forked from mirrors/nixpkgs
d8f8d052c5
kopete-4.10.4-kopete-linphonemediaengine.patch ---------------------------------------------- patch copied from here: https://bugs.kde.org/show_bug.cgi?id=318825 kopete-4.10.4-kopete-stun.patch ------------------------------- when compiling kopete/protocols/jabber/googletalk/libjingle/talk/session/phone/channelmanager.cc it would produce this error: kopete/protocols/jabber/googletalk/libjingle/talk/p2p/base/stunrequest.h:91:9: error: ‘StunMessageType’ does not name a type problem: this is cased by a cyclic use of stun.h, stunrequest.h and channelmanager.cc with the outcome, that kdenetwork couldn't be compiled since kopete fails to build. solution: move the StunMessageType enum into its own #ifndef
23 lines
1.4 KiB
Diff
23 lines
1.4 KiB
Diff
diff --git a/kopete/protocols/jabber/googletalk/libjingle/talk/session/phone/linphonemediaengine.cc b/kopete/protocols/jabber/googletalk/libjingle/talk/session/phone/linphonemediaengine.cc
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index 88fdbd1..57c6c05 100644
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--- a/kopete/protocols/jabber/googletalk/libjingle/talk/session/phone/linphonemediaengine.cc
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+++ b/kopete/protocols/jabber/googletalk/libjingle/talk/session/phone/linphonemediaengine.cc
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@@ -200,7 +200,7 @@ bool LinphoneVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs)
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LOG(LS_INFO) << "Using " << i->name << "/" << i->clockrate;
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pt_ = i->id;
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audio_stream_ = audio_stream_start(&av_profile, -1, "localhost", port1, i->id, 250, 0); /* -1 means that function will choose some free port */
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- port2 = rtp_session_get_local_port(audio_stream_->session);
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+ port2 = rtp_session_get_local_port(audio_stream_->ms.session);
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first = false;
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}
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}
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@@ -211,7 +211,7 @@ bool LinphoneVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs)
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// working with a buggy client; let's try PCMU.
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LOG(LS_WARNING) << "Received empty list of codces; using PCMU/8000";
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audio_stream_ = audio_stream_start(&av_profile, -1, "localhost", port1, 0, 250, 0); /* -1 means that function will choose some free port */
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- port2 = rtp_session_get_local_port(audio_stream_->session);
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+ port2 = rtp_session_get_local_port(audio_stream_->ms.session);
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}
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return true;
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