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/neteq_input.h

@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
+
+#include <algorithm>
+#include <memory>
+#include <string>
+
+#include "webrtc/api/optional.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
+#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
+#include "webrtc/rtc_base/buffer.h"
+
+namespace webrtc {
+namespace test {
+
+// Interface class for input to the NetEqTest class.
+class NetEqInput {
+ public:
+  struct PacketData {
+    std::string ToString() const;
+
+    RTPHeader header;
+    rtc::Buffer payload;
+    double time_ms;
+  };
+
+  virtual ~NetEqInput() = default;
+
+  // Returns at what time (in ms) NetEq::InsertPacket should be called next, or
+  // empty if the source is out of packets.
+  virtual rtc::Optional<int64_t> NextPacketTime() const = 0;
+
+  // Returns at what time (in ms) NetEq::GetAudio should be called next, or
+  // empty if no more output events are available.
+  virtual rtc::Optional<int64_t> NextOutputEventTime() const = 0;
+
+  // Returns the time (in ms) for the next event from either NextPacketTime()
+  // or NextOutputEventTime(), or empty if both are out of events.
+  rtc::Optional<int64_t> NextEventTime() const {
+    const auto a = NextPacketTime();
+    const auto b = NextOutputEventTime();
+    // Return the minimum of non-empty |a| and |b|, or empty if both are empty.
+    if (a) {
+      return b ? rtc::Optional<int64_t>(std::min(*a, *b)) : a;
+    }
+    return b ? b : rtc::Optional<int64_t>();
+  }
+
+  // Returns the next packet to be inserted into NetEq. The packet following the
+  // returned one is pre-fetched in the NetEqInput object, such that future
+  // calls to NextPacketTime() or NextHeader() will return information from that
+  // packet.
+  virtual std::unique_ptr<PacketData> PopPacket() = 0;
+
+  // Move to the next output event. This will make NextOutputEventTime() return
+  // a new value (potentially the same if several output events share the same
+  // time).
+  virtual void AdvanceOutputEvent() = 0;
+
+  // Returns true if the source has come to an end. An implementation must
+  // eventually return true from this method, or the test will end up in an
+  // infinite loop.
+  virtual bool ended() const = 0;
+
+  // Returns the RTP header for the next packet, i.e., the packet that will be
+  // delivered next by PopPacket().
+  virtual rtc::Optional<RTPHeader> NextHeader() const = 0;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_