1 files changed, 211 insertions, 0 deletions

diff --git a/kopete/protocols/jabber/jingle/libjingle/talk/third_party/mediastreamer/msrtpsend.c b/kopete/protocols/jabber/jingle/libjingle/talk/third_party/mediastreamer/msrtpsend.c
new file mode 100644
index 00000000..cfcb6b34
--- /dev/null
+++ b/kopete/protocols/jabber/jingle/libjingle/talk/third_party/mediastreamer/msrtpsend.c
@@ -0,0 +1,211 @@
+/*
+  The mediastreamer library aims at providing modular media processing and I/O
+	for linphone, but also for any telephony application.
+  Copyright (C) 2001  Simon MORLAT simon.morlat@linphone.org
+  										
+  This library is free software; you can redistribute it and/or
+  modify it under the terms of the GNU Lesser General Public
+  License as published by the Free Software Foundation; either
+  version 2.1 of the License, or (at your option) any later version.
+
+  This library is distributed in the hope that it will be useful,
+  but WITHOUT ANY WARRANTY; without even the implied warranty of
+  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+  Lesser General Public License for more details.
+
+  You should have received a copy of the GNU Lesser General Public
+  License along with this library; if not, write to the Free Software
+  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+*/
+
+#include "msrtpsend.h"
+#include <ortp/telephonyevents.h>
+#include "mssync.h"
+#include "mscodec.h"
+
+
+
+static MSRtpSendClass *ms_rtp_send_class=NULL;
+
+MSFilter * ms_rtp_send_new(void)
+{
+	MSRtpSend *r;
+	
+	r=g_new(MSRtpSend,1);
+	
+	if (ms_rtp_send_class==NULL)
+	{
+		ms_rtp_send_class=g_new(MSRtpSendClass,1);
+		ms_rtp_send_class_init(ms_rtp_send_class);
+	}
+	MS_FILTER(r)->klass=MS_FILTER_CLASS(ms_rtp_send_class);
+	ms_rtp_send_init(r);
+	return(MS_FILTER(r));
+}
+	
+
+void ms_rtp_send_init(MSRtpSend *r)
+{
+	ms_filter_init(MS_FILTER(r));
+	MS_FILTER(r)->infifos=r->f_inputs;
+	MS_FILTER(r)->inqueues=r->q_inputs;
+	MS_FILTER(r)->r_mingran=MSRTPSEND_DEF_GRAN;
+	memset(r->f_inputs,0,sizeof(MSFifo*)*MSRTPSEND_MAX_INPUTS);
+	memset(r->q_inputs,0,sizeof(MSFifo*)*MSRTPSEND_MAX_INPUTS);
+	r->rtpsession=NULL;
+	r->ts=0;
+	r->ts_inc=0;
+	r->flags=0;
+	r->delay=0;
+}
+
+void ms_rtp_send_class_init(MSRtpSendClass *klass)
+{
+	ms_filter_class_init(MS_FILTER_CLASS(klass));
+	ms_filter_class_set_name(MS_FILTER_CLASS(klass),"RTPSend");
+	MS_FILTER_CLASS(klass)->max_qinputs=MSRTPSEND_MAX_INPUTS;
+	MS_FILTER_CLASS(klass)->max_finputs=MSRTPSEND_MAX_INPUTS;
+	MS_FILTER_CLASS(klass)->r_maxgran=MSRTPSEND_DEF_GRAN;
+	MS_FILTER_CLASS(klass)->destroy=(MSFilterDestroyFunc)ms_rtp_send_destroy;
+	MS_FILTER_CLASS(klass)->process=(MSFilterProcessFunc)ms_rtp_send_process;
+	MS_FILTER_CLASS(klass)->setup=(MSFilterSetupFunc)ms_rtp_send_setup;
+}
+
+void ms_rtp_send_set_timing(MSRtpSend *r, guint32 ts_inc, gint payload_size)
+{
+	r->ts_inc=ts_inc;
+	r->packet_size=payload_size;
+	if (r->ts_inc!=0) r->flags|=RTPSEND_CONFIGURED;
+	else r->flags&=~RTPSEND_CONFIGURED;
+	MS_FILTER(r)->r_mingran=payload_size;	
+	/*g_message("ms_rtp_send_set_timing: ts_inc=%i",ts_inc);*/
+}
+
+guint32 get_new_timestamp(MSRtpSend *r,guint32 synctime)
+{
+	guint32 clockts;
+	/* use the sync system time to compute a timestamp */
