diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc index 16b741cef5..cc3c7593f7 100644 --- a/talk/media/webrtc/webrtcvideoengine2.cc +++ b/talk/media/webrtc/webrtcvideoengine2.cc @@ -1206,9 +1206,11 @@ void WebRtcVideoChannel2::OnPacketReceived( sp.ssrcs.push_back(ssrc); AddRecvStream(sp); - if (!call_->Receiver()->DeliverPacket( - reinterpret_cast(packet->data()), packet->length())) { - LOG(LS_WARNING) << "Failed to deliver RTP packet."; + if (call_->Receiver()->DeliverPacket( + reinterpret_cast(packet->data()), packet->length()) != + webrtc::PacketReceiver::DELIVERY_OK) { + LOG(LS_WARNING) << "Failed to deliver RTP packet after creating default " + "receiver."; return; } } @@ -1216,8 +1218,9 @@ void WebRtcVideoChannel2::OnPacketReceived( void WebRtcVideoChannel2::OnRtcpReceived( talk_base::Buffer* packet, const talk_base::PacketTime& packet_time) { - if (!call_->Receiver()->DeliverPacket( - reinterpret_cast(packet->data()), packet->length())) { + if (call_->Receiver()->DeliverPacket( + reinterpret_cast(packet->data()), packet->length()) != + webrtc::PacketReceiver::DELIVERY_OK) { LOG(LS_WARNING) << "Failed to deliver RTCP packet."; } }