Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/rtp_rtcp/source/rtp_format.cc

@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2014 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/modules/rtp_rtcp/source/rtp_format.h"
+
+#include <utility>
+
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h"
+
+namespace webrtc {
+RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type,
+                                     size_t max_payload_len,
+                                     size_t last_packet_reduction_len,
+                                     const RTPVideoTypeHeader* rtp_type_header,
+                                     FrameType frame_type) {
+  switch (type) {
+    case kRtpVideoH264:
+      RTC_CHECK(rtp_type_header);
+      return new RtpPacketizerH264(max_payload_len, last_packet_reduction_len,
+                                   rtp_type_header->H264.packetization_mode);
+    case kRtpVideoVp8:
+      RTC_CHECK(rtp_type_header);
+      return new RtpPacketizerVp8(rtp_type_header->VP8, max_payload_len,
+                                  last_packet_reduction_len);
+    case kRtpVideoVp9:
+      RTC_CHECK(rtp_type_header);
+      return new RtpPacketizerVp9(rtp_type_header->VP9, max_payload_len,
+                                  last_packet_reduction_len);
+    case kRtpVideoGeneric:
+      return new RtpPacketizerGeneric(frame_type, max_payload_len,
+                                      last_packet_reduction_len);
+    case kRtpVideoNone:
+      RTC_NOTREACHED();
+  }
+  return nullptr;
+}
+
+RtpDepacketizer* RtpDepacketizer::Create(RtpVideoCodecTypes type) {
+  switch (type) {
+    case kRtpVideoH264:
+      return new RtpDepacketizerH264();
+    case kRtpVideoVp8:
+      return new RtpDepacketizerVp8();
+    case kRtpVideoVp9:
+      return new RtpDepacketizerVp9();
+    case kRtpVideoGeneric:
+      return new RtpDepacketizerGeneric();
+    case kRtpVideoNone:
+      assert(false);
+  }
+  return nullptr;
+}
+}  // namespace webrtc