Mercurial > pidgin
diff src/mediastreamer/msrtpsend.c @ 12024:e67993da8a22
[gaim-migrate @ 14317]
I strongly suspect CruiseControl is going to yell at me for this.
A voice chat API, GUI + mediastreamer. This is what I'm using for Google Talk.
This doesn't actually do anything at all. There's no code in the Jabber plugin yet
to use this API (although it Works For Me). All it will do is compile and link.
If you're lucky.
To build this, you should install oRTP from Linphone, Speex and iLBC (also
from linphone, I believe). To not build this, ./configure --disable-vv.
Most of the configure.ac and Makefile.am hackery was lifted right out of
Linphone with a few modifications. It seems to work if you have everything
installed or if you --disable-vv. I haven't really tested not having
everything installed and not --disabling-vv.
It's kinda funky to include all of mediastreamer in the source tree like this,
but linphone doesn't build it as a separate library. I'll probably wind up
writing them a patch to build it as a .so so we can link it dynamically instead.
This code certainly isn't finished. It'll adapt as I progress on the Google code,
but it's certainly of more use here in CVS than in my personal tree.
committer: Tailor Script <tailor@pidgin.im>
author | Sean Egan <seanegan@gmail.com> |
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date | Wed, 09 Nov 2005 08:07:20 +0000 |
parents | |
children |
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--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/src/mediastreamer/msrtpsend.c Wed Nov 09 08:07:20 2005 +0000 @@ -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; +}