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/rtcp_packet.cc

@@ -0,0 +1,85 @@
+/*
+ *  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/rtcp_packet.h"
+
+#include "webrtc/rtc_base/checks.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr size_t RtcpPacket::kHeaderLength;
+
+rtc::Buffer RtcpPacket::Build() const {
+  rtc::Buffer packet(BlockLength());
+
+  size_t length = 0;
+  bool created = Create(packet.data(), &length, packet.capacity(), nullptr);
+  RTC_DCHECK(created) << "Invalid packet is not supported.";
+  RTC_DCHECK_EQ(length, packet.size())
+      << "BlockLength mispredicted size used by Create";
+
+  return packet;
+}
+
+bool RtcpPacket::BuildExternalBuffer(uint8_t* buffer,
+                                     size_t max_length,
+                                     PacketReadyCallback* callback) const {
+  size_t index = 0;
+  if (!Create(buffer, &index, max_length, callback))
+    return false;
+  return OnBufferFull(buffer, &index, callback);
+}
+
+bool RtcpPacket::OnBufferFull(uint8_t* packet,
+                              size_t* index,
+                              PacketReadyCallback* callback) const {
+  if (*index == 0)
+    return false;
+  RTC_DCHECK(callback) << "Fragmentation not supported.";
+  callback->OnPacketReady(packet, *index);
+  *index = 0;
+  return true;
+}
+
+size_t RtcpPacket::HeaderLength() const {
+  size_t length_in_bytes = BlockLength();
+  RTC_DCHECK_GT(length_in_bytes, 0);
+  RTC_DCHECK_EQ(length_in_bytes % 4, 0) << "Padding not supported";
+  // Length in 32-bit words without common header.
+  return (length_in_bytes - kHeaderLength) / 4;
+}
+
+// From RFC 3550, RTP: A Transport Protocol for Real-Time Applications.
+//
+// RTP header format.
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P| RC/FMT  |      PT       |             length            |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+void RtcpPacket::CreateHeader(
+    uint8_t count_or_format,  // Depends on packet type.
+    uint8_t packet_type,
+    size_t length,
+    uint8_t* buffer,
+    size_t* pos) {
+  RTC_DCHECK_LE(length, 0xffffU);
+  RTC_DCHECK_LE(count_or_format, 0x1f);
+  constexpr uint8_t kVersionBits = 2 << 6;
+  constexpr uint8_t kNoPaddingBit = 0 << 5;
+  buffer[*pos + 0] = kVersionBits | kNoPaddingBit | count_or_format;
+  buffer[*pos + 1] = packet_type;
+  buffer[*pos + 2] = (length >> 8) & 0xff;
+  buffer[*pos + 3] = length & 0xff;
+  *pos += kHeaderLength;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc