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Commit 4dbdd0c3 authored by Pali Rohár's avatar Pali Rohár

libjingle: Regenerate linphonemediaengine patches

parent 3b8969d9
diff --git a/protocols/jabber/googletalk/libjingle/talk/session/phone/linphonemediaengine.cc b/protocols/jabber/googletalk/libjingle/talk/session/phone/linphonemediaengine.cc
index 88fdbd1..6da35e0 100644
--- a/protocols/jabber/googletalk/libjingle/talk/session/phone/linphonemediaengine.cc
+++ b/protocols/jabber/googletalk/libjingle/talk/session/phone/linphonemediaengine.cc
@@ -200,7 +200,11 @@ 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 */
+#ifdef MEDIASTREAMER_OLD
port2 = rtp_session_get_local_port(audio_stream_->session);
+#else
+ port2 = rtp_session_get_local_port(audio_stream_->ms.session);
+#endif
first = false;
}
}
@@ -211,7 +215,11 @@ 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 */
+#ifdef MEDIASTREAMER_OLD
port2 = rtp_session_get_local_port(audio_stream_->session);
+#else
+ port2 = rtp_session_get_local_port(audio_stream_->ms.session);
+#endif
}
return true;
diff --git a/protocols/jabber/libjingle/talk/session/phone/linphonemediaengine.cc b/protocols/jabber/libjingle/talk/session/phone/linphonemediaengine.cc
index 6da35e0..e337dd4 100644
--- a/protocols/jabber/libjingle/talk/session/phone/linphonemediaengine.cc
+++ b/protocols/jabber/libjingle/talk/session/phone/linphonemediaengine.cc
@@ -171,6 +171,16 @@ bool LinphoneVoiceChannel::SetPlayout(bool playout) {
return true;
}
+#ifdef MEDIASTREAMER_LESS_2_11
+static inline RtpSession * audio_stream_get_rtp_session(const AudioStream *stream) {
+#ifdef MEDIASTREAMER_LESS_2_9
+ return stream->session;
+#else
+ return stream->ms.session;
+#endif
+}
+#endif
+
bool LinphoneVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs) {
bool first = true;
@@ -200,11 +210,7 @@ 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 */
-#ifdef MEDIASTREAMER_OLD
- port2 = rtp_session_get_local_port(audio_stream_->session);
-#else
- port2 = rtp_session_get_local_port(audio_stream_->ms.session);
-#endif
+ port2 = rtp_session_get_local_port(audio_stream_get_rtp_session(audio_stream_));
first = false;
}
}
@@ -215,11 +221,7 @@ 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 */
-#ifdef MEDIASTREAMER_OLD
- port2 = rtp_session_get_local_port(audio_stream_->session);
-#else
- port2 = rtp_session_get_local_port(audio_stream_->ms.session);
-#endif
+ port2 = rtp_session_get_local_port(audio_stream_get_rtp_session(audio_stream_));
}
return true;
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