Index: webrtc/test/call_test.cc |
diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc |
index aadcb8d5b89dcf320b00eb51ab2b41504aa23cc5..ed8a18102a57c2a43fb5ace98c030afeeb7c6fc9 100644 |
--- a/webrtc/test/call_test.cc |
+++ b/webrtc/test/call_test.cc |
@@ -16,7 +16,6 @@ |
#include "webrtc/config.h" |
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h" |
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
-#include "webrtc/test/call_test.h" |
#include "webrtc/test/testsupport/fileutils.h" |
#include "webrtc/voice_engine/include/voe_base.h" |
@@ -276,11 +275,11 @@ void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) { |
// TODO(brandtr): Update this when we support multistream protection. |
RTC_DCHECK(num_flexfec_streams_ <= 1); |
if (num_flexfec_streams_ == 1) { |
- FlexfecReceiveStream::Config flexfec_config; |
- flexfec_config.flexfec_payload_type = kFlexfecPayloadType; |
- flexfec_config.flexfec_ssrc = kFlexfecSendSsrc; |
- flexfec_config.protected_media_ssrcs = {kVideoSendSsrcs[0]}; |
- flexfec_receive_configs_.push_back(flexfec_config); |
+ FlexfecReceiveStream::Config config; |
+ config.payload_type = kFlexfecPayloadType; |
+ config.remote_ssrc = kFlexfecSendSsrc; |
+ config.protected_media_ssrcs = {kVideoSendSsrcs[0]}; |
+ flexfec_receive_configs_.push_back(config); |
} |
} |