[asterisk-scf-commits] asterisk-scf/integration/media_rtp_pjmedia.git branch "master" updated.

[Commits to the Asterisk SCF project code repositories]

branch "master" has been updated
       via  3b87167160824ca5f5f68c0396bfc6aa95d638c1 (commit)
      from  2d4dffbc88cfeaa15848db1c7b4adcf26787fba0 (commit)

Summary of changes:
 src/RTPSink.cpp         |    6 +++
 test/TestRTPpjmedia.cpp |   98 +++++++++++++++++++++++-----------------------
 2 files changed, 55 insertions(+), 49 deletions(-)


- Log -----------------------------------------------------------------
commit 3b87167160824ca5f5f68c0396bfc6aa95d638c1
Author: Joshua Colp <jcolp at digium.com>
Date:   Wed Sep 15 13:15:37 2010 -0300

    Move test case for no remote address to correct position, and also don't write frames out if no remote address has been set.

diff --git a/src/RTPSink.cpp b/src/RTPSink.cpp
index 9d6745c..e51b776 100644
--- a/src/RTPSink.cpp
+++ b/src/RTPSink.cpp
@@ -71,6 +71,12 @@ StreamSinkRTPImpl::StreamSinkRTPImpl(RTPSessionImplPtr session) : mImpl(new Stre
  */
 void StreamSinkRTPImpl::write(const AsteriskSCF::Media::V1::FrameSeq& frames, const Ice::Current&)
 {
+	// Don't even bother if no remote address information is present
+	if (mImpl->mRemoteAddress.empty() || !mImpl->mRemotePort)
+	{
+	   return;
+	}
+
 	for (FrameSeq::const_iterator frame = frames.begin(); frame != frames.end(); ++frame)
 	{
 		AudioFormatPtr audioformat;
diff --git a/test/TestRTPpjmedia.cpp b/test/TestRTPpjmedia.cpp
index dc4229a..19d4a79 100644
--- a/test/TestRTPpjmedia.cpp
+++ b/test/TestRTPpjmedia.cpp
@@ -382,6 +382,55 @@ BOOST_AUTO_TEST_CASE(ConfirmBlankRemoteAddressonSinks)
 }
 
 /**
+ * Attempt to send a frame to an RTP session sink that has no destination yet
+ */
+BOOST_AUTO_TEST_CASE(TransmitFrametoEmptySink)
+{
+   bool graceful = false;
+
+   try
+   {
+      AudioFormatPtr format = new AudioFormat();
+      format->name = "test_format";
+      format->frameSize = 20;
+
+      AudioFramePtr frame = new AudioFrame();
+      frame->mediaformat = format;
+
+      /* Populate the payload with some useless data, but enough to confirm the payload passes unaltered. */
+      frame->payload.push_back('a');
+      frame->payload.push_back('b');
+      frame->payload.push_back('c');
+      frame->payload.push_back('d');
+      frame->payload.push_back('e');
+      frame->payload.push_back('f');
+      frame->payload.push_back('g');
+      frame->payload.push_back('h');
+      frame->payload.push_back('i');
+      frame->payload.push_back('j');
+
+      FrameSeq frames;
+      frames.push_back(frame);
+
+      StreamSinkSeq sinks = Testbed.session->getSinks();
+      StreamSinkRTPPrx sink = StreamSinkRTPPrx::uncheckedCast(sinks.front());
+      sink->write(frames);
+
+      graceful = true;
+   }
+   catch (const Ice::Exception &e)
+   {
+      BOOST_TEST_MESSAGE(e.ice_name());
+      BOOST_TEST_MESSAGE(e.what());
+   }
+   catch (...)
+   {
+   }
+
+   BOOST_CHECK(graceful);
+}
+
+/**
  * Check that we can set the remote address information properly
  */
 BOOST_AUTO_TEST_CASE(ConfirmRemoteAddressSetting)
@@ -490,55 +539,6 @@ BOOST_AUTO_TEST_CASE(PushPayloadMappings)
 }
 
 /**
- * Attempt to send a frame to an RTP session sink that has no destination yet
- */
-BOOST_AUTO_TEST_CASE(TransmitFrametoEmptySink)
-{
-	bool graceful = false;
-
-	try
-	{
-		AudioFormatPtr format = new AudioFormat();
-		format->name = "test_format";
-		format->frameSize = 20;
-
-		AudioFramePtr frame = new AudioFrame();
-		frame->mediaformat = format;
-
-		/* Populate the payload with some useless data, but enough to confirm the payload passes unaltered. */
-		frame->payload.push_back('a');
-		frame->payload.push_back('b');
-		frame->payload.push_back('c');
-		frame->payload.push_back('d');
-		frame->payload.push_back('e');
-		frame->payload.push_back('f');
-		frame->payload.push_back('g');
-		frame->payload.push_back('h');
-		frame->payload.push_back('i');
-		frame->payload.push_back('j');
-
-		FrameSeq frames;
-		frames.push_back(frame);
-
-		StreamSinkSeq sinks = Testbed.session->getSinks();
-		StreamSinkRTPPrx sink = StreamSinkRTPPrx::uncheckedCast(sinks.front());
-		sink->write(frames);
-
-		graceful = true;
-	}
-	catch (const Ice::Exception &e)
-	{
-		BOOST_TEST_MESSAGE(e.ice_name());
-		BOOST_TEST_MESSAGE(e.what());
-	}
-	catch (...)
-	{
-	}
-
-	BOOST_CHECK(graceful);
-}
-
-/**
  * Setup RTP loopback from the session to, well, the session
  */
 BOOST_AUTO_TEST_CASE(SetupLoopback)

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asterisk-scf/integration/media_rtp_pjmedia.git