| /* | 
 |  *  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_DECISION_LOGIC_NORMAL_H_ | 
 | #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_ | 
 |  | 
 | #include "webrtc/base/constructormagic.h" | 
 | #include "webrtc/modules/audio_coding/neteq/decision_logic.h" | 
 | #include "webrtc/typedefs.h" | 
 |  | 
 | namespace webrtc { | 
 |  | 
 | // Implementation of the DecisionLogic class for playout modes kPlayoutOn and | 
 | // 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) { | 
 |   } | 
 |  | 
 |  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 | 
 |   // be NULL. The mode resulting form the last call to NetEqImpl::GetAudio is | 
 |   // supplied in |prev_mode|. If there is a DTMF event to play, |play_dtmf| | 
 |   // should be set to true. The output variable |reset_decoder| will be set to | 
 |   // true if a reset is required; otherwise it is left unchanged (i.e., it can | 
 |   // remain true if it was true before the call). | 
 |   Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer, | 
 |                                     const Expand& expand, | 
 |                                     size_t decoder_frame_length, | 
 |                                     const RTPHeader* packet_header, | 
 |                                     Modes prev_mode, | 
 |                                     bool play_dtmf, | 
 |                                     bool* reset_decoder) override; | 
 |  | 
 |   // Returns the operation to do given that the expected packet is not | 
 |   // available, but a packet further into the future is at hand. | 
 |   virtual Operations FuturePacketAvailable( | 
 |       const SyncBuffer& sync_buffer, | 
 |       const Expand& expand, | 
 |       size_t decoder_frame_length, | 
 |       Modes prev_mode, | 
 |       uint32_t target_timestamp, | 
 |       uint32_t available_timestamp, | 
 |       bool play_dtmf); | 
 |  | 
 |   // Returns the operation to do given that the expected packet is available. | 
 |   virtual Operations ExpectedPacketAvailable(Modes prev_mode, bool play_dtmf); | 
 |  | 
 |   // Returns the operation given that no packets are available (except maybe | 
 |   // a DTMF event, flagged by setting |play_dtmf| true). | 
 |   virtual Operations NoPacket(bool play_dtmf); | 
 |  | 
 |  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); | 
 |  | 
 |   // 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; } | 
 |  | 
 |   // 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; | 
 |  | 
 |   DISALLOW_COPY_AND_ASSIGN(DecisionLogicNormal); | 
 | }; | 
 |  | 
 | }  // namespace webrtc | 
 | #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_ |