ACM: Move NACK functionality inside NetEq

webrtc/modules/audio_coding/neteq/nack.h

 /*
  *  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_MAIN_ACM2_NACK_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_NACK_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_NACK_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NACK_H_
 
 #include <vector>
 #include <map>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
 #include "webrtc/test/testsupport/gtest_prod_util.h"
 
 //
 // The Nack class keeps track of the lost packets, an estimate of time-to-play
@@ skipping 19 matching lines @@
 // re-transmission is meaning less for old packet. Therefore, in that case,
 // after reset the sampling rate has to be updated.
 //
 // Thread Safety
 // =============
 // Please note that this class in not thread safe. The class must be protected
 // if different APIs are called from different threads.
 //
 namespace webrtc {
 
-namespace acm2 {
-
 class Nack {
  public:
   // A limit for the size of the NACK list.
   static const size_t kNackListSizeLimit = 500;  // 10 seconds for 20 ms frame
                                                  // packets.
   // Factory method.
   static Nack* Create(int nack_threshold_packets);
 
   ~Nack();
 
   // Set a maximum for the size of the NACK list. If the last received packet
   // has sequence number of N, then NACK list will not contain any element
   // with sequence number earlier than N - |max_nack_list_size|.
   //
   // The largest maximum size is defined by |kNackListSizeLimit|
-  int SetMaxNackListSize(size_t max_nack_list_size);
+  void SetMaxNackListSize(size_t max_nack_list_size);
 
   // Set the sampling rate.
   //
   // If associated sampling rate of the received packets is changed, call this
   // function to update sampling rate. Note that if there is any change in
   // received codec then NetEq will flush its buffer and NACK has to be reset.
   // After Reset() is called sampling rate has to be set.
   void UpdateSampleRate(int sample_rate_hz);
 
   // Update the sequence number and the timestamp of the last decoded RTP. This
@@ skipping 37 matching lines @@
     // packets. This will have a very minor effect on NACK performance.
     uint32_t estimated_timestamp;
 
     // True if the packet is considered missing. Otherwise indicates packet is
     // late.
     bool is_missing;
   };
 
   class NackListCompare {
    public:
-    bool operator() (uint16_t sequence_number_old,
-                     uint16_t sequence_number_new) const {
+    bool operator()(uint16_t sequence_number_old,
+                    uint16_t sequence_number_new) const {
       return IsNewerSequenceNumber(sequence_number_new, sequence_number_old);
     }
   };
 
   typedef std::map<uint16_t, NackElement, NackListCompare> NackList;
 
   // Constructor.
   explicit Nack(int nack_threshold_packets);
 
   // This API is used only for testing to assess whether time-to-play is
@@ skipping 60 matching lines @@
   // A list of missing packets to be retransmitted. Components of the list
   // contain the sequence number of missing packets and the estimated time that
   // each pack is going to be played out.
   NackList nack_list_;
 
   // NACK list will not keep track of missing packets prior to
   // |sequence_num_last_received_rtp_| - |max_nack_list_size_|.
   size_t max_nack_list_size_;
 };
 
-}  // namespace acm2
-
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_NACK_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NACK_H_