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https://github.com/mollyim/webrtc.git
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01ab7d501b
It replaces the relative arrival delay tracker which is equivalent. This results in a slight bit-exactness change but nothing that should affect quality. Bug: webrtc:13322 Change-Id: I6ed5d6fdfa724859122928a8838acce27ac2e5d0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263380 Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37004}
107 lines
3.5 KiB
C++
107 lines
3.5 KiB
C++
/*
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* Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_coding/neteq/packet_arrival_history.h"
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#include <algorithm>
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#include "api/neteq/tick_timer.h"
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#include "modules/include/module_common_types_public.h"
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namespace webrtc {
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PacketArrivalHistory::PacketArrivalHistory(int window_size_ms)
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: window_size_ms_(window_size_ms) {}
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void PacketArrivalHistory::Insert(uint32_t rtp_timestamp,
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int64_t arrival_time_ms) {
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RTC_DCHECK(sample_rate_khz_ > 0);
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int64_t unwrapped_rtp_timestamp = timestamp_unwrapper_.Unwrap(rtp_timestamp);
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if (!newest_rtp_timestamp_ ||
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unwrapped_rtp_timestamp > *newest_rtp_timestamp_) {
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newest_rtp_timestamp_ = unwrapped_rtp_timestamp;
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}
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history_.emplace_back(unwrapped_rtp_timestamp / sample_rate_khz_,
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arrival_time_ms);
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MaybeUpdateCachedArrivals(history_.back());
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while (history_.front().rtp_timestamp_ms + window_size_ms_ <
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unwrapped_rtp_timestamp / sample_rate_khz_) {
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if (&history_.front() == min_packet_arrival_) {
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min_packet_arrival_ = nullptr;
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}
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if (&history_.front() == max_packet_arrival_) {
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max_packet_arrival_ = nullptr;
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}
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history_.pop_front();
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}
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if (!min_packet_arrival_ || !max_packet_arrival_) {
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for (const PacketArrival& packet : history_) {
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MaybeUpdateCachedArrivals(packet);
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}
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}
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}
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void PacketArrivalHistory::MaybeUpdateCachedArrivals(
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const PacketArrival& packet_arrival) {
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if (!min_packet_arrival_ || packet_arrival <= *min_packet_arrival_) {
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min_packet_arrival_ = &packet_arrival;
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}
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if (!max_packet_arrival_ || packet_arrival >= *max_packet_arrival_) {
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max_packet_arrival_ = &packet_arrival;
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}
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}
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void PacketArrivalHistory::Reset() {
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history_.clear();
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min_packet_arrival_ = nullptr;
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max_packet_arrival_ = nullptr;
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timestamp_unwrapper_ = TimestampUnwrapper();
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newest_rtp_timestamp_ = absl::nullopt;
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}
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int PacketArrivalHistory::GetDelayMs(uint32_t rtp_timestamp,
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int64_t time_ms) const {
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RTC_DCHECK(sample_rate_khz_ > 0);
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int64_t unwrapped_rtp_timestamp_ms =
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timestamp_unwrapper_.UnwrapWithoutUpdate(rtp_timestamp) /
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sample_rate_khz_;
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PacketArrival packet(unwrapped_rtp_timestamp_ms, time_ms);
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return GetPacketArrivalDelayMs(packet);
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}
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int PacketArrivalHistory::GetMaxDelayMs() const {
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if (!max_packet_arrival_) {
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return 0;
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}
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return GetPacketArrivalDelayMs(*max_packet_arrival_);
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}
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bool PacketArrivalHistory::IsNewestRtpTimestamp(uint32_t rtp_timestamp) const {
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if (!newest_rtp_timestamp_) {
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return false;
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}
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int64_t unwrapped_rtp_timestamp =
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timestamp_unwrapper_.UnwrapWithoutUpdate(rtp_timestamp);
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return unwrapped_rtp_timestamp == *newest_rtp_timestamp_;
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}
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int PacketArrivalHistory::GetPacketArrivalDelayMs(
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const PacketArrival& packet_arrival) const {
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if (!min_packet_arrival_) {
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return 0;
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}
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return std::max(static_cast<int>(packet_arrival.arrival_time_ms -
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min_packet_arrival_->arrival_time_ms -
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(packet_arrival.rtp_timestamp_ms -
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min_packet_arrival_->rtp_timestamp_ms)),
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0);
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}
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} // namespace webrtc
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