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

webrtc/pc/rtpsenderreceiver_unittest.cc

 /*
  *  Copyright 2012 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 557 matching lines @@
 
   DestroyAudioRtpSender();
 }
 
 TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) {
   CreateAudioRtpSender();
 
   EXPECT_EQ(-1, voice_media_channel_->max_bps());
   webrtc::RtpParameters params = audio_rtp_sender_->GetParameters();
   EXPECT_EQ(1, params.encodings.size());
-  EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps);
-  params.encodings[0].max_bitrate_bps = 1000;
+  EXPECT_FALSE(params.encodings[0].max_bitrate_bps);
+  params.encodings[0].max_bitrate_bps = rtc::Optional<int>(1000);
   EXPECT_TRUE(audio_rtp_sender_->SetParameters(params));
 
   // Read back the parameters and verify they have been changed.
   params = audio_rtp_sender_->GetParameters();
   EXPECT_EQ(1, params.encodings.size());
-  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+  EXPECT_EQ(rtc::Optional<int>(1000), params.encodings[0].max_bitrate_bps);
 
   // Verify that the audio channel received the new parameters.
   params = voice_media_channel_->GetRtpSendParameters(kAudioSsrc);
   EXPECT_EQ(1, params.encodings.size());
-  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+  EXPECT_EQ(rtc::Optional<int>(1000), params.encodings[0].max_bitrate_bps);
 
   // Verify that the global bitrate limit has not been changed.
   EXPECT_EQ(-1, voice_media_channel_->max_bps());
 
   DestroyAudioRtpSender();
 }
 
 TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) {
   CreateVideoRtpSender();
 
   RtpParameters params = video_rtp_sender_->GetParameters();
   EXPECT_EQ(1u, params.encodings.size());
   EXPECT_TRUE(video_rtp_sender_->SetParameters(params));
 
   DestroyVideoRtpSender();
 }
 
 TEST_F(RtpSenderReceiverTest, SetVideoMaxSendBitrate) {
   CreateVideoRtpSender();
 
   EXPECT_EQ(-1, video_media_channel_->max_bps());
   webrtc::RtpParameters params = video_rtp_sender_->GetParameters();
   EXPECT_EQ(1, params.encodings.size());
-  EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps);
-  params.encodings[0].max_bitrate_bps = 1000;
+  EXPECT_FALSE(params.encodings[0].max_bitrate_bps);
+  params.encodings[0].max_bitrate_bps = rtc::Optional<int>(1000);
   EXPECT_TRUE(video_rtp_sender_->SetParameters(params));
 
   // Read back the parameters and verify they have been changed.
   params = video_rtp_sender_->GetParameters();
   EXPECT_EQ(1, params.encodings.size());
-  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+  EXPECT_EQ(rtc::Optional<int>(1000), params.encodings[0].max_bitrate_bps);
 
   // Verify that the video channel received the new parameters.
   params = video_media_channel_->GetRtpSendParameters(kVideoSsrc);
   EXPECT_EQ(1, params.encodings.size());
-  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+  EXPECT_EQ(rtc::Optional<int>(1000), params.encodings[0].max_bitrate_bps);
 
   // Verify that the global bitrate limit has not been changed.
   EXPECT_EQ(-1, video_media_channel_->max_bps());
 
   DestroyVideoRtpSender();
 }
 
 TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetParameters) {
   CreateAudioRtpReceiver();
 
@@ skipping 161 matching lines @@
 // Make sure the signal from "GetOnDestroyedSignal()" fires when the sender is
 // destroyed, which is needed for the DTMF sender.
 TEST_F(RtpSenderReceiverTest, TestOnDestroyedSignal) {
   CreateAudioRtpSender();
   EXPECT_FALSE(audio_sender_destroyed_signal_fired_);
   audio_rtp_sender_ = nullptr;
   EXPECT_TRUE(audio_sender_destroyed_signal_fired_);
 }
 
 }  // namespace webrtc