Index: webrtc/modules/video_coding/main/test/rtp_player.cc |
diff --git a/webrtc/modules/video_coding/main/test/rtp_player.cc b/webrtc/modules/video_coding/main/test/rtp_player.cc |
index e53a4ab9c890876d261c708e82a40adcef77b1ff..74a5b95877cc3484df67e8d831d73628f503f3eb 100644 |
--- a/webrtc/modules/video_coding/main/test/rtp_player.cc |
+++ b/webrtc/modules/video_coding/main/test/rtp_player.cc |
@@ -217,11 +217,10 @@ class SsrcHandlers { |
RtpRtcp::Configuration configuration; |
configuration.clock = clock; |
- configuration.id = 1; |
configuration.audio = false; |
handler->rtp_module_.reset(RtpReceiver::CreateVideoReceiver( |
- configuration.id, configuration.clock, handler->payload_sink_.get(), |
- NULL, handler->rtp_payload_registry_.get())); |
+ configuration.clock, handler->payload_sink_.get(), NULL, |
+ handler->rtp_payload_registry_.get())); |
if (handler->rtp_module_.get() == NULL) { |
return -1; |
} |