Mercurial > pidgin
view src/mediastreamer/msrtpsend.c @ 12540:b0ba132271a9
[gaim-migrate @ 14853]
this doesn't make it work, but it does make it more correct
committer: Tailor Script <tailor@pidgin.im>
author | Nathan Walp <nwalp@pidgin.im> |
---|---|
date | Sat, 17 Dec 2005 21:01:40 +0000 |
parents | e67993da8a22 |
children |
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/* 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; }