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/*
* Copyright (c) 2022 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_PACKET_ARRIVAL_HISTORY_H_
#define MODULES_AUDIO_CODING_NETEQ_PACKET_ARRIVAL_HISTORY_H_
#include <cstddef>
#include <cstdint>
#include <deque>
#include <map>
#include "api/neteq/tick_timer.h"
#include "rtc_base/numerics/sequence_number_unwrapper.h"
namespace webrtc {
// Stores timing information about previously received packets.
// The history has a fixed window size beyond which old data is automatically
// pruned.
class PacketArrivalHistory {
public:
explicit PacketArrivalHistory(const TickTimer* tick_timer,
int window_size_ms);
virtual ~PacketArrivalHistory() = default;
// Insert packet with `rtp_timestamp` into the history. Returns true if the
// packet was inserted, false if the timestamp is too old or if the timestamp
// already exists.
bool Insert(uint32_t rtp_timestamp, int packet_length_samples);
// The delay for `rtp_timestamp` at time `now` is calculated as
// `(now - p.arrival_timestamp) - (rtp_timestamp - p.rtp_timestamp)` where `p`
// is chosen as the packet arrival in the history that maximizes the delay.
virtual int GetDelayMs(uint32_t rtp_timestamp) const;
// Get the maximum packet arrival delay observed in the history, excluding
// reordered packets.
virtual int GetMaxDelayMs() const;
bool IsNewestRtpTimestamp(uint32_t rtp_timestamp) const;
void Reset();
void set_sample_rate(int sample_rate) {
sample_rate_khz_ = sample_rate / 1000;
}
size_t size() const { return history_.size(); }
private:
struct PacketArrival {
PacketArrival(int64_t rtp_timestamp,
int64_t arrival_timestamp,
int length_samples)
: rtp_timestamp(rtp_timestamp),
arrival_timestamp(arrival_timestamp),
length_samples(length_samples) {}
PacketArrival() = default;
int64_t rtp_timestamp;
int64_t arrival_timestamp;
int length_samples;
bool operator==(const PacketArrival& other) const {
return rtp_timestamp == other.rtp_timestamp &&
arrival_timestamp == other.arrival_timestamp &&
length_samples == other.length_samples;
}
bool operator!=(const PacketArrival& other) const {
return !(*this == other);
}
bool operator<=(const PacketArrival& other) const {
return arrival_timestamp - rtp_timestamp <=
other.arrival_timestamp - other.rtp_timestamp;
}
bool operator>=(const PacketArrival& other) const {
return arrival_timestamp - rtp_timestamp >=
other.arrival_timestamp - other.rtp_timestamp;
}
bool contains(const PacketArrival& other) const {
return rtp_timestamp <= other.rtp_timestamp &&
rtp_timestamp + length_samples >=
other.rtp_timestamp + other.length_samples;
}
};
int GetPacketArrivalDelayMs(const PacketArrival& packet_arrival) const;
// Checks if the packet is older than the window size.
bool IsObsolete(const PacketArrival& packet_arrival) const;
// Check if the packet exists or fully overlaps with a packet in the history.
bool Contains(const PacketArrival& packet_arrival) const;
const TickTimer* tick_timer_;
const int window_size_ms_;
int sample_rate_khz_ = 0;
RtpTimestampUnwrapper timestamp_unwrapper_;
// Packet history ordered by rtp timestamp.
std::map<int64_t, PacketArrival> history_;
// Tracks min/max packet arrivals in `history_` in ascending/descending order.
// Reordered packets are excluded.
std::deque<PacketArrival> min_packet_arrivals_;
std::deque<PacketArrival> max_packet_arrivals_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_PACKET_ARRIVAL_HISTORY_H_
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