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/audio_coding/neteq/tools/rtp_file_source.cc

@@ -0,0 +1,103 @@
+/*
+ *  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/audio_coding/neteq/tools/rtp_file_source.h"
+
+#include <assert.h>
+#include <string.h>
+#ifdef WIN32
+#include <winsock2.h>
+#else
+#include <netinet/in.h>
+#endif
+
+#include <memory>
+
+#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/test/rtp_file_reader.h"
+
+namespace webrtc {
+namespace test {
+
+RtpFileSource* RtpFileSource::Create(const std::string& file_name) {
+  RtpFileSource* source = new RtpFileSource();
+  RTC_CHECK(source->OpenFile(file_name));
+  return source;
+}
+
+bool RtpFileSource::ValidRtpDump(const std::string& file_name) {
+  std::unique_ptr<RtpFileReader> temp_file(
+      RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
+  return !!temp_file;
+}
+
+bool RtpFileSource::ValidPcap(const std::string& file_name) {
+  std::unique_ptr<RtpFileReader> temp_file(
+      RtpFileReader::Create(RtpFileReader::kPcap, file_name));
+  return !!temp_file;
+}
+
+RtpFileSource::~RtpFileSource() {
+}
+
+bool RtpFileSource::RegisterRtpHeaderExtension(RTPExtensionType type,
+                                               uint8_t id) {
+  assert(parser_.get());
+  return parser_->RegisterRtpHeaderExtension(type, id);
+}
+
+std::unique_ptr<Packet> RtpFileSource::NextPacket() {
+  while (true) {
+    RtpPacket temp_packet;
+    if (!rtp_reader_->NextPacket(&temp_packet)) {
+      return NULL;
+    }
+    if (temp_packet.original_length == 0) {
+      // May be an RTCP packet.
+      // Read the next one.
+      continue;
+    }
+    std::unique_ptr<uint8_t[]> packet_memory(new uint8_t[temp_packet.length]);
+    memcpy(packet_memory.get(), temp_packet.data, temp_packet.length);
+    std::unique_ptr<Packet> packet(new Packet(
+        packet_memory.release(), temp_packet.length,
+        temp_packet.original_length, temp_packet.time_ms, *parser_.get()));
+    if (!packet->valid_header()) {
+      continue;
+    }
+    if (filter_.test(packet->header().payloadType) ||
+        (use_ssrc_filter_ && packet->header().ssrc != ssrc_)) {
+      // This payload type should be filtered out. Continue to the next packet.
+      continue;
+    }
+    return packet;
+  }
+}
+
+RtpFileSource::RtpFileSource()
+    : PacketSource(),
+      parser_(RtpHeaderParser::Create()) {}
+
+bool RtpFileSource::OpenFile(const std::string& file_name) {
+  rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
+  if (rtp_reader_)
+    return true;
+  rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kPcap, file_name));
+  if (!rtp_reader_) {
+    FATAL() << "Couldn't open input file as either a rtpdump or .pcap. Note "
+               "that .pcapng is not supported.";
+  }
+  return true;
+}
+
+}  // namespace test
+}  // namespace webrtc