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diff --git a/webrtc/modules/audio_coding/acm2/initial_delay_manager.h b/webrtc/modules/audio_coding/acm2/initial_delay_manager.h
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+++ b/webrtc/modules/audio_coding/acm2/initial_delay_manager.h
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+/*
+ *  Copyright (c) 2013 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_ACM2_INITIAL_DELAY_MANAGER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/include/module_common_types.h"
+
+namespace webrtc {
+
+namespace acm2 {
+
+class InitialDelayManager {
+ public:
+  enum PacketType {
+    kUndefinedPacket, kCngPacket, kAvtPacket, kAudioPacket, kSyncPacket };
+
+  // Specifies a stream of sync-packets.
+  struct SyncStream {
+    SyncStream()
+        : num_sync_packets(0),
+          receive_timestamp(0),
+          timestamp_step(0) {
+      memset(&rtp_info, 0, sizeof(rtp_info));
+    }
+
+    int num_sync_packets;
+
+    // RTP header of the first sync-packet in the sequence.
+    WebRtcRTPHeader rtp_info;
+
+    // Received timestamp of the first sync-packet in the sequence.
+    uint32_t receive_timestamp;
+
+    // Samples per packet.
+    uint32_t timestamp_step;
+  };
+
+  InitialDelayManager(int initial_delay_ms, int late_packet_threshold);
+
+  // Update with the last received RTP header, |header|, and received timestamp,
+  // |received_timestamp|. |type| indicates the packet type. If codec is changed
+  // since the last time |new_codec| should be true. |sample_rate_hz| is the
+  // decoder's sampling rate in Hz. |header| has a field to store sampling rate
+  // but we are not sure if that is properly set at the send side, and |header|
+  // is declared constant in the caller of this function
+  // (AcmReceiver::InsertPacket()). |sync_stream| contains information required
+  // to generate a stream of sync packets.
+  void UpdateLastReceivedPacket(const WebRtcRTPHeader& header,
+                                uint32_t receive_timestamp,
+                                PacketType type,
+                                bool new_codec,
+                                int sample_rate_hz,
+                                SyncStream* sync_stream);
+
+  // Based on the last received timestamp and given the current timestamp,
+  // sequence of late (or perhaps missing) packets is computed.
+  void LatePackets(uint32_t timestamp_now, SyncStream* sync_stream);
+
+  // Get playout timestamp.
+  // Returns true if the timestamp is valid (when buffering), otherwise false.
+  bool GetPlayoutTimestamp(uint32_t* playout_timestamp);
+
+  // True if buffered audio is less than the given initial delay (specified at
+  // the constructor). Buffering might be disabled by the client of this class.
+  bool buffering() { return buffering_; }
+
+  // Disable buffering in the class.
+  void DisableBuffering();
+
+  // True if any packet received for buffering.
+  bool PacketBuffered() { return last_packet_type_ != kUndefinedPacket; }
+
+ private:
+  static const uint8_t kInvalidPayloadType = 0xFF;
+
+  // Update playout timestamps. While buffering, this is about
+  // |initial_delay_ms| millisecond behind the latest received timestamp.
+  void UpdatePlayoutTimestamp(const RTPHeader& current_header,
+                              int sample_rate_hz);
+
+  // Record an RTP headr and related parameter
+  void RecordLastPacket(const WebRtcRTPHeader& rtp_info,
+                        uint32_t receive_timestamp,
+                        PacketType type);
+
+  PacketType last_packet_type_;
+  WebRtcRTPHeader last_packet_rtp_info_;
+  uint32_t last_receive_timestamp_;
+  uint32_t timestamp_step_;
+  uint8_t audio_payload_type_;
+  const int initial_delay_ms_;
+  int buffered_audio_ms_;
+  bool buffering_;
+
+  // During the initial phase where packets are being accumulated and silence
+  // is played out, |playout_ts| is a timestamp which is equal to
+  // |initial_delay_ms_| milliseconds earlier than the most recently received
+  // RTP timestamp.
+  uint32_t playout_timestamp_;
+
+  // If the number of late packets exceed this value (computed based on current
+  // timestamp and last received timestamp), sequence of sync-packets is
+  // specified.
+  const int late_packet_threshold_;
+};
+
+}  // namespace acm2
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_