Mercurial > pidgin
view src/mediastreamer/msrtpsend.c @ 12125:3c1bac709234
[gaim-migrate @ 14425]
Change /core/savedstatus/current and /core/savedstatus/idleaway
to ints (they used to be strings, where the value was the title of a
GaimSavedStatus).
The value is now equal to the "creation" timestamp of a saved_status.
The creation timestamp is used as the unique key. The primary reason
for this is to allow for saved statuses to have NULL titles. NULL titles
are needed for transient statuses. I also added a "last_used" timestamp.
This all paves the way for keeping track of recently used statuses
committer: Tailor Script <tailor@pidgin.im>
author | Mark Doliner <mark@kingant.net> |
---|---|
date | Fri, 18 Nov 2005 07:23:29 +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 "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; }