chromium/third_party/webrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.cc

/*
 *  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 "test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.h"

#include <set>
#include <unordered_map>

#include "absl/types/optional.h"
#include "api/units/timestamp.h"
#include "rtc_base/checks.h"
#include "system_wrappers/include/clock.h"
#include "test/pc/e2e/analyzer/video/dvqa/pausable_state.h"

namespace webrtc {
namespace {

template <typename T>
absl::optional<T> MaybeGetValue(const std::unordered_map<size_t, T>& map,
                                size_t key) { … }

}  // namespace

StreamState::StreamState(size_t sender,
                         std::set<size_t> receivers,
                         Timestamp stream_started_time,
                         Clock* clock)
    : … { … }

uint16_t StreamState::PopFront(size_t peer) { … }

void StreamState::AddPeer(size_t peer) { … }

void StreamState::RemovePeer(size_t peer) { … }

PausableState* StreamState::GetPausableState(size_t peer) { … }

void StreamState::SetLastRenderedFrameTime(size_t peer, Timestamp time) { … }

absl::optional<Timestamp> StreamState::last_rendered_frame_time(
    size_t peer) const { … }

size_t StreamState::GetLongestReceiverQueue() const { … }

}  // namespace webrtc