NetEq: Replace timescale_holdoff_ with a Countdown timer

webrtc/modules/audio_coding/neteq/decision_logic_normal.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.
  */
 
@@ skipping 10 matching lines @@
 // kPlayoutStreaming.
 class DecisionLogicNormal : public DecisionLogic {
  public:
   // Constructor.
   DecisionLogicNormal(int fs_hz,
                       size_t output_size_samples,
                       NetEqPlayoutMode playout_mode,
                       DecoderDatabase* decoder_database,
                       const PacketBuffer& packet_buffer,
                       DelayManager* delay_manager,
-                      BufferLevelFilter* buffer_level_filter)
-      : DecisionLogic(fs_hz, output_size_samples, playout_mode,
-                      decoder_database, packet_buffer, delay_manager,
-                      buffer_level_filter) {
-  }
+                      BufferLevelFilter* buffer_level_filter,
+                      const TickTimer* tick_timer)
+      : DecisionLogic(fs_hz,
+                      output_size_samples,
+                      playout_mode,
+                      decoder_database,
+                      packet_buffer,
+                      delay_manager,
+                      buffer_level_filter,
+                      tick_timer) {}
 
  protected:
   static const int kAllowMergeWithoutExpandMs = 20;  // 20 ms.
   static const int kReinitAfterExpands = 100;
   static const int kMaxWaitForPacket = 10;
 
   // Returns the operation that should be done next. |sync_buffer| and |expand|
   // are provided for reference. |decoder_frame_length| is the number of samples
   // obtained from the last decoded frame. If there is a packet available, the
   // packet header should be supplied in |packet_header|; otherwise it should
@@ skipping 33 matching lines @@
  private:
   // Returns the operation given that the next available packet is a comfort
   // noise payload (RFC 3389 only, not codec-internal).
   Operations CngOperation(Modes prev_mode,
                           uint32_t target_timestamp,
                           uint32_t available_timestamp,
                           size_t generated_noise_samples);
 
   // Checks if enough time has elapsed since the last successful timescale
   // operation was done (i.e., accelerate or preemptive expand).
-  bool TimescaleAllowed() const { return timescale_hold_off_ == 0; }
+  bool TimescaleAllowed() const {
+    return !timescale_countdown_ || timescale_countdown_->Finished();
+  }
 
   // Checks if the current (filtered) buffer level is under the target level.
   bool UnderTargetLevel() const;
 
   // Checks if |timestamp_leap| is so long into the future that a reset due
   // to exceeding kReinitAfterExpands will be done.
   bool ReinitAfterExpands(uint32_t timestamp_leap) const;
 
   // Checks if we still have not done enough expands to cover the distance from
   // the last decoded packet to the next available packet, the distance beeing
   // conveyed in |timestamp_leap|.
   bool PacketTooEarly(uint32_t timestamp_leap) const;
 
   // Checks if num_consecutive_expands_ >= kMaxWaitForPacket.
   bool MaxWaitForPacket() const;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogicNormal);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_