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author | André Fabian Silva Delgado <emulatorman@parabola.nu> | 2013-12-23 02:13:57 -0200 |
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committer | André Fabian Silva Delgado <emulatorman@parabola.nu> | 2013-12-23 02:17:47 -0200 |
commit | a70d3b5735ef66f045d45d26da8adc05b1841133 (patch) | |
tree | 79a3849baded6c27cb1f01cd7bf25775a7892c79 /libre/kdenetwork-kopete-libre/mediastreamer29.patch | |
parent | 9e17f935d4121eb2915297b1ea91d74b56390914 (diff) | |
download | abslibre-a70d3b5735ef66f045d45d26da8adc05b1841133.tar.gz abslibre-a70d3b5735ef66f045d45d26da8adc05b1841133.tar.bz2 abslibre-a70d3b5735ef66f045d45d26da8adc05b1841133.zip |
kde packages: updating libre and nonprism packages to 4.12.0 version
* add description for kdelibs-libre package
Diffstat (limited to 'libre/kdenetwork-kopete-libre/mediastreamer29.patch')
-rw-r--r-- | libre/kdenetwork-kopete-libre/mediastreamer29.patch | 22 |
1 files changed, 0 insertions, 22 deletions
diff --git a/libre/kdenetwork-kopete-libre/mediastreamer29.patch b/libre/kdenetwork-kopete-libre/mediastreamer29.patch deleted file mode 100644 index eb0b2297e..000000000 --- a/libre/kdenetwork-kopete-libre/mediastreamer29.patch +++ /dev/null @@ -1,22 +0,0 @@ -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; |