+	PayloadType *pt=rtp_profile_get_payload(r->rtpsession->profile,r->rtpsession->payload_type);
+	g_return_val_if_fail(pt!=NULL,0);
+	clockts=(guint32)(((double)synctime * (double)pt->clock_rate)/1000.0);
+	ms_trace("ms_rtp_send_process: sync->time=%i clock=%i",synctime,clockts);
+	if (r->flags & RTPSEND_CONFIGURED){
+		if (RTP_TIMESTAMP_IS_STRICTLY_NEWER_THAN(clockts,r->ts+(2*r->ts_inc) )){
+			r->ts=clockts;
+		}
+		else r->ts+=r->ts_inc;
+	}else{
+		r->ts=clockts;
+	}
+	return r->ts;
+}
+
+
+void ms_rtp_send_process(MSRtpSend *r)
+{
+	MSFifo *fi;
+	MSQueue *qi;
+	MSSync *sync= r->sync;
+	int gran=ms_sync_get_samples_per_tick(sync);
+	guint32 ts;
+	void *s;
+	guint skip;
+	guint32 synctime=sync->time;
+	
+	g_return_if_fail(gran>0);
+	if (r->rtpsession==NULL) return;
+
+	ms_filter_lock(MS_FILTER(r));
+	skip=r->delay!=0;
+	if (skip) r->delay--;
+	/* process output fifo and output queue*/
+	fi=r->f_inputs[0];
+	if (fi!=NULL)
+	{
+		ts=get_new_timestamp(r,synctime);
+		/* try to read r->packet_size bytes and send them in a rtp packet*/
+		ms_fifo_get_read_ptr(fi,r->packet_size,&s);
+		if (!skip){
+			rtp_session_send_with_ts(r->rtpsession,s,r->packet_size,ts);
+			ms_trace("len=%i, ts=%i ",r->packet_size,ts);
+		}
+	}
+	qi=r->q_inputs[0];
+	if (qi!=NULL)
+	{
+		MSMessage *msg;
+		/* read a MSMessage and send it through the network*/
+		while ( (msg=ms_queue_get(qi))!=NULL){
+			ts=get_new_timestamp(r,synctime);
+			if (!skip) {
+				/*g_message("Sending packet with ts=%u",ts);*/
+				rtp_session_send_with_ts(r->rtpsession,msg->data,msg->size,ts);
+				
+			}
+			ms_message_destroy(msg);
+		}
+	}
+	ms_filter_unlock(MS_FILTER(r));
+}
+
+void ms_rtp_send_destroy( MSRtpSend *obj)
+{
+	g_free(obj);
+}
+
+RtpSession * ms_rtp_send_set_session(MSRtpSend *obj,RtpSession *session)
+{
+	RtpSession *old=obj->rtpsession;
+	obj->rtpsession=session;
+	obj->ts=0;
+	obj->ts_inc=0;
+	return old;
+}
+
+void ms_rtp_send_setup(MSRtpSend *r, MSSync *sync)
+{
+	MSFilter *codec;
+	MSCodecInfo *info;
+	r->sync=sync;
+	codec=ms_filter_search_upstream_by_type(MS_FILTER(r),MS_FILTER_AUDIO_CODEC);
+	if (codec==NULL) codec=ms_filter_search_upstream_by_type(MS_FILTER(r),MS_FILTER_VIDEO_CODEC);
+	if (codec==NULL){
+		g_warning("ms_rtp_send_setup: could not find upstream codec.");
+		return;
+	}
+	info=MS_CODEC_INFO(codec->klass->info);
+	if (info->info.type==MS_FILTER_AUDIO_CODEC){
+		int ts_inc=info->fr_size/2;
+		int psize=info->dt_size;
+		if (ts_inc==0){
+			/* dont'use the normal frame size: this is a variable frame size codec */
+			/* use the MS_FILTER(codec)->r_mingran */
+			ts_inc=MS_FILTER(codec)->r_mingran/2;
+			psize=0;
+		}
+		ms_rtp_send_set_timing(r,ts_inc,psize);
+	}
+}
+
+gint ms_rtp_send_dtmf(MSRtpSend *r, gchar dtmf)
+{
+	gint res;
+
+	if (r->rtpsession==NULL) return -1;
+	if (rtp_session_telephone_events_supported(r->rtpsession)==-1){
+		g_warning("ERROR : telephone events not supported.\n");
+ 		return -1;
+	}
+
+	ms_filter_lock(MS_FILTER(r));
+	g_message("Sending DTMF.");
+	res=rtp_session_send_dtmf(r->rtpsession, dtmf, r->ts);
+	if (res==0){
+		/* //r->ts+=r->ts_inc; */
+		r->delay+=2;
+	}else g_warning("Could not send dtmf.");
+
+	ms_filter_unlock(MS_FILTER(r));
+
+	return res;
+}