| Index: webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
|
| diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
|
| index 7c7db4dabc7e936a8ad7ee4b6fad16ecba031dbe..b53574c625c3ffdf6a4234ee2eb5ad0bb53d15e6 100644
|
| --- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
|
| +++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
|
| @@ -94,6 +94,8 @@ class RtpRtcpAPITest : public ::testing::Test {
|
| }
|
| ~RtpRtcpAPITest() {}
|
|
|
| + const uint32_t initial_ssrc = 8888;
|
| +
|
| void SetUp() override {
|
| RtpRtcp::Configuration configuration;
|
| configuration.audio = true;
|
| @@ -101,6 +103,7 @@ class RtpRtcpAPITest : public ::testing::Test {
|
| configuration.outgoing_transport = &null_transport_;
|
| configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
|
| module_.reset(RtpRtcp::CreateRtpRtcp(configuration));
|
| + module_->SetSSRC(initial_ssrc);
|
| rtp_payload_registry_.reset(new RTPPayloadRegistry());
|
| rtp_receiver_.reset(RtpReceiver::CreateAudioReceiver(
|
| &fake_clock_, NULL, NULL, rtp_payload_registry_.get()));
|
|
|