[SCM] kopete packaging branch, master, updated. debian/4.13.1-1-61-gdb6961a

[Maximiliano Curia]

Gitweb-URL: http://git.debian.org/?p=pkg-kde/kde-sc/kopete.git;a=commitdiff;h=612c159

The following commit has been merged in the master branch:
commit 612c15948e541410891e6cfa111b9d279dbb353f
Author: Howard Chan <smartboyhw at gmail.com>
Date:   Thu Aug 8 23:35:10 2013 +0800

    Introduce patch at http://bugsfiles.kde.org/attachment.cgi?id=80534
    to fix audio_stream
---
 debian/changelog                        |  2 ++
 debian/patches/04_fix_audio_stream.diff | 23 +++++++++++++++++++++++
 debian/patches/series                   |  1 +
 3 files changed, 26 insertions(+)

diff --git a/debian/changelog b/debian/changelog
index ec36d1a..a1cd600 100644
--- a/debian/changelog
+++ b/debian/changelog
@@ -2,6 +2,8 @@ kopete (4:4.11.0-0ubuntu1) UNRELEASED; urgency=low
 
   * New upstream release
   * Add libsrtp0-dev and libasound2-dev as build-dep of kopete
+  * Introduce patch at http://bugsfiles.kde.org/attachment.cgi?id=80534
+    to fix audio_stream
 
  -- Howard Chan <smartboyhw at gmail.com>  Thu, 08 Aug 2013 12:25:57 +0800
 
diff --git a/debian/patches/04_fix_audio_stream.diff b/debian/patches/04_fix_audio_stream.diff
new file mode 100644
index 0000000..bbbb717
--- /dev/null
+++ b/debian/patches/04_fix_audio_stream.diff
@@ -0,0 +1,23 @@
+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;
+
diff --git a/debian/patches/series b/debian/patches/series
index b2ed00c..98136c5 100644
--- a/debian/patches/series
+++ b/debian/patches/series
@@ -1 +1,2 @@
 kubuntu_03_no_mediastreamer_in_wlm.diff
+04_fix_audio_stream.diff

-- 
kopete packaging