Adding 120 ms frame length support in 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 186 matching lines @@
                               int* max_num_packets) 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;
+  Operations last_operation_for_test() const;
 
  protected:
   static const int kOutputSizeMs = 10;
-  static const size_t kMaxFrameSize = 2880;  // 60 ms @ 48 kHz.
+  static const size_t kMaxFrameSize = 5760;  // 120 ms @ 48 kHz.
   // TODO(hlundin): Provide a better value for kSyncBufferSize.
   static const size_t kSyncBufferSize = 2 * kMaxFrameSize;
 
   // Inserts a new packet into NetEq. This is used by the InsertPacket method
   // above. Returns 0 on success, otherwise an error code.
   // TODO(hlundin): Merge this with InsertPacket above?
   int InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
                            rtc::ArrayView<const uint8_t> payload,
                            uint32_t receive_timestamp,
                            bool is_sync_packet)
@@ skipping 155 matching lines @@
   RandomVector random_vector_ GUARDED_BY(crit_sect_);
   std::unique_ptr<ComfortNoise> comfort_noise_ GUARDED_BY(crit_sect_);
   Rtcp rtcp_ GUARDED_BY(crit_sect_);
   StatisticsCalculator stats_ GUARDED_BY(crit_sect_);
   int fs_hz_ GUARDED_BY(crit_sect_);
   int fs_mult_ GUARDED_BY(crit_sect_);
   int last_output_sample_rate_hz_ GUARDED_BY(crit_sect_);
   size_t output_size_samples_ GUARDED_BY(crit_sect_);
   size_t decoder_frame_length_ GUARDED_BY(crit_sect_);
   Modes last_mode_ GUARDED_BY(crit_sect_);
+  Operations last_operation_ GUARDED_BY(crit_sect_);
   std::unique_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_);
   size_t decoded_buffer_length_ GUARDED_BY(crit_sect_);
   std::unique_ptr<int16_t[]> decoded_buffer_ GUARDED_BY(crit_sect_);
   uint32_t playout_timestamp_ GUARDED_BY(crit_sect_);
   bool new_codec_ GUARDED_BY(crit_sect_);
   uint32_t timestamp_ GUARDED_BY(crit_sect_);
   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_);
   std::unique_ptr<Nack> nack_ GUARDED_BY(crit_sect_);
   bool nack_enabled_ GUARDED_BY(crit_sect_);
   AudioFrame::VADActivity last_vad_activity_ GUARDED_BY(crit_sect_) =
       AudioFrame::kVadPassive;
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_