NetEq: Rename Nack to NackTracker to avoid name collisions in GN

webrtc/modules/audio_coding/neteq/nack.h

Index: webrtc/modules/audio_coding/neteq/nack.h
diff --git a/webrtc/modules/audio_coding/neteq/nack.h b/webrtc/modules/audio_coding/neteq/nack.h
deleted file mode 100644
index c46a85a7700876ce323025cf5c411bdc47c0b214..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/neteq/nack.h
+++ /dev/null
@@ -1,208 +0,0 @@
-/*
- *  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_NETEQ_NACK_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NACK_H_
-
-#include <vector>
-#include <map>
-
-#include "webrtc/base/gtest_prod_util.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
-
-//
-// The Nack class keeps track of the lost packets, an estimate of time-to-play
-// for each packet is also given.
-//
-// Every time a packet is pushed into NetEq, LastReceivedPacket() has to be
-// called to update the NACK list.
-//
-// Every time 10ms audio is pulled from NetEq LastDecodedPacket() should be
-// called, and time-to-play is updated at that moment.
-//
-// If packet N is received, any packet prior to |N - NackThreshold| which is not
-// arrived is considered lost, and should be labeled as "missing" (the size of
-// the list might be limited and older packet eliminated from the list). Packets
-// |N - NackThreshold|, |N - NackThreshold + 1|, ..., |N - 1| are considered
-// "late." A "late" packet with sequence number K is changed to "missing" any
-// time a packet with sequence number newer than |K + NackList| is arrived.
-//
-// The Nack class has to know about the sample rate of the packets to compute
-// time-to-play. So sample rate should be set as soon as the first packet is
-// received. If there is a change in the receive codec (sender changes codec)
-// then Nack should be reset. This is because NetEQ would flush its buffer and
-// 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 {
-
-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|
-  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
-  // API should be called every time 10 ms audio is pulled from NetEq.
-  void UpdateLastDecodedPacket(uint16_t sequence_number, uint32_t timestamp);
-
-  // Update the sequence number and the timestamp of the last received RTP. This
-  // API should be called every time a packet pushed into ACM.
-  void UpdateLastReceivedPacket(uint16_t sequence_number, uint32_t timestamp);
-
-  // Get a list of "missing" packets which have expected time-to-play larger
-  // than the given round-trip-time (in milliseconds).
-  // Note: Late packets are not included.
-  std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const;
-
-  // Reset to default values. The NACK list is cleared.
-  // |nack_threshold_packets_| & |max_nack_list_size_| preserve their values.
-  void Reset();
-
- private:
-  // This test need to access the private method GetNackList().
-  FRIEND_TEST_ALL_PREFIXES(NackTest, EstimateTimestampAndTimeToPlay);
-
-  struct NackElement {
-    NackElement(int64_t initial_time_to_play_ms,
-                uint32_t initial_timestamp,
-                bool missing)
-        : time_to_play_ms(initial_time_to_play_ms),
-          estimated_timestamp(initial_timestamp),
-          is_missing(missing) {}
-
-    // Estimated time (ms) left for this packet to be decoded. This estimate is
-    // updated every time jitter buffer decodes a packet.
-    int64_t time_to_play_ms;
-
-    // A guess about the timestamp of the missing packet, it is used for
-    // estimation of |time_to_play_ms|. The estimate might be slightly wrong if
-    // there has been frame-size change since the last received packet and the
-    // missing packet. However, the risk of this is low, and in case of such
-    // errors, there will be a minor misestimation in time-to-play of missing
-    // 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 {
-      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
-  // computed correctly.
-  NackList GetNackList() const;
-
-  // Given the |sequence_number_current_received_rtp| of currently received RTP,
-  // recognize packets which are not arrive and add to the list.
-  void AddToList(uint16_t sequence_number_current_received_rtp);
-
-  // This function subtracts 10 ms of time-to-play for all packets in NACK list.
-  // This is called when 10 ms elapsed with no new RTP packet decoded.
-  void UpdateEstimatedPlayoutTimeBy10ms();
-
-  // Given the |sequence_number_current_received_rtp| and
-  // |timestamp_current_received_rtp| of currently received RTP update number
-  // of samples per packet.
-  void UpdateSamplesPerPacket(uint16_t sequence_number_current_received_rtp,
-                              uint32_t timestamp_current_received_rtp);
-
-  // Given the |sequence_number_current_received_rtp| of currently received RTP
-  // update the list. That is; some packets will change from late to missing,
-  // some packets are inserted as missing and some inserted as late.
-  void UpdateList(uint16_t sequence_number_current_received_rtp);
-
-  // Packets which are considered late for too long (according to
-  // |nack_threshold_packets_|) are flagged as missing.
-  void ChangeFromLateToMissing(uint16_t sequence_number_current_received_rtp);
-
-  // Packets which have sequence number older that
-  // |sequence_num_last_received_rtp_| - |max_nack_list_size_| are removed
-  // from the NACK list.
-  void LimitNackListSize();
-
-  // Estimate timestamp of a missing packet given its sequence number.
-  uint32_t EstimateTimestamp(uint16_t sequence_number);
-
-  // Compute time-to-play given a timestamp.
-  int64_t TimeToPlay(uint32_t timestamp) const;
-
-  // If packet N is arrived, any packet prior to N - |nack_threshold_packets_|
-  // which is not arrived is considered missing, and should be in NACK list.
-  // Also any packet in the range of N-1 and N - |nack_threshold_packets_|,
-  // exclusive, which is not arrived is considered late, and should should be
-  // in the list of late packets.
-  const int nack_threshold_packets_;
-
-  // Valid if a packet is received.
-  uint16_t sequence_num_last_received_rtp_;
-  uint32_t timestamp_last_received_rtp_;
-  bool any_rtp_received_;  // If any packet received.
-
-  // Valid if a packet is decoded.
-  uint16_t sequence_num_last_decoded_rtp_;
-  uint32_t timestamp_last_decoded_rtp_;
-  bool any_rtp_decoded_;  // If any packet decoded.
-
-  int sample_rate_khz_;  // Sample rate in kHz.
-
-  // Number of samples per packet. We update this every time we receive a
-  // packet, not only for consecutive packets.
-  int samples_per_packet_;
-
-  // 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 webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NACK_H_