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/*
* 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 <memory>
#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<DelayManager> delay_manager,
std::unique_ptr<BufferLevelFilter> 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<int> 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<int> cng_timeout_ms;
};
Config config_;
std::unique_ptr<DelayManager> delay_manager_;
std::unique_ptr<BufferLevelFilter> 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<TickTimer::Countdown> timescale_countdown_;
int time_stretched_cn_samples_ = 0;
bool buffer_flush_ = false;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
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