webrtc/modules/audio_coding/neteq/decision_logic.h
Jakob Ivarsson c782cf883c Introduce a stable playout delay mode for NetEq.
A packet arrival history is used to store the timing of incoming packets and tracks the earliest and latest packets by taking the difference between rtp timestamp and arrival time. The history is windowed to 2 seconds by default. The packet arrival history will replace the relative arrival delay tracker in a follow up cl.

The playout delay is estimated by taking the difference between the current playout timestamp and the earliest packet arrival in the history. This method works better when DTX is used compared to the buffer level filter that it replaces.

The threshold for acceleration is changed to be the maximum of the target delay and the maximum packet arrival delay in the history. This prevents any acceleration immediately after an underrun and gives some time to adapt the target delay to new network conditions.

The logic when to decode the next packet after a packet loss is also changed to do concealment for the full loss duration unless the delay is too high.

The new mode is default disabled and can be enabled using a field trial.

Bug: webrtc:13322,webrtc:13966
Change-Id: Idfa0020584591261475b9ca350cc7c6531de9911
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259820
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36899}
2022-05-16 15:39:14 +00:00

200 lines
7.6 KiB
C++

/*
* 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;
// These methods test the `cng_state_` for different conditions.
bool CngRfc3389On() const override { return cng_state_ == kCngRfc3389On; }
bool CngOff() const override { return cng_state_ == kCngOff; }
// Resets the `cng_state_` to kCngOff.
void SetCngOff() override { cng_state_ = kCngOff; }
void ExpandDecision(NetEq::Operation operation) override {}
// Adds `value` to `sample_memory_`.
void AddSampleMemory(int32_t value) override { sample_memory_ += value; }
int TargetLevelMs() const override;
absl::optional<int> PacketArrived(int fs_hz,
bool should_update_stats,
const PacketArrivedInfo& info) override;
void RegisterEmptyPacket() override {}
void NotifyMutedState() 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;
enum CngState { kCngOff, kCngRfc3389On, kCngInternalOn };
// 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 `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;
bool MaxWaitForPacket() const;
bool ShouldContinueExpand(NetEqController::NetEqStatus status) const;
int GetNextPacketDelayMs(NetEqController::NetEqStatus status) const;
int GetPlayoutDelayMs(NetEqController::NetEqStatus status) const;
int LowThreshold() const;
int HighThreshold() const;
int LowThresholdCng() const;
int HighThresholdCng() const;
// Runtime configurable options through field trial
// WebRTC-Audio-NetEqDecisionLogicConfig.
struct Config {
Config();
bool enable_stable_playout_delay = false;
int reinit_after_expands = 100;
int deceleration_target_level_offset_ms = 85;
int packet_history_size_ms = 2000;
};
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_;
CngState cng_state_ = kCngOff; // Remember if comfort noise is interrupted by
// other event (e.g., DTMF).
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 num_consecutive_expands_ = 0;
int time_stretched_cn_samples_ = 0;
bool last_pack_cng_or_dtmf_ = true;
bool buffer_flush_ = false;
int last_playout_delay_ms_ = 0;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_