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Bug 863306 - Turn off rtcp-mux in config

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This commit is contained in:
Ethan Hugg 2013-08-20 13:21:27 -07:00
parent 98330c01bc
commit c69d8f108a
3 changed files with 21 additions and 10 deletions
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4
CLOBBER
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@ -18,4 +18,6 @@
# Modifying this file will now automatically clobber the buildbot machines \o/ # Modifying this file will now automatically clobber the buildbot machines \o/
# #
touching resources in Android requires a clobber, and I did that in bug 880259 In Bug 863306 comment #42 sheriff KWierso states that we should CLOBBER every time
that WebRTC code is changed, which will be awesome.

2
media/webrtc/signaling/src/sipcc/core/includes/config.h
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@ -179,7 +179,7 @@ static const int gDscpCallControl = 1;
static const int gSpeakerEnabled = 1; static const int gSpeakerEnabled = 1;
static const char gExternalNumberMask[] = ""; static const char gExternalNumberMask[] = "";
static const char gVersion[] = "0.1"; static const char gVersion[] = "0.1";
static const boolean gRTCPMUX = TRUE; static const boolean gRTCPMUX = FALSE;
static boolean gRTPSAVPF = TRUE; /* TRUE = RTP/SAVPF , FALSE = RTP/SAVP */ static boolean gRTPSAVPF = TRUE; /* TRUE = RTP/SAVPF , FALSE = RTP/SAVP */
static const boolean gMAXAVBITRATE = FALSE; /* Following six are OPUS fmtp options */ static const boolean gMAXAVBITRATE = FALSE; /* Following six are OPUS fmtp options */
static const boolean gMAXCODEDAUDIOBW = FALSE; static const boolean gMAXCODEDAUDIOBW = FALSE;

25
media/webrtc/signaling/test/signaling_unittests.cpp
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@ -37,6 +37,7 @@ MtransportTestUtils *test_utils;
nsCOMPtr<nsIThread> gThread; nsCOMPtr<nsIThread> gThread;
static int kDefaultTimeout = 5000; static int kDefaultTimeout = 5000;
static bool fRtcpMux = false;
static std::string callerName = "caller"; static std::string callerName = "caller";
static std::string calleeName = "callee"; static std::string calleeName = "callee";
@ -1683,11 +1684,14 @@ TEST_F(SignalingTest, FullCall)
// Check the low-level media pipeline // Check the low-level media pipeline
// for RTP and RTCP flows // for RTP and RTCP flows
// The first Local pipeline gets stored at 0 // The first Local pipeline gets stored at 0
a1_.CheckMediaPipeline(0, 0, a1_.CheckMediaPipeline(0, 0, fRtcpMux ?
PIPELINE_LOCAL | PIPELINE_RTCP_MUX | PIPELINE_SEND); PIPELINE_LOCAL | PIPELINE_RTCP_MUX | PIPELINE_SEND :
PIPELINE_LOCAL | PIPELINE_SEND);
// The first Remote pipeline gets stored at 1 // The first Remote pipeline gets stored at 1
a2_.CheckMediaPipeline(0, 1, PIPELINE_RTCP_MUX); a2_.CheckMediaPipeline(0, 1, fRtcpMux ?
PIPELINE_RTCP_MUX :
0);
} }
TEST_F(SignalingTest, FullCallAudioOnly) TEST_F(SignalingTest, FullCallAudioOnly)
@ -2410,8 +2414,10 @@ TEST_F(SignalingTest, FullCallAudioNoMuxVideoMux)
// Answer should have only one a=rtcp-mux line // Answer should have only one a=rtcp-mux line
size_t match = a2_.getLocalDescription().find("\r\na=rtcp-mux"); size_t match = a2_.getLocalDescription().find("\r\na=rtcp-mux");
ASSERT_NE(match, std::string::npos); if (fRtcpMux) {
match = a2_.getLocalDescription().find("\r\na=rtcp-mux", match + 1); ASSERT_NE(match, std::string::npos);
match = a2_.getLocalDescription().find("\r\na=rtcp-mux", match + 1);
}
ASSERT_EQ(match, std::string::npos); ASSERT_EQ(match, std::string::npos);
ASSERT_TRUE_WAIT(a1_.IceCompleted() == true, kDefaultTimeout); ASSERT_TRUE_WAIT(a1_.IceCompleted() == true, kDefaultTimeout);
@ -2431,14 +2437,17 @@ TEST_F(SignalingTest, FullCallAudioNoMuxVideoMux)
a1_.CheckMediaPipeline(0, 0, PIPELINE_LOCAL | PIPELINE_SEND); a1_.CheckMediaPipeline(0, 0, PIPELINE_LOCAL | PIPELINE_SEND);
// Now check video mux. // Now check video mux.
a1_.CheckMediaPipeline(0, 1, a1_.CheckMediaPipeline(0, 1, fRtcpMux ?
PIPELINE_LOCAL | PIPELINE_RTCP_MUX | PIPELINE_SEND | PIPELINE_VIDEO); PIPELINE_LOCAL | PIPELINE_RTCP_MUX | PIPELINE_SEND | PIPELINE_VIDEO :
PIPELINE_LOCAL | PIPELINE_SEND | PIPELINE_VIDEO);
// The first Remote pipeline gets stored at 1 // The first Remote pipeline gets stored at 1
a2_.CheckMediaPipeline(0, 1, 0); a2_.CheckMediaPipeline(0, 1, 0);
// Now check video mux. // Now check video mux.
a2_.CheckMediaPipeline(0, 2, PIPELINE_RTCP_MUX | PIPELINE_VIDEO); a2_.CheckMediaPipeline(0, 2, fRtcpMux ?
PIPELINE_RTCP_MUX | PIPELINE_VIDEO :
PIPELINE_VIDEO);
} }
} // End namespace test. } // End namespace test.