Embed a cricket::MediaConfig in RTCConfiguration.

webrtc/api/mediaconstraintsinterface_unittest.cc

 /*
  *  Copyright 2016 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 "webrtc/api/mediaconstraintsinterface.h"
 
 #include "webrtc/api/test/fakeconstraints.h"
 #include "webrtc/base/gunit.h"
 
 namespace webrtc {
 
 namespace {
 
+// Checks all settings touched by CopyConstraintsIntoRtcConfiguration,
+// plus audio_jitter_buffer_max_packets.
 bool Matches(const PeerConnectionInterface::RTCConfiguration& a,
              const PeerConnectionInterface::RTCConfiguration& b) {
-  return a.audio_jitter_buffer_max_packets ==
+  return a.disable_ipv6 == b.disable_ipv6 &&
+         a.audio_jitter_buffer_max_packets ==
              b.audio_jitter_buffer_max_packets &&
-         a.disable_prerenderer_smoothing == b.disable_prerenderer_smoothing;
+         a.enable_rtp_data_channel == b.enable_rtp_data_channel &&
+         a.screencast_min_bitrate == b.screencast_min_bitrate &&
+         a.combined_audio_video_bwe == b.combined_audio_video_bwe &&
+         a.enable_dtls_srtp == b.enable_dtls_srtp &&
+         a.media_config.enable_dscp == b.media_config.enable_dscp &&
+         a.media_config.video.enable_cpu_overuse_detection ==
+             b.media_config.video.enable_cpu_overuse_detection &&
+         a.media_config.video.disable_prerenderer_smoothing ==
+             b.media_config.video.disable_prerenderer_smoothing &&
+         a.media_config.video.suspend_below_min_bitrate ==
+             b.media_config.video.suspend_below_min_bitrate;
 }
 
 TEST(MediaConstraintsInterface, CopyConstraintsIntoRtcConfiguration) {
   FakeConstraints constraints;
   PeerConnectionInterface::RTCConfiguration old_configuration;
   PeerConnectionInterface::RTCConfiguration configuration;
 
   CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
   EXPECT_TRUE(Matches(old_configuration, configuration));
 
@@ skipping 18 matching lines @@
   configuration.audio_jitter_buffer_max_packets = 34;
   CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
   EXPECT_EQ(34, configuration.audio_jitter_buffer_max_packets);
   ASSERT_TRUE(configuration.enable_dtls_srtp);
   EXPECT_TRUE(*(configuration.enable_dtls_srtp));
 }
 
 }  // namespace
 
 }  // namespace webrtc