This release adds microphone processing controls and local chat controls, expands platform packages, and synchronizes the website and guides. It also fixes native H.264 input/format handling, signaling recovery, fresh STUN discovery, and UDP ownership during overlapping connections. This is **v1.8.0**. Browser/Server v23, Native control v9 and capture v7 remain unchanged, with the existing 1440p ceiling. Additional resolutions, capability negotiation and Magicsock are separate follow-up work. ## Validation - Full local Web check: 79 files / 1,317 tests, TypeScript and production build. - Full Go checks, platform builds and Windows native capture regressions/probe. - Native H.264 Host lifecycle, Browser-to-App fanout and App-exit recovery, and Native H.264 through embedded SFU: all passed with owned process/port/profile cleanup. - Website build, documentation and repository hygiene passed. Existing UI acceptance covers narrow screens, both languages/themes, architecture downloads and all 23 film checks; final packaging and Go race checks run on the merged revision in CI. - Physical macOS/Linux capture, representative public networks and unmatched field reports retain their documented evidence limits. No claim that every blur/disconnection report is resolved. ## Release notes ### 中文 这是一次功能与兼容性更新:麦克风和聊天增加了实用的小选项,App 与 Server 支持更多设备,同时修复了原生分享和连接恢复中的若干问题。现有 1.x 协议保持不变。 #### 新增与改进 - 浏览器麦克风新增「降噪与回声消除」开关,默认开启回声消除、降噪和自动增益;分享音乐或使用虚拟音频输入时,可按需关闭。 - 可调整弹幕字号和透明度,并将当前保留的聊天记录导出为 TXT。当前会话最多保留 1,000 条文字消息,刷新页面后清空;需要保存时请先导出。 - 新增 Intel Mac、Linux ARM64 App,以及 Linux ARM64 Server 和容器镜像。官网下载区可直接选择对应架构,并显示最新版本和发布时间。 - 更新官网演示、README 截图和中英文教程,明确浏览器分享也可选择窗口或屏幕;开始分享前显示房间号占位,保持底栏控件位置稳定。 - 官网新增直达问题排查的入口,补充浏览器 WebRTC 限制、校园网和 Windows 显卡相关的排查步骤。 - App 模式选择页显示免费开源说明、作者与贡献者署名,以及官方网站和源码链接。 #### 修复 - 修正部分 H.264 原生分享的格式兼容性校验,以及 Windows 编码器输入长度和输出缓冲区对齐问题。 - 清理已失效的重连任务和 Native 接收信令,避免旧连接继续影响当前会话。 - 修复 App 重连时可能复用过期公网地址的问题;重新探测使用当前地址,并处理探测丢包和过期响应,保留其他正常连接。 - 修复 App 新旧媒体连接重叠时共用端口造成的冲突,避免准备或撤销新连接打断仍在使用的画面;关闭连接时及时取消未完成的端口映射请求。 - 无站点密码时,连接恢复不再误引导用户输入站点密码。 - 房间失效或会话终止时,开播前后的原因提示统一显示在电视状态图标中,避免额外提示条。 - 修正 Linux 原生采集的时间戳与关键帧时间基准。感谢 Pedro Antonio 在 [#449](https://github.com/TNTcraftHIM/Piik/pull/449) 的贡献。 #### 使用说明 下载对应平台的完整压缩包,退出旧 App 后解压使用。Windows 继续提供 x64;macOS 提供 Apple 芯片与 Intel 版本,原生采集需要 macOS 13 或更新版本。Linux 提供 x64 与 ARM64,原生采集还需安装[系统组件](https://github.com/TNTcraftHIM/Piik/blob/main/native/capture/linux/README.md)。macOS 和 Linux 的实机覆盖仍有限,欢迎反馈。 ### English This update adds useful microphone and chat controls, brings Piik to more devices, and fixes several native-sharing and connection-recovery issues. The existing 1.x protocols remain unchanged. #### New and improved - Browser microphones now have a “Noise & echo reduction” toggle. Echo cancellation, noise suppression and automatic gain control are on by default; turn them off when appropriate for music or virtual audio inputs. - Adjust chat-overlay text size and opacity, and export retained messages as TXT. The current session keeps up to 1,000 text messages and clears on page refresh; export first if you want to keep them. - Added Intel Mac and Linux ARM64 App packages, plus Linux ARM64 Server and container images. The website provides architecture selectors and shows the latest release and publication date. - Updated the website demonstration, README screenshots and bilingual guides, clarifying that browser capture can also select windows or screens. A room-code placeholder keeps the bottom bar stable before sharing starts. - Added direct website links to troubleshooting, with steps for browser WebRTC restrictions, campus networks and Windows graphics issues. - The App mode-selection page now shows its free and open-source status, author and contributor credit, and links to the official website and source. #### Fixes - Corrected H.264 format compatibility checks and Windows encoder input-length and output-buffer alignment handling. - Retired stale reconnect tasks and Native receive signals so they cannot continue affecting the current session. - Fixed App reconnection reusing an expired public address. Fresh discovery handles lost probes and stale responses while preserving other working connections. - Fixed UDP socket conflicts between overlapping App media connections so preparing or canceling a replacement preserves the active picture. Closing a connection now cancels unfinished gateway mapping requests. - Connection recovery on passwordless sites no longer incorrectly asks for a site passphrase. - Expired-room and session-termination reasons now use the same TV status indicator before and during sharing, without an extra notice strip. - Corrected Linux native capture timestamps and keyframe time bases. Thanks to Pedro Antonio for [#449](https://github.com/TNTcraftHIM/Piik/pull/449). #### Updating Download the complete archive for your platform, quit the