ACM: Move NACK functionality inside NetEq

webrtc/modules/audio_coding/neteq/neteq_impl.h

 /*
  *  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.
  */
 
@@ skipping 21 matching lines @@
 class ComfortNoise;
 class CriticalSectionWrapper;
 class DecisionLogic;
 class DecoderDatabase;
 class DelayManager;
 class DelayPeakDetector;
 class DtmfBuffer;
 class DtmfToneGenerator;
 class Expand;
 class Merge;
+class Nack;
 class Normal;
 class PacketBuffer;
 class PayloadSplitter;
 class PostDecodeVad;
 class PreemptiveExpand;
 class RandomVector;
 class SyncBuffer;
 class TimestampScaler;
 struct AccelerateFactory;
 struct DtmfEvent;
@@ skipping 128 matching lines @@
   // When LastError() returns kDecoderErrorCode or kComfortNoiseErrorCode, check
   // this method to get the decoder's error code.
   int LastDecoderError() override;
 
   // Flushes both the packet buffer and the sync buffer.
   void FlushBuffers() override;
 
   void PacketBufferStatistics(int* current_num_packets,
                               int* max_num_packets) const override;
 
-  // Get sequence number and timestamp of the latest RTP.
-  // This method is to facilitate NACK.
-  int DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const override;
+  void EnableNack(size_t max_nack_list_size) override;
+
+  void DisableNack() override;
+
+  std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
 
   // This accessor method is only intended for testing purposes.
   const SyncBuffer* sync_buffer_for_test() const;
 
  protected:
   static const int kOutputSizeMs = 10;
   static const size_t kMaxFrameSize = 2880;  // 60 ms @ 48 kHz.
   // TODO(hlundin): Provide a better value for kSyncBufferSize.
   static const size_t kSyncBufferSize = 2 * kMaxFrameSize;
 
@@ skipping 183 matching lines @@
   bool reset_decoder_ GUARDED_BY(crit_sect_);
   uint8_t current_rtp_payload_type_ GUARDED_BY(crit_sect_);
   uint8_t current_cng_rtp_payload_type_ GUARDED_BY(crit_sect_);
   uint32_t ssrc_ GUARDED_BY(crit_sect_);
   bool first_packet_ GUARDED_BY(crit_sect_);
   int error_code_ GUARDED_BY(crit_sect_);  // Store last error code.
   int decoder_error_code_ GUARDED_BY(crit_sect_);
   const BackgroundNoiseMode background_noise_mode_ GUARDED_BY(crit_sect_);
   NetEqPlayoutMode playout_mode_ GUARDED_BY(crit_sect_);
   bool enable_fast_accelerate_ GUARDED_BY(crit_sect_);
-
-  // These values are used by NACK module to estimate time-to-play of
-  // a missing packet. Occasionally, NetEq might decide to decode more
-  // than one packet. Therefore, these values store sequence number and
-  // timestamp of the first packet pulled from the packet buffer. In
-  // such cases, these values do not exactly represent the sequence number
-  // or timestamp associated with a 10ms audio pulled from NetEq. NACK
-  // module is designed to compensate for this.
-  int decoded_packet_sequence_number_ GUARDED_BY(crit_sect_);
-  uint32_t decoded_packet_timestamp_ GUARDED_BY(crit_sect_);
+  rtc::scoped_ptr<Nack> nack_ GUARDED_BY(crit_sect_);
+  bool nack_enabled_ GUARDED_BY(crit_sect_);
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_