Unify newapi::RtcpMode and RTCPMethod.

webrtc/video/video_send_stream_tests.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <algorithm>  // max
@@ skipping 314 matching lines @@
 
       // Send lossy receive reports to trigger FEC enabling.
       if (send_count_++ % 2 != 0) {
         // Receive statistics reporting having lost 50% of the packets.
         FakeReceiveStatistics lossy_receive_stats(
             kSendSsrcs[0], header.sequenceNumber, send_count_ / 2, 127);
         RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(),
                                &lossy_receive_stats, nullptr);
         EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
 
-        rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
+        rtcp_sender.SetRTCPStatus(kRtcpReducedSize);
         rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
 
         RTCPSender::FeedbackState feedback_state;
 
         EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
       }
 
       int encapsulated_payload_type = -1;
       if (header.payloadType == kRedPayloadType) {
         encapsulated_payload_type =
@@ skipping 61 matching lines @@
 
       // Nack second packet after receiving the third one.
       if (++send_count_ == 3) {
         uint16_t nack_sequence_number = header.sequenceNumber - 1;
         nacked_sequence_number_ = nack_sequence_number;
         NullReceiveStatistics null_stats;
         RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(), &null_stats,
                                nullptr);
         EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
 
-        rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
+        rtcp_sender.SetRTCPStatus(kRtcpReducedSize);
         rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
 
         RTCPSender::FeedbackState feedback_state;
 
         EXPECT_EQ(0,
                   rtcp_sender.SendRTCP(
                       feedback_state, kRtcpNack, 1, &nack_sequence_number));
       }
 
       uint16_t sequence_number = header.sequenceNumber;
@@ skipping 166 matching lines @@
     void TriggerLossReport(const RTPHeader& header) {
       // Send lossy receive reports to trigger FEC enabling.
       if (packet_count_++ % 2 != 0) {
         // Receive statistics reporting having lost 50% of the packets.
         FakeReceiveStatistics lossy_receive_stats(
             kSendSsrcs[0], header.sequenceNumber, packet_count_ / 2, 127);
         RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(),
                                &lossy_receive_stats, nullptr);
         EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
 
-        rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
+        rtcp_sender.SetRTCPStatus(kRtcpReducedSize);
         rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
 
         RTCPSender::FeedbackState feedback_state;
 
         EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
       }
     }
 
     virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
       // Increase frame size for next encoded frame, in the context of the
@@ skipping 209 matching lines @@
       kWaitingForStats
     };
 
     virtual void SendRtcpFeedback(int remb_value)
         EXCLUSIVE_LOCKS_REQUIRED(crit_) {
       FakeReceiveStatistics receive_stats(
           kSendSsrcs[0], last_sequence_number_, rtp_count_, 0);
       RTCPSender rtcp_sender(false, clock_, &receive_stats, nullptr);
       EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
 
-      rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
+      rtcp_sender.SetRTCPStatus(kRtcpReducedSize);
       rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
       if (remb_value > 0) {
         rtcp_sender.SetREMBStatus(true);
         rtcp_sender.SetREMBData(remb_value, std::vector<uint32_t>());
       }
       RTCPSender::FeedbackState feedback_state;
       EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
     }
 
     internal::TransportAdapter transport_adapter_;
@@ skipping 39 matching lines @@
       if (last_packet_time_ms_ > 0 &&
           clock_->TimeInMilliseconds() - last_packet_time_ms_ >
               kVideoMutedThresholdMs)
         observation_complete_->Set();
       // Receive statistics reporting having lost 50% of the packets.
       FakeReceiveStatistics receive_stats(kSendSsrcs[0], 1, 1, 0);
       RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(), &receive_stats,
                              nullptr);
       EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
 
-      rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
+      rtcp_sender.SetRTCPStatus(kRtcpReducedSize);
       rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
 
       RTCPSender::FeedbackState feedback_state;
 
       EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
       return SEND_PACKET;
     }
 
     void SetReceivers(PacketReceiver* send_transport_receiver,
                       PacketReceiver* receive_transport_receiver) override {
@@ skipping 40 matching lines @@
    public:
     BitrateObserver()
         : SendTest(kDefaultTimeoutMs),
           feedback_transport_(ReceiveTransport()),
           bitrate_capped_(false) {
       RtpRtcp::Configuration config;
       feedback_transport_.Enable();
       config.outgoing_transport = &feedback_transport_;
       rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
       rtp_rtcp_->SetREMBStatus(true);
-      rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound);
+      rtp_rtcp_->SetRTCPStatus(kRtcpReducedSize);
     }
 
     void OnStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       stream_ = send_stream;
     }
 
    private:
     DeliveryStatus DeliverPacket(MediaType media_type,
@@ skipping 932 matching lines @@
       EXPECT_TRUE(vp9videoHeader->flexible_mode);
       observation_complete_->Set();
     }
 
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc