| /* |
| * 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 |