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This is a reland of2a7c57c34f
Original change's description: > Reland "Refactor NetEq delay manager logic." > > This is a reland off8e62fcb14
> > Original change's description: > > Refactor NetEq delay manager logic. > > > > - Removes dependence on sequence number for calculating target delay. > > - Changes target delay unit to milliseconds instead of number of > > packets. > > - Moves acceleration/preemptive expand thresholds to decision logic. > > Tests for this will be added in a follow up cl. > > > > Bug: webrtc:10333 > > Change-Id: If690aae4abf41ef1d9353f0ff01fb7d121cf8a26 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186265 > > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> > > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#32326} > > Bug: webrtc:10333 > Change-Id: Iad5e7063f63b84762959ee5b412f5f14a7b2cd06 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186943 > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32332} Bug: webrtc:10333 Change-Id: If2244ee9a3d56a0cfa9b602e7bdf448dc6340147 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187356 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32367}
130 lines
4.8 KiB
C++
130 lines
4.8 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_
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#define MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_
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#include <string.h> // Provide access to size_t.
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#include <deque>
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#include <memory>
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#include "absl/types/optional.h"
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#include "api/neteq/tick_timer.h"
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#include "modules/audio_coding/neteq/histogram.h"
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#include "rtc_base/constructor_magic.h"
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namespace webrtc {
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class DelayManager {
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public:
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DelayManager(int max_packets_in_buffer,
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int base_minimum_delay_ms,
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int histogram_quantile,
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const TickTimer* tick_timer,
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std::unique_ptr<Histogram> histogram);
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// Create a DelayManager object. Notify the delay manager that the packet
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// buffer can hold no more than |max_packets_in_buffer| packets (i.e., this
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// is the number of packet slots in the buffer) and that the target delay
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// should be greater than or equal to |base_minimum_delay_ms|. Supply a
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// PeakDetector object to the DelayManager.
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static std::unique_ptr<DelayManager> Create(int max_packets_in_buffer,
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int base_minimum_delay_ms,
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const TickTimer* tick_timer);
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virtual ~DelayManager();
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// Updates the delay manager with a new incoming packet, with |timestamp| from
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// the RTP header. This updates the statistics and a new target buffer level
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// is calculated. Returns the relative delay if it can be calculated. If
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// |reset| is true, restarts the relative arrival delay calculation from this
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// packet.
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virtual absl::optional<int> Update(uint32_t timestamp,
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int sample_rate_hz,
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bool reset = false);
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// Resets all state.
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virtual void Reset();
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// Gets the target buffer level in milliseconds.
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virtual int TargetDelayMs() const;
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// Notifies the DelayManager of how much audio data is carried in each packet.
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virtual int SetPacketAudioLength(int length_ms);
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// Accessors and mutators.
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// Assuming |delay| is in valid range.
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virtual bool SetMinimumDelay(int delay_ms);
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virtual bool SetMaximumDelay(int delay_ms);
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virtual bool SetBaseMinimumDelay(int delay_ms);
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virtual int GetBaseMinimumDelay() const;
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// These accessors are only intended for testing purposes.
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int effective_minimum_delay_ms_for_test() const {
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return effective_minimum_delay_ms_;
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}
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int histogram_quantile() const { return histogram_quantile_; }
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Histogram* histogram() const { return histogram_.get(); }
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private:
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// Provides value which minimum delay can't exceed based on current buffer
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// size and given |maximum_delay_ms_|. Lower bound is a constant 0.
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int MinimumDelayUpperBound() const;
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// Updates |delay_history_|.
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void UpdateDelayHistory(int iat_delay_ms,
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uint32_t timestamp,
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int sample_rate_hz);
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// Calculate relative packet arrival delay from |delay_history_|.
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int CalculateRelativePacketArrivalDelay() const;
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// Updates |effective_minimum_delay_ms_| delay based on current
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// |minimum_delay_ms_|, |base_minimum_delay_ms_| and |maximum_delay_ms_|
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// and buffer size.
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void UpdateEffectiveMinimumDelay();
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// Makes sure that |delay_ms| is less than maximum delay, if any maximum
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// is set. Also, if possible check |delay_ms| to be less than 75% of
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// |max_packets_in_buffer_|.
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bool IsValidMinimumDelay(int delay_ms) const;
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bool IsValidBaseMinimumDelay(int delay_ms) const;
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bool first_packet_received_;
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// TODO(jakobi): set maximum buffer delay instead of number of packets.
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const int max_packets_in_buffer_;
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std::unique_ptr<Histogram> histogram_;
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const int histogram_quantile_;
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const TickTimer* tick_timer_;
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int base_minimum_delay_ms_;
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int effective_minimum_delay_ms_; // Used as lower bound for target delay.
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int minimum_delay_ms_; // Externally set minimum delay.
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int maximum_delay_ms_; // Externally set maximum allowed delay.
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int packet_len_ms_ = 0;
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std::unique_ptr<TickTimer::Stopwatch>
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packet_iat_stopwatch_; // Time elapsed since last packet.
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int target_level_ms_; // Currently preferred buffer level.
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uint32_t last_timestamp_; // Timestamp for the last received packet.
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int num_reordered_packets_ = 0;
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struct PacketDelay {
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int iat_delay_ms;
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uint32_t timestamp;
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};
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std::deque<PacketDelay> delay_history_;
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RTC_DISALLOW_COPY_AND_ASSIGN(DelayManager);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_
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