Source code
Revision control
Copy as Markdown
Other Tools
/*
* 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.
*/
#include "modules/audio_coding/neteq/packet_arrival_history.h"
#include <algorithm>
#include <cstdint>
#include "api/neteq/tick_timer.h"
#include "rtc_base/checks.h"
namespace webrtc {
PacketArrivalHistory::PacketArrivalHistory(const TickTimer* tick_timer,
int window_size_ms)
: tick_timer_(tick_timer), window_size_ms_(window_size_ms) {}
bool PacketArrivalHistory::Insert(uint32_t rtp_timestamp,
int packet_length_samples) {
int64_t arrival_timestamp =
tick_timer_->ticks() * tick_timer_->ms_per_tick() * sample_rate_khz_;
PacketArrival packet(timestamp_unwrapper_.Unwrap(rtp_timestamp),
arrival_timestamp, packet_length_samples);
if (IsObsolete(packet)) {
return false;
}
if (Contains(packet)) {
return false;
}
history_.emplace(packet.rtp_timestamp, packet);
if (packet != history_.rbegin()->second) {
// Packet was reordered.
return true;
}
// Remove old packets.
while (IsObsolete(history_.begin()->second)) {
if (history_.begin()->second == min_packet_arrivals_.front()) {
min_packet_arrivals_.pop_front();
}
if (history_.begin()->second == max_packet_arrivals_.front()) {
max_packet_arrivals_.pop_front();
}
history_.erase(history_.begin());
}
// Ensure ordering constraints.
while (!min_packet_arrivals_.empty() &&
packet <= min_packet_arrivals_.back()) {
min_packet_arrivals_.pop_back();
}
while (!max_packet_arrivals_.empty() &&
packet >= max_packet_arrivals_.back()) {
max_packet_arrivals_.pop_back();
}
min_packet_arrivals_.push_back(packet);
max_packet_arrivals_.push_back(packet);
return true;
}
void PacketArrivalHistory::Reset() {
history_.clear();
min_packet_arrivals_.clear();
max_packet_arrivals_.clear();
timestamp_unwrapper_.Reset();
}
int PacketArrivalHistory::GetDelayMs(uint32_t rtp_timestamp) const {
int64_t unwrapped_rtp_timestamp =
timestamp_unwrapper_.PeekUnwrap(rtp_timestamp);
int64_t current_timestamp =
tick_timer_->ticks() * tick_timer_->ms_per_tick() * sample_rate_khz_;
PacketArrival packet(unwrapped_rtp_timestamp, current_timestamp,
/*duration_ms=*/0);
return GetPacketArrivalDelayMs(packet);
}
int PacketArrivalHistory::GetMaxDelayMs() const {
if (max_packet_arrivals_.empty()) {
return 0;
}
return GetPacketArrivalDelayMs(max_packet_arrivals_.front());
}
bool PacketArrivalHistory::IsNewestRtpTimestamp(uint32_t rtp_timestamp) const {
if (history_.empty()) {
return true;
}
int64_t unwrapped_rtp_timestamp =
timestamp_unwrapper_.PeekUnwrap(rtp_timestamp);
return unwrapped_rtp_timestamp == history_.rbegin()->second.rtp_timestamp;
}
int PacketArrivalHistory::GetPacketArrivalDelayMs(
const PacketArrival& packet_arrival) const {
if (min_packet_arrivals_.empty()) {
return 0;
}
RTC_DCHECK_NE(sample_rate_khz_, 0);
// TODO(jakobi): Timestamps are first converted to millis for bit-exactness.
return std::max<int>(
packet_arrival.arrival_timestamp / sample_rate_khz_ -
min_packet_arrivals_.front().arrival_timestamp / sample_rate_khz_ -
(packet_arrival.rtp_timestamp / sample_rate_khz_ -
min_packet_arrivals_.front().rtp_timestamp / sample_rate_khz_),
0);
}
bool PacketArrivalHistory::IsObsolete(
const PacketArrival& packet_arrival) const {
if (history_.empty()) {
return false;
}
return packet_arrival.rtp_timestamp + window_size_ms_ * sample_rate_khz_ <
history_.rbegin()->second.rtp_timestamp;
}
bool PacketArrivalHistory::Contains(const PacketArrival& packet_arrival) const {
auto it = history_.upper_bound(packet_arrival.rtp_timestamp);
if (it == history_.begin()) {
return false;
}
--it;
return it->second.contains(packet_arrival);
}
} // namespace webrtc