| 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;
|
| }
|
|
|