Mercurial > pidgin
view src/mediastreamer/msrtprecv.c @ 12288:3897229ccb33
[gaim-migrate @ 14592]
" This patch fixes two serious bugs in the hidden
conversation/queuing code.
First, it removes the need to explicitly present
conversations created under queuing conditions.
Instead, when an incoming message is received and a
conversation does not exist, a hidden conversation is
created in a received-im-msg signal handler by swapping
out the create_conversation function in the gtkconv
ui_ops. This fixes a bug which could allow
conversations to be created in a hidden state when they
should be visible (i.e. buddy pounce open an im window
action). This required a second search for a
conversation in server.c after the signal is emitted.
Second, it fixes a bug which would cause gaim to crash
when quitting with a queued message. Fixing this
simplified the code a bit by removing the
private_remove_gtkconv function and instead adding a
check in gaim_gtk_conv_window_remove_gtkconv to prevent
the hidden_convwin from being destroyed when the last
conversation is removed." --charkins
committer: Tailor Script <tailor@pidgin.im>
author | Luke Schierer <lschiere@pidgin.im> |
---|---|
date | Fri, 02 Dec 2005 00:40:32 +0000 |
parents | e67993da8a22 |
children |
line wrap: on
line source
/* 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 "msrtprecv.h" /* some utilities to convert mblk_t to MSMessage and vice-versa */ MSMessage *msgb_2_ms_message(mblk_t* mp){ MSMessage *msg; MSBuffer *msbuf; if (mp->b_datap->ref_count!=1) return NULL; /* cannot handle properly non-unique buffers*/ /* create a MSBuffer using the mblk_t buffer */ msg=ms_message_alloc(); msbuf=ms_buffer_alloc(0); msbuf->buffer=mp->b_datap->db_base; msbuf->size=(char*)mp->b_datap->db_lim-(char*)mp->b_datap->db_base; ms_message_set_buf(msg,msbuf); msg->size=mp->b_wptr-mp->b_rptr; msg->data=mp->b_rptr; /* free the mblk_t */ g_free(mp->b_datap); g_free(mp); return msg; } static MSRtpRecvClass *ms_rtp_recv_class=NULL; MSFilter * ms_rtp_recv_new(void) { MSRtpRecv *r; r=g_new(MSRtpRecv,1); ms_rtp_recv_init(r); if (ms_rtp_recv_class==NULL) { ms_rtp_recv_class=g_new0(MSRtpRecvClass,1); ms_rtp_recv_class_init(ms_rtp_recv_class); } MS_FILTER(r)->klass=MS_FILTER_CLASS(ms_rtp_recv_class); return(MS_FILTER(r)); } /* FOR INTERNAL USE*/ void ms_rtp_recv_init(MSRtpRecv *r) { ms_filter_init(MS_FILTER(r)); MS_FILTER(r)->outfifos=r->f_outputs; MS_FILTER(r)->outqueues=r->q_outputs; memset(r->f_outputs,0,sizeof(MSFifo*)*MSRTPRECV_MAX_OUTPUTS); memset(r->q_outputs,0,sizeof(MSFifo*)*MSRTPRECV_MAX_OUTPUTS); r->rtpsession=NULL; r->stream_started=0; } void ms_rtp_recv_class_init(MSRtpRecvClass *klass) { ms_filter_class_init(MS_FILTER_CLASS(klass)); ms_filter_class_set_name(MS_FILTER_CLASS(klass),"RTPRecv"); MS_FILTER_CLASS(klass)->max_qoutputs=MSRTPRECV_MAX_OUTPUTS; MS_FILTER_CLASS(klass)->max_foutputs=MSRTPRECV_MAX_OUTPUTS; MS_FILTER_CLASS(klass)->w_maxgran=MSRTPRECV_DEF_GRAN; ms_filter_class_set_attr(MS_FILTER_CLASS(klass),FILTER_IS_SOURCE); MS_FILTER_CLASS(klass)->destroy=(MSFilterDestroyFunc)ms_rtp_recv_destroy; MS_FILTER_CLASS(klass)->process=(MSFilterProcessFunc)ms_rtp_recv_process; MS_FILTER_CLASS(klass)->setup=(MSFilterSetupFunc)ms_rtp_recv_setup; } void ms_rtp_recv_process(MSRtpRecv *r) { MSFifo *fo; MSQueue *qo; MSSync *sync= r->sync; void *d; mblk_t *mp; gint len; gint gran=ms_sync_get_samples_per_tick(MS_SYNC(sync)); if (r->rtpsession==NULL) return; /* process output fifo and output queue*/ fo=r->f_outputs[0]; if (fo!=NULL) { while( (mp=rtp_session_recvm_with_ts(r->rtpsession,r->prev_ts))!=NULL) { /* try to get rtp packets and paste them to the output fifo */ r->stream_started=1; len=mp->b_cont->b_wptr-mp->b_cont->b_rptr; ms_fifo_get_write_ptr(fo,len,&d); if (d!=NULL){ memcpy(d,mp->b_cont->b_rptr,len); }else ms_warning("ms_rtp_recv_process: no space on output fifo !"); freemsg(mp); } r->prev_ts+=gran; } qo=r->q_outputs[0]; if (qo!=NULL) { guint32 clock; gint got=0; /* we are connected with queues (surely for video)*/ /* use the sync system time to compute a timestamp */ PayloadType *pt=rtp_profile_get_payload(r->rtpsession->profile,r->rtpsession->payload_type); if (pt==NULL) { ms_warning("ms_rtp_recv_process(): NULL RtpPayload- skipping."); return; } clock=(guint32)(((double)sync->time*(double)pt->clock_rate)/1000.0); /*g_message("Querying packet with timestamp %u",clock);*/ /* get rtp packet, and send them through the output queue */ while ( (mp=rtp_session_recvm_with_ts(r->rtpsession,clock))!=NULL ){ MSMessage *msg; mblk_t *mdata; /*g_message("Got packet with timestamp %u",clock);*/ got++; r->stream_started=1; mdata=mp->b_cont; freeb(mp); msg=msgb_2_ms_message(mdata); ms_queue_put(qo,msg); } } } void ms_rtp_recv_destroy( MSRtpRecv *obj) { g_free(obj); } RtpSession * ms_rtp_recv_set_session(MSRtpRecv *obj,RtpSession *session) { RtpSession *old=obj->rtpsession; obj->rtpsession=session; obj->prev_ts=0; return old; } void ms_rtp_recv_setup(MSRtpRecv *r,MSSync *sync) { r->sync=sync; r->stream_started=0; }