Create //webrtc/api:libjingle_peerconnection_api + refactorings.

webrtc/api/mediaconstraintsinterface_unittest.cc

Index: webrtc/api/mediaconstraintsinterface_unittest.cc
diff --git a/webrtc/api/mediaconstraintsinterface_unittest.cc b/webrtc/api/mediaconstraintsinterface_unittest.cc
deleted file mode 100644
index dcf4bb7fde700207f4565ea1dafbd38f6824348c..0000000000000000000000000000000000000000
--- a/webrtc/api/mediaconstraintsinterface_unittest.cc
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- *  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.disable_ipv6 == b.disable_ipv6 &&
-         a.audio_jitter_buffer_max_packets ==
-             b.audio_jitter_buffer_max_packets &&
-         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));
-
-  constraints.SetMandatory(MediaConstraintsInterface::kEnableIPv6, "true");
-  CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
-  EXPECT_FALSE(configuration.disable_ipv6);
-  constraints.SetMandatory(MediaConstraintsInterface::kEnableIPv6, "false");
-  CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
-  EXPECT_TRUE(configuration.disable_ipv6);
-
-  constraints.SetMandatory(MediaConstraintsInterface::kScreencastMinBitrate,
-                           27);
-  CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
-  EXPECT_TRUE(configuration.screencast_min_bitrate);
-  EXPECT_EQ(27, *(configuration.screencast_min_bitrate));
-
-  // An empty set of constraints will not overwrite
-  // values that are already present.
-  constraints = FakeConstraints();
-  configuration = old_configuration;
-  configuration.enable_dtls_srtp = rtc::Optional<bool>(true);
-  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