (f->ticker->time*(uint64_t)d->rate)/(uint64_t)1000) ;
-	int diff;
-	int delta = d->rate / 50;	/*20 ms at 8000Hz */
+	int diffts;
 	uint32_t netts;
+	int difftime_ts;
 
-	netts = packet_ts + d->tsoff;
-	diff = curts - netts;
-
-#ifdef AMD_HACK
-	if (diff > delta) {
-		d->tsoff = curts - packet_ts;
-		netts = packet_ts + d->tsoff;
-		ms_message("synchronizing timestamp, diff=%i", diff);
-	}
-	else if (diff < -delta) {
-		/* d->tsoff = curts - packet_ts; */
-		/* hardware clock is going slower than sound card on my PDA... */
-	}
-#else
-	if ((diff > delta) || (diff < -(delta * 5))) {
+	if (d->last_sent_time==-1){
 		d->tsoff = curts - packet_ts;
-		netts = packet_ts + d->tsoff;
-		ms_message("synchronizing timestamp, diff=%i", diff);
+	}else{
+		diffts=packet_ts-d->last_ts;
+		difftime_ts=((f->ticker->time-d->last_sent_time)*d->rate)/1000;
+		/* detect timestamp jump in the stream and adjust so that they become continuous on the network*/
+		if (abs(diffts-difftime_ts)>(d->rate/5)){
+			uint32_t tsoff=curts - packet_ts;
+			ms_message("Adjusting output timestamp by %i",(tsoff-d->tsoff));
+			d->tsoff = tsoff;
+		}
 	}
-#endif
 
-	/*ms_message("returned ts=%u, orig_ts=%u",netts,packet_ts); */
+	netts = packet_ts + d->tsoff;
+	d->last_sent_time=f->ticker->time;
+	d->last_ts=packet_ts;
 	return netts;
 }
 
@@ -417,9 +428,7 @@ static int receiver_set_session(MSFilter * f, void *arg)
 											  rtp_session_get_recv_payload_type
 											  (s));
 	if (pt != NULL) {
-		if (strcasecmp("g722", pt->mime_type)==0 )
-			d->rate=8000;
-		else d->rate = pt->clock_rate;
+		d->rate = pt->clock_rate;
 	} else {
 		ms_warning("Receiving undefined payload type %i ?",
 		    rtp_session_get_recv_payload_type(s));
@@ -439,7 +448,10 @@ static int receiver_get_sr(MSFilter *f, void *arg){
 	pt=rtp_profile_get_payload(rtp_session_get_profile(d->session),
 									rtp_session_get_recv_payload_type(d->session));
 	if (pt != NULL) {
-		*(int*)arg=pt->clock_rate;
+		if (strcasecmp(pt->mime_type,"G722")==0)
+			*(int*)arg=16000;
+		else
+			*(int*)arg=pt->clock_rate;
 	}else{
 		ms_warning("Could not obtain sample rate, payload type is unknown.");
 		return -1;

src/mtu.c

@@ -209,7 +209,7 @@ int ms_discover_mtu(const char *host){
 	mtu=1500-28;//was the size of the inital buf
 	do{
 		int send_returned;
-		char *buf=ms_malloc(mtu);//avoid mtu greater than the beginning
+		char *buf=ms_malloc0(mtu);//avoid mtu greater than the beginning
 		if(buf == NULL)
 		{
 			ms_error("malloc(): %s",strerror(errno));
@@ -220,14 +220,6 @@ int ms_discover_mtu(const char *host){
 		}
 		send_returned = send(sock,buf,mtu,0);
 		ms_free(buf);
-		if( send_returned < 0)//mtu actually change...
-		{
-			ms_error("send(): %s",strerror(errno));
-			err = close(sock);
-			if (err!=0)
-				ms_error("close(): %s", strerror(errno));
-			return -1;
-		}