mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00

The goal is to reduce the amount of time stretching done in response to network jitter. Specifically, we should be able to “ride” over delay spikes if the current delay is sufficient, without decelerating playout. We should also avoid accelerating immediately after a buffer underrun, until we are reasonably sure that the jitter has passed. This is achieved by increasing the deadband where we choose to do normal playout, based on the maximum delay in the short term packet arrival history. The buffer level filter is still used to report the average delay for A/V sync purposes. The new behavior is behind a flag and will be experimented with before it is made default. Bug: webrtc:13322 Change-Id: I5fba0c9d46d835dbe5401669598fa031512ccced Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299500 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39730}
181 lines
7.1 KiB
C++
181 lines
7.1 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;
|
|
|
|
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_
|