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authorAndré Fabian Silva Delgado <emulatorman@lavabit.com>2013-08-19 14:26:05 -0300
committerAndré Fabian Silva Delgado <emulatorman@lavabit.com>2013-08-19 14:26:05 -0300
commit5fa644227cb2c395249d843dc0a814c1b9982a50 (patch)
tree04a9a675f85cd6087a0ab0b011c6de969c346813 /libre/kdenetwork-kopete-libre/mediastreamer29.patch
parent0e3fa90965b0af932c0372eb51c4ccf273b3d989 (diff)
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kde packages: updating libre packages to 4.11.0 version
Diffstat (limited to 'libre/kdenetwork-kopete-libre/mediastreamer29.patch')
-rw-r--r--libre/kdenetwork-kopete-libre/mediastreamer29.patch22
1 files changed, 22 insertions, 0 deletions
diff --git a/libre/kdenetwork-kopete-libre/mediastreamer29.patch b/libre/kdenetwork-kopete-libre/mediastreamer29.patch
new file mode 100644
index 000000000..eb0b2297e
--- /dev/null
+++ b/libre/kdenetwork-kopete-libre/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;