/* * 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 MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_ #define MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_ #include #include "api/neteq/neteq.h" #include "api/neteq/neteq_controller.h" #include "api/neteq/tick_timer.h" #include "modules/audio_coding/neteq/buffer_level_filter.h" #include "modules/audio_coding/neteq/delay_manager.h" #include "modules/audio_coding/neteq/packet_arrival_history.h" #include "rtc_base/experiments/field_trial_parser.h" namespace webrtc { // This is the class for the decision tree implementation. class DecisionLogic : public NetEqController { public: DecisionLogic(NetEqController::Config config); DecisionLogic(NetEqController::Config config, std::unique_ptr delay_manager, std::unique_ptr buffer_level_filter); ~DecisionLogic() override; DecisionLogic(const DecisionLogic&) = delete; DecisionLogic& operator=(const DecisionLogic&) = delete; // Not used. void Reset() override {} // Resets parts of the state. Typically done when switching codecs. void SoftReset() override; // Sets the sample rate and the output block size. void SetSampleRate(int fs_hz, size_t output_size_samples) override; // Given info about the latest received packet, and current jitter buffer // status, returns the operation. `target_timestamp` and `expand_mutefactor` // are provided for reference. `last_packet_samples` is the number of samples // obtained from the last decoded frame. If there is a packet available, it // should be supplied in `packet`; otherwise it should be NULL. The mode // resulting from the last call to NetEqImpl::GetAudio is supplied in // `last_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). NetEq::Operation GetDecision(const NetEqController::NetEqStatus& status, bool* reset_decoder) override; void ExpandDecision(NetEq::Operation operation) override {} // Adds `value` to `sample_memory_`. void AddSampleMemory(int32_t value) override { sample_memory_ += value; } int TargetLevelMs() const override; int UnlimitedTargetLevelMs() const override; absl::optional PacketArrived(int fs_hz, bool should_update_stats, const PacketArrivedInfo& info) override; void RegisterEmptyPacket() override {} bool SetMaximumDelay(int delay_ms) override { return delay_manager_->SetMaximumDelay(delay_ms); } bool SetMinimumDelay(int delay_ms) override { return delay_manager_->SetMinimumDelay(delay_ms); } bool SetBaseMinimumDelay(int delay_ms) override { return delay_manager_->SetBaseMinimumDelay(delay_ms); } int GetBaseMinimumDelay() const override { return delay_manager_->GetBaseMinimumDelay(); } bool PeakFound() const override { return false; } int GetFilteredBufferLevel() const override; // Accessors and mutators. void set_sample_memory(int32_t value) override { sample_memory_ = value; } size_t noise_fast_forward() const override { return noise_fast_forward_; } size_t packet_length_samples() const override { return packet_length_samples_; } void set_packet_length_samples(size_t value) override { packet_length_samples_ = value; } void set_prev_time_scale(bool value) override { prev_time_scale_ = value; } private: // The value 5 sets maximum time-stretch rate to about 100 ms/s. static const int kMinTimescaleInterval = 5; // Updates the `buffer_level_filter_` with the current buffer level // `buffer_size_samples`. void FilterBufferLevel(size_t buffer_size_samples); // Returns the operation given that the next available packet is a comfort // noise payload (RFC 3389 only, not codec-internal). virtual NetEq::Operation CngOperation(NetEqController::NetEqStatus status); // Returns the operation given that no packets are available (except maybe // a DTMF event, flagged by setting `play_dtmf` true). virtual NetEq::Operation NoPacket(NetEqController::NetEqStatus status); // Returns the operation to do given that the expected packet is available. virtual NetEq::Operation ExpectedPacketAvailable( NetEqController::NetEqStatus status); // Returns the operation to do given that the expected packet is not // available, but a packet further into the future is at hand. virtual NetEq::Operation FuturePacketAvailable( NetEqController::NetEqStatus status); // 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_countdown_ || timescale_countdown_->Finished(); } // Checks if the current (filtered) buffer level is under the target level. bool UnderTargetLevel() const; // Checks if an ongoing concealment should be continued due to low buffer // level, even though the next packet is available. bool PostponeDecode(NetEqController::NetEqStatus status) const; // Checks if the timestamp leap is so long into the future that a reset due // to exceeding the expand limit will be done. bool ReinitAfterExpands(NetEqController::NetEqStatus status) const; // Checks if we still have not done enough expands to cover the distance from // the last decoded packet to the next available packet. bool PacketTooEarly(NetEqController::NetEqStatus status) const; bool MaxWaitForPacket(NetEqController::NetEqStatus status) const; bool ShouldContinueExpand(NetEqController::NetEqStatus status) const; int GetPlayoutDelayMs(NetEqController::NetEqStatus status) const; // Runtime configurable options through field trial // WebRTC-Audio-NetEqDecisionLogicConfig. struct Config { Config(); bool enable_stable_delay_mode = false; bool combine_concealment_decision = false; int deceleration_target_level_offset_ms = 85; int packet_history_size_ms = 2000; absl::optional cng_timeout_ms; }; Config config_; std::unique_ptr delay_manager_; std::unique_ptr buffer_level_filter_; PacketArrivalHistory packet_arrival_history_; const TickTimer* tick_timer_; int sample_rate_khz_; size_t output_size_samples_; size_t noise_fast_forward_ = 0; size_t packet_length_samples_ = 0; int sample_memory_ = 0; bool prev_time_scale_ = false; bool disallow_time_stretching_; std::unique_ptr timescale_countdown_; int time_stretched_cn_samples_ = 0; bool buffer_flush_ = false; }; } // namespace webrtc #endif // MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_