Fix clang style errors in rtp_rtcp and dependant targets

Mark functions with override instead of virtual.
Add explicit non-trivial constructors/assign operators/destructors.
Define them in .cc files instead of inlining
use auto* instead of auto when deduced type is raw pointer

Bug: webrtc:163
Change-Id: I4d8a05d6a64fcc2ca16d02c5fcf9488fda832a6d
Reviewed-on: https://webrtc-review.googlesource.com/48781
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21927}

api/rtpreceiverinterface.cc

@@ -0,0 +1,48 @@
+/*
+ *  Copyright 2018 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 "api/rtpreceiverinterface.h"
+
+namespace webrtc {
+
+RtpSource::RtpSource(int64_t timestamp_ms,
+                     uint32_t source_id,
+                     RtpSourceType source_type)
+    : timestamp_ms_(timestamp_ms),
+      source_id_(source_id),
+      source_type_(source_type) {}
+
+RtpSource::RtpSource(int64_t timestamp_ms,
+                     uint32_t source_id,
+                     RtpSourceType source_type,
+                     uint8_t audio_level)
+    : timestamp_ms_(timestamp_ms),
+      source_id_(source_id),
+      source_type_(source_type),
+      audio_level_(audio_level) {}
+
+RtpSource::RtpSource(const RtpSource&) = default;
+RtpSource& RtpSource::operator=(const RtpSource&) = default;
+RtpSource::~RtpSource() = default;
+
+std::vector<rtc::scoped_refptr<MediaStreamInterface>>
+RtpReceiverInterface::streams() const {
+  return {};
+}
+
+std::vector<RtpSource> RtpReceiverInterface::GetSources() const {
+  return {};
+}
+
+int RtpReceiverInterface::AttachmentId() const {
+  return 0;
+}
+
+}  // namespace webrtc