/* * 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, int 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) { } // Destructor. virtual ~DecisionLogicNormal() {} 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). virtual Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer, const Expand& expand, int decoder_frame_length, const RTPHeader* packet_header, Modes prev_mode, bool play_dtmf, bool* reset_decoder); // 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, int 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_