mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-16 15:20:42 +01:00
c782cf883c
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}
77 lines
2.7 KiB
C++
77 lines
2.7 KiB
C++
/*
|
|
* 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 <memory>
|
|
|
|
#include "api/neteq/tick_timer.h"
|
|
#include "modules/include/module_common_types_public.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);
|
|
|
|
// 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.
|
|
int GetDelayMs(uint32_t rtp_timestamp, int64_t times_ms) const;
|
|
|
|
// Get the maximum packet arrival delay observed in the history.
|
|
int GetMaxDelayMs() const;
|
|
|
|
void Reset();
|
|
|
|
void set_sample_rate(int sample_rate) {
|
|
sample_rate_khz_ = sample_rate / 1000;
|
|
}
|
|
|
|
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_;
|
|
TimestampUnwrapper timestamp_unwrapper_;
|
|
int sample_rate_khz_ = 0;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_CODING_NETEQ_PACKET_ARRIVAL_HISTORY_H_
|