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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 <cstdint>
#include <deque>
#include "absl/types/optional.h"
#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(int window_size_ms);
virtual ~PacketArrivalHistory() = default;
// Insert packet with `rtp_timestamp` and `arrival_time_ms` into the history.
void Insert(uint32_t rtp_timestamp, int64_t arrival_time_ms);
// The delay for `rtp_timestamp` at `time_ms` is calculated as
// `(time_ms - p.arrival_time_ms) - (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, int64_t time_ms) const;
// Get the maximum packet arrival delay observed in the history.
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_ms, int64_t arrival_time_ms)
: rtp_timestamp_ms(rtp_timestamp_ms),
arrival_time_ms(arrival_time_ms) {}
int64_t rtp_timestamp_ms;
int64_t arrival_time_ms;
bool operator<=(const PacketArrival& other) const {
return arrival_time_ms - rtp_timestamp_ms <=
other.arrival_time_ms - other.rtp_timestamp_ms;
}
bool operator>=(const PacketArrival& other) const {
return arrival_time_ms - rtp_timestamp_ms >=
other.arrival_time_ms - other.rtp_timestamp_ms;
}
};
std::deque<PacketArrival> history_;
int GetPacketArrivalDelayMs(const PacketArrival& packet_arrival) const;
// Updates `min_packet_arrival_` and `max_packet_arrival_`.
void MaybeUpdateCachedArrivals(const PacketArrival& packet);
const PacketArrival* min_packet_arrival_ = nullptr;
const PacketArrival* max_packet_arrival_ = nullptr;
const int window_size_ms_;
RtpTimestampUnwrapper timestamp_unwrapper_;
absl::optional<int64_t> newest_rtp_timestamp_;
int sample_rate_khz_ = 0;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_PACKET_ARRIVAL_HISTORY_H_
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