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authoraurelien <aurelien@cwb.io>2013-09-23 10:33:22 +0200
committeraurelien <aurelien@cwb.io>2013-09-23 10:33:22 +0200
commit2735c9d273c54b555681f30f5c685998fed7b484 (patch)
treef8172dbea44d1c4751a99a62fb236a14c9d2dc20 /nonprism/kdenetwork-kopete-libre-nonprism/mediastreamer29.patch
parentd78a32a6ccb8663a62a093b54e947e2efc840fe0 (diff)
parent97f65e06566f3bb15bc7395624017b8e6ca0d043 (diff)
Merge branch 'master' of ssh://parabolagnulinux.org:1863/srv/git/abslibre
Diffstat (limited to 'nonprism/kdenetwork-kopete-libre-nonprism/mediastreamer29.patch')
-rw-r--r--nonprism/kdenetwork-kopete-libre-nonprism/mediastreamer29.patch22
1 files changed, 22 insertions, 0 deletions
diff --git a/nonprism/kdenetwork-kopete-libre-nonprism/mediastreamer29.patch b/nonprism/kdenetwork-kopete-libre-nonprism/mediastreamer29.patch
new file mode 100644
index 000000000..eb0b2297e
--- /dev/null
+++ b/nonprism/kdenetwork-kopete-libre-nonprism/mediastreamer29.patch
@@ -0,0 +1,22 @@
+diff --git a/protocols/jabber/googletalk/libjingle/talk/session/phone/linphonemediaengine.cc b/protocols/jabber/googletalk/libjingle/talk/session/phone/linphonemediaengine.cc
+index 88fdbd1..57c6c05 100644
+--- a/protocols/jabber/googletalk/libjingle/talk/session/phone/linphonemediaengine.cc
++++ b/protocols/jabber/googletalk/libjingle/talk/session/phone/linphonemediaengine.cc
+@@ -200,7 +200,7 @@ bool LinphoneVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs)
+ LOG(LS_INFO) << "Using " << i->name << "/" << i->clockrate;
+ pt_ = i->id;
+ audio_stream_ = audio_stream_start(&av_profile, -1, "localhost", port1, i->id, 250, 0); /* -1 means that function will choose some free port */
+- port2 = rtp_session_get_local_port(audio_stream_->session);
++ port2 = rtp_session_get_local_port(audio_stream_->ms.session);
+ first = false;
+ }
+ }
+@@ -211,7 +211,7 @@ bool LinphoneVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs)
+ // working with a buggy client; let's try PCMU.
+ LOG(LS_WARNING) << "Received empty list of codces; using PCMU/8000";
+ audio_stream_ = audio_stream_start(&av_profile, -1, "localhost", port1, 0, 250, 0); /* -1 means that function will choose some free port */
+- port2 = rtp_session_get_local_port(audio_stream_->session);
++ port2 = rtp_session_get_local_port(audio_stream_->ms.session);
+ }
+
+ return true;