Disable reduced-size RTCP in default config.

webrtc/video/end_to_end_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>
@@ skipping 2771 matching lines @@
     VideoReceiveStream::Decoder decoder;
     decoder.decoder = &fake_decoder;
     decoder.payload_type = 123;
     decoder.payload_name = "FAKE";
     receive_config.decoders.push_back(decoder);
     VideoReceiveStream* receive_stream =
         call->CreateVideoReceiveStream(receive_config);
     call->DestroyVideoReceiveStream(receive_stream);
   }
 }
+
+void VerifyEmptyNackConfig(const NackConfig& config) {
+  EXPECT_EQ(0, config.rtp_history_ms)
+      << "Enabling NACK requires rtcp-fb: nack negotiation.";
+}
+
+void VerifyEmptyFecConfig(const FecConfig& config) {
+  EXPECT_EQ(-1, config.ulpfec_payload_type)
+      << "Enabling FEC requires rtpmap: ulpfec negotiation.";
+  EXPECT_EQ(-1, config.red_payload_type)
+      << "Enabling FEC requires rtpmap: red negotiation.";
+  EXPECT_EQ(-1, config.red_rtx_payload_type)
+      << "Enabling RTX in FEC requires rtpmap: rtx negotiation.";
+}
+
+TEST_F(EndToEndTest, DefaultSendConfigDoesNotContainExtensions) {

[mflodman, 2015/06/05 11:50:03]

I'd prefer another name than Extensions, since this does more than that. Maybe
something around verification of default packets? Or RTP?

[pbos-webrtc, 2015/06/05 12:01:56]

Done.

+  VideoSendStream::Config default_send_config;
+  EXPECT_EQ(0, default_send_config.rtp.nack.rtp_history_ms)
+      << "Enabling NACK require rtcp-fb: nack negotiation.";
+  EXPECT_TRUE(default_send_config.rtp.rtx.ssrcs.empty())
+      << "Enabling RTX requires rtpmap: rtx negotiation.";
+  EXPECT_TRUE(default_send_config.rtp.extensions.empty())
+      << "Enabling RTP extensions require negotiation.";
+
+  VerifyEmptyNackConfig(default_send_config.rtp.nack);
+  VerifyEmptyFecConfig(default_send_config.rtp.fec);
+}
+
+TEST_F(EndToEndTest, DefaultReceiveConfigDoesNotContainExtensions) {
+  VideoReceiveStream::Config default_receive_config;
+  EXPECT_EQ(newapi::kRtcpCompound, default_receive_config.rtp.rtcp_mode)
+      << "Reduced-size RTCP require rtcp-rsize to be negotiated.";
+  EXPECT_FALSE(default_receive_config.rtp.remb)
+      << "REMB require rtcp-fb: goog-remb to be negotiated.";
+  EXPECT_FALSE(
+      default_receive_config.rtp.rtcp_xr.receiver_reference_time_report)
+      << "RTCP XR settings require rtcp-xr to be negotiated.";
+  EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
+      << "Enabling RTX requires rtpmap: rtx negotiation.";
+  EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
+      << "Enabling RTP extensions require negotiation.";
+
+  VerifyEmptyNackConfig(default_receive_config.rtp.nack);
+  VerifyEmptyFecConfig(default_receive_config.rtp.fec);
+}
+
 }  // namespace webrtc