Adding C++ versions of currently spec'd "RtpParameters" structs.

webrtc/pc/channel_unittest.cc

 /*
  *  Copyright 2009 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.
  */
 
@@ skipping 1941 matching lines @@
   bool SetRemoteContentWithBitrateLimit(int remote_limit) {
     typename T::Content content;
     CreateContent(0, kPcmuCodec, kH264Codec, &content);
     content.set_bandwidth(remote_limit);
     return channel1_->SetRemoteContent(&content, CA_OFFER, NULL);
   }
 
   webrtc::RtpParameters BitrateLimitedParameters(int limit) {
     webrtc::RtpParameters parameters;
     webrtc::RtpEncodingParameters encoding;
-    encoding.max_bitrate_bps = limit;
+    encoding.max_bitrate_bps = rtc::Optional<int>(limit);
     parameters.encodings.push_back(encoding);
     return parameters;
   }
 
   void VerifyMaxBitrate(const webrtc::RtpParameters& parameters,
                         int expected_bitrate) {
     EXPECT_EQ(1UL, parameters.encodings.size());
     EXPECT_EQ(expected_bitrate, parameters.encodings[0].max_bitrate_bps);
   }
 
@@ skipping 1801 matching lines @@
   ASSERT_TRUE(voice_channel_.Init_w(nullptr, nullptr, &fake_rtp_dtls_transport_,
                                     &fake_rtcp_dtls_transport_));
   EXPECT_DEATH(voice_channel_.SetTransports(&fake_rtp_dtls_transport_,
                                             &fake_rtp_dtls_transport_),
                "");
 }
 
 #endif  // RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
 
 // TODO(pthatcher): TestSetReceiver?