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/mock/mock_delay_manager.h

@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2012 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_MOCK_MOCK_DELAY_MANAGER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_
+
+#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
+
+#include "webrtc/test/gmock.h"
+
+namespace webrtc {
+
+class MockDelayManager : public DelayManager {
+ public:
+  MockDelayManager(size_t max_packets_in_buffer,
+                   DelayPeakDetector* peak_detector,
+                   const TickTimer* tick_timer)
+      : DelayManager(max_packets_in_buffer, peak_detector, tick_timer) {}
+  virtual ~MockDelayManager() { Die(); }
+  MOCK_METHOD0(Die, void());
+  MOCK_CONST_METHOD0(iat_vector,
+      const IATVector&());
+  MOCK_METHOD3(Update,
+      int(uint16_t sequence_number, uint32_t timestamp, int sample_rate_hz));
+  MOCK_METHOD1(CalculateTargetLevel,
+      int(int iat_packets));
+  MOCK_METHOD1(SetPacketAudioLength,
+      int(int length_ms));
+  MOCK_METHOD0(Reset,
+      void());
+  MOCK_CONST_METHOD0(PeakFound,
+      bool());
+  MOCK_METHOD1(UpdateCounters,
+      void(int elapsed_time_ms));
+  MOCK_METHOD0(ResetPacketIatCount,
+      void());
+  MOCK_CONST_METHOD2(BufferLimits,
+      void(int* lower_limit, int* higher_limit));
+  MOCK_CONST_METHOD0(TargetLevel,
+      int());
+  MOCK_METHOD0(RegisterEmptyPacket, void());
+  MOCK_METHOD1(set_extra_delay_ms,
+      void(int16_t delay));
+  MOCK_CONST_METHOD0(base_target_level,
+      int());
+  MOCK_METHOD1(set_streaming_mode,
+      void(bool value));
+  MOCK_CONST_METHOD0(last_pack_cng_or_dtmf,
+      int());
+  MOCK_METHOD1(set_last_pack_cng_or_dtmf,
+      void(int value));
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_