Create //webrtc/api:libjingle_peerconnection_api + refactorings.

Create a new target //webrtc/api:libjingle_peerconnection_api and start moving
things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection
to //webrtc/pc:libjingle_peerconnection.

Moved the RTCStatsCollectorCallback into its own header file, so that
PeerConnectionInterface can include that instead of pulling in
RTCStatsCollector and PeerConnection and everything.

Separated cricket::MediaType into its own header/source set, so that it
can be used in the api.

BUG=webrtc:5883

Review-Url: https://codereview.webrtc.org/2514883002
Cr-Commit-Position: refs/heads/master@{#16210}

webrtc/pc/mediaconstraintsinterface_unittest.cc

@@ -0,0 +1,75 @@
+/*
+ *  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