Attachment #6247 for bug #29079

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Lines 22-30 IF (MEDIASTREAMER_FOUND) Link Here

(-)a/cmake/modules/FindMediastreamer.cmake (-3 / +8 lines)
22	IF (NOT MEDIASTREAMER_FIND_QUIETLY)	22	IF (NOT MEDIASTREAMER_FIND_QUIETLY)
23	MESSAGE(STATUS "Found Mediastreamer: ${MEDIASTREAMER_LIBRARIES} (version: ${MEDIASTREAMER_VERSION})")	23	MESSAGE(STATUS "Found Mediastreamer: ${MEDIASTREAMER_LIBRARIES} (version: ${MEDIASTREAMER_VERSION})")
24	IF (MEDIASTREAMER_VERSION VERSION_LESS 2.9)	24	IF (MEDIASTREAMER_VERSION VERSION_LESS 2.9)
25	SET(MEDIASTREAMER_OLD TRUE)	25	SET(MEDIASTREAMER_LESS_2_9 TRUE)
		26	SET(MEDIASTREAMER_LESS_2_11 TRUE)
		27	ELSEIF (MEDIASTREAMER_VERSION VERSION_LESS 2.11)
		28	SET(MEDIASTREAMER_LESS_2_9 FALSE)
		29	SET(MEDIASTREAMER_LESS_2_11 TRUE)
26	ELSE (MEDIASTREAMER_VERSION VERSION_LESS 2.9)	30	ELSE (MEDIASTREAMER_VERSION VERSION_LESS 2.9)
27	SET(MEDIASTREAMER_OLD FALSE)	31	SET(MEDIASTREAMER_LESS_2_9 FALSE)
		32	SET(MEDIASTREAMER_LESS_2_11 FALSE)
28	ENDIF (MEDIASTREAMER_VERSION VERSION_LESS 2.9)	33	ENDIF (MEDIASTREAMER_VERSION VERSION_LESS 2.9)
29	ENDIF (NOT MEDIASTREAMER_FIND_QUIETLY)	34	ENDIF (NOT MEDIASTREAMER_FIND_QUIETLY)
30	ELSE (MEDIASTREAMER_FOUND)	35	ELSE (MEDIASTREAMER_FOUND)
Lines 33-36 ELSE (MEDIASTREAMER_FOUND) Link Here
33	ENDIF (MEDIASTREAMER_FIND_REQUIRED)	38	ENDIF (MEDIASTREAMER_FIND_REQUIRED)
34	ENDIF (MEDIASTREAMER_FOUND)	39	ENDIF (MEDIASTREAMER_FOUND)
35		40
36	MARK_AS_ADVANCED(MEDIASTREAMER_INCLUDE_DIR MEDIASTREAMER_LIBRARIES MEDIASTREAMER_OLD)	41	MARK_AS_ADVANCED(MEDIASTREAMER_INCLUDE_DIR MEDIASTREAMER_LIBRARIES MEDIASTREAMER_LESS_2_9 MEDIASTREAMER_LESS_2_11)

Lines 28-36 if ( NOT WIN32 ) Link Here

(-)a/protocols/jabber/libjingle/CMakeLists.txt (-3 / +7 lines)
28	endif ( NOT APPLE )	28	endif ( NOT APPLE )
29	endif ( NOT WIN32 )	29	endif ( NOT WIN32 )
30		30
31	if ( MEDIASTREAMER_OLD )	31	if ( MEDIASTREAMER_LESS_2_9 )
32	add_definitions ( -DMEDIASTREAMER_OLD )	32	add_definitions ( -DMEDIASTREAMER_LESS_2_9 )
33	endif ( MEDIASTREAMER_OLD )	33	endif ( MEDIASTREAMER_LESS_2_9 )
		34
		35	if ( MEDIASTREAMER_LESS_2_11 )
		36	add_definitions ( -DMEDIASTREAMER_LESS_2_11 )
		37	endif ( MEDIASTREAMER_LESS_2_11 )
34		38
35	if ( CMAKE_BUILD_TYPE STREQUAL "Debug" )	39	if ( CMAKE_BUILD_TYPE STREQUAL "Debug" )
36	add_definitions ( -D_DEBUG )	40	add_definitions ( -D_DEBUG )




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;




  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;

      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_get_rtp_session(audio_stream_));
      port2 = rtp_session_get_local_port(audio_stream_->session);
#else
      port2 = rtp_session_get_local_port(audio_stream_->ms.session);
#endif
      first = false;
    }
  }

    // 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_get_rtp_session(audio_stream_));
    port2 = rtp_session_get_local_port(audio_stream_->session);
#else
    port2 = rtp_session_get_local_port(audio_stream_->ms.session);
#endif
  }

  return true;

