]> sjero.net Git - linphone/commitdiff
Linphone DCCP Support: Updates as a result of mediastreamer2 API changes
authorSamuel Jero <sj323707@ohio.edu>
Sat, 15 Jun 2013 22:36:26 +0000 (18:36 -0400)
committerSamuel Jero <sj323707@ohio.edu>
Sat, 15 Jun 2013 22:36:26 +0000 (18:36 -0400)
console/sipomatic.c
coreapi/conference.c
coreapi/linphonecall.c

index e7a1c6a88830640b0d3315a130a05cbc0d5f6a53..5e436becdcd00157f6324d027c0340fc598372ae 100644 (file)
@@ -97,7 +97,7 @@ void call_accept(Call *call)
        osip_message_set_content_type(msg,"application/sdp");
        osip_message_set_body(msg,call->sdpc->answerstr,strlen(call->sdpc->answerstr));
        eXosip_call_send_answer(call->tid,200,msg);
-       call->audio_stream=audio_stream_new(call->audio.localport,call->audio.localport+1,call->root->ipv6);
+       call->audio_stream=audio_stream_new(call->audio.localport,call->audio.localport+1,call->root->ipv6, FALSE, 0);
        audio_stream_start_with_files(call->audio_stream, call->profile,
                                call->audio.remaddr,call->audio.remoteport,call->audio.remoteport+1,
                                 call->audio.pt,20,hellofile,record_file);
index 0d8ea39a0fdd0e6b44f76581961744a63d5e4ff7..9c75b57f1fdeddc989074a485cb3cf0cd56c1235 100644 (file)
@@ -126,7 +126,7 @@ static RtpProfile *make_dummy_profile(int samplerate){
 static void add_local_endpoint(LinphoneConference *conf,LinphoneCore *lc){
        /*create a dummy audiostream in order to extract the local part of it */
        /* network address and ports have no meaning and are not used here. */
-       AudioStream *st=audio_stream_new(65000,65001,FALSE);
+       AudioStream *st=audio_stream_new(65000,65001,FALSE,FALSE,0);
        MSSndCard *playcard=lc->sound_conf.lsd_card ? 
                        lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard;
        MSSndCard *captcard=lc->sound_conf.capt_sndcard;
index d5366e5c66a3da90a322392b1adaa029a39626b9..389a7a6b1b4eaf55a947d7c98f6a08aaf10a1d03 100644 (file)
@@ -1198,7 +1198,9 @@ void linphone_call_init_audio_stream(LinphoneCall *call){
        int dscp;
 
        if (call->audiostream != NULL) return;
-       call->audiostream=audiostream=audio_stream_new(call->audio_port,call->audio_port+1,linphone_core_ipv6_enabled(lc));
+       call->audiostream=audiostream=audio_stream_new(call->audio_port,call->audio_port+1,
+                                                                               linphone_core_ipv6_enabled(lc),linphone_core_get_dccp(lc),
+                                                                               linphone_core_get_ccid(lc));
        dscp=linphone_core_get_audio_dscp(lc);
        if (dscp!=-1)
                audio_stream_set_dscp(audiostream,dscp);
@@ -1260,8 +1262,9 @@ void linphone_call_init_video_stream(LinphoneCall *call){
        if ((lc->video_conf.display || lc->video_conf.capture) && call->params.has_video){
                int video_recv_buf_size=lp_config_get_int(lc->config,"video","recv_buf_size",0);
                int dscp=linphone_core_get_video_dscp(lc);
-               
-               call->videostream=video_stream_new(call->video_port,call->video_port+1,linphone_core_ipv6_enabled(lc));
+               call->videostream=video_stream_new(call->video_port,call->video_port+1,
+                                                                               linphone_core_ipv6_enabled(lc),linphone_core_get_dccp(lc),
+                                                                               linphone_core_get_ccid(lc));
                if (dscp!=-1)
                        video_stream_set_dscp(call->videostream,dscp);
                video_stream_enable_display_filter_auto_rotate(call->videostream, lp_config_get_int(lc->config,"video","display_filter_auto_rotate",0));