view src/mediastreamer/msrtpsend.c @ 12115:e9790eb93216

[gaim-migrate @ 14415] quoth charkins: " This patch has a few small fixes for the visibility stuff in gtkblist.c. First, tracking of the ICONIFIED state of the blist was removed. This was intended to allow the blist to "remember" if it was minimized between restarts. Unfortunately, this is not possible because the ICONIFIED state gets set when the blist is on a different desktop with many window managers. Second, while talking about the ICONIFIED issue on #gtk@GIMPNet, muntyan_ asked about a bug where the blist would get shown on an account re-connect with 1.5.0. Luke mentioned something about this with cvs as well. This patch introduces a check in gaim_gtk_blist_show() to prevent the window from being shown if it already exists and is visible. Third, sadrul pointed me to a one-line fix for the missing blist on startup. I added a second line to make sure the blist restores its proper size as well. Finally, when the last visibility manager is removed, gaim will now minimize the blist if it was previously hidden, rather than showing it. This could prevent a race condition with out-of-process applets, preventing gaim from maintaining the visibility state properly between restarts. This was 'cvs diff'ed against the last available anon cvs from Friday. Hopefully it'll apply cleanly." it did. committer: Tailor Script <[e-mail]>

/*
  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 [e-mail]
  										
  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;
}