Return to bug 29079

Lines 171-176 bool LinphoneVoiceChannel::SetPlayout(bool playout) { Link Here

(-)a/protocols/jabber/libjingle/talk/session/phone/linphonemediaengine.cc (-10 / +12 lines)
171	return true;	171	return true;
172	}	172	}
173		173
		174	#ifdef MEDIASTREAMER_LESS_2_11
		175	static inline RtpSession * audio_stream_get_rtp_session(const AudioStream *stream) {
		176	#ifdef MEDIASTREAMER_LESS_2_9
		177	return stream->session;
		178	#else
		179	return stream->ms.session;
		180	#endif
		181	}
		182	#endif
		183
174	bool LinphoneVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs) {	184	bool LinphoneVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs) {
175		185
176	bool first = true;	186	bool first = true;
Lines 200-210 bool LinphoneVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs) Link Here
200	LOG(LS_INFO) << "Using " << i->name << "/" << i->clockrate;	210	LOG(LS_INFO) << "Using " << i->name << "/" << i->clockrate;
201	pt_ = i->id;	211	pt_ = i->id;
202	audio_stream_ = audio_stream_start(&av_profile, -1, "localhost", port1, i->id, 250, 0); /* -1 means that function will choose some free port */	212	audio_stream_ = audio_stream_start(&av_profile, -1, "localhost", port1, i->id, 250, 0); /* -1 means that function will choose some free port */
203	#ifdef MEDIASTREAMER_OLD	213	port2 = rtp_session_get_local_port(audio_stream_get_rtp_session(audio_stream_));
204	port2 = rtp_session_get_local_port(audio_stream_->session);
205	#else
206	port2 = rtp_session_get_local_port(audio_stream_->ms.session);
207	#endif
208	first = false;	214	first = false;
209	}	215	}
210	}	216	}
Lines 215-225 bool LinphoneVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs) Link Here
215	// working with a buggy client; let's try PCMU.	221	// working with a buggy client; let's try PCMU.
216	LOG(LS_WARNING) << "Received empty list of codces; using PCMU/8000";	222	LOG(LS_WARNING) << "Received empty list of codces; using PCMU/8000";
217	audio_stream_ = audio_stream_start(&av_profile, -1, "localhost", port1, 0, 250, 0); /* -1 means that function will choose some free port */	223	audio_stream_ = audio_stream_start(&av_profile, -1, "localhost", port1, 0, 250, 0); /* -1 means that function will choose some free port */
218	#ifdef MEDIASTREAMER_OLD	224	port2 = rtp_session_get_local_port(audio_stream_get_rtp_session(audio_stream_));
219	port2 = rtp_session_get_local_port(audio_stream_->session);
220	#else
221	port2 = rtp_session_get_local_port(audio_stream_->ms.session);
222	#endif
223	}	225	}
224		226
225	return true;	227	return true;

Line 0 Link Here

(-)a/protocols/jabber/libjingle/patches/08_mediastreamer_2_11.patch (+47 lines)
	1	diff --git a/protocols/jabber/libjingle/talk/session/phone/linphonemediaengine.cc b/protocols/jabber/libjingle/talk/session/phone/linphonemediaengine.cc
	2	index 6da35e0..e337dd4 100644
	3	--- a/protocols/jabber/libjingle/talk/session/phone/linphonemediaengine.cc
	4	+++ b/protocols/jabber/libjingle/talk/session/phone/linphonemediaengine.cc
	5	@@ -171,6 +171,16 @@ bool LinphoneVoiceChannel::SetPlayout(bool playout) {
	6	return true;
	7	}
	8
	9	+#ifdef MEDIASTREAMER_LESS_2_11
	10	+static inline RtpSession * audio_stream_get_rtp_session(const AudioStream *stream) {
	11	+#ifdef MEDIASTREAMER_LESS_2_9
	12	+ return stream->session;
	13	+#else
	14	+ return stream->ms.session;
	15	+#endif
	16	+}
	17	+#endif
	18	+
	19	bool LinphoneVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs) {
	20
	21	bool first = true;
	22	@@ -200,11 +210,7 @@ bool LinphoneVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs)
	23	LOG(LS_INFO) << "Using " << i->name << "/" << i->clockrate;
	24	pt_ = i->id;
	25	audio_stream_ = audio_stream_start(&av_profile, -1, "localhost", port1, i->id, 250, 0); /* -1 means that function will choose some free port */
	26	-#ifdef MEDIASTREAMER_OLD
	27	- port2 = rtp_session_get_local_port(audio_stream_->session);
	28	-#else
	29	- port2 = rtp_session_get_local_port(audio_stream_->ms.session);
	30	-#endif
	31	+ port2 = rtp_session_get_local_port(audio_stream_get_rtp_session(audio_stream_));
	32	first = false;
	33	}
	34	}
	35	@@ -215,11 +221,7 @@ bool LinphoneVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs)
	36	// working with a buggy client; let's try PCMU.
	37	LOG(LS_WARNING) << "Received empty list of codces; using PCMU/8000";
	38	audio_stream_ = audio_stream_start(&av_profile, -1, "localhost", port1, 0, 250, 0); /* -1 means that function will choose some free port */
	39	-#ifdef MEDIASTREAMER_OLD
	40	- port2 = rtp_session_get_local_port(audio_stream_->session);
	41	-#else
	42	- port2 = rtp_session_get_local_port(audio_stream_->ms.session);
	43	-#endif
	44	+ port2 = rtp_session_get_local_port(audio_stream_get_rtp_session(audio_stream_));
	45	}
	46
	47	return true;