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author | André Fabian Silva Delgado <emulatorman@lavabit.com> | 2013-08-19 14:26:05 -0300 |
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committer | André Fabian Silva Delgado <emulatorman@lavabit.com> | 2013-08-19 14:26:05 -0300 |
commit | 5fa644227cb2c395249d843dc0a814c1b9982a50 (patch) | |
tree | 04a9a675f85cd6087a0ab0b011c6de969c346813 /libre/kdenetwork-kopete-libre/mediastreamer29.patch | |
parent | 0e3fa90965b0af932c0372eb51c4ccf273b3d989 (diff) | |
download | abslibre-5fa644227cb2c395249d843dc0a814c1b9982a50.tar.gz abslibre-5fa644227cb2c395249d843dc0a814c1b9982a50.tar.bz2 abslibre-5fa644227cb2c395249d843dc0a814c1b9982a50.zip |
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.patch | 22 |
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; |