| /* | |
| * 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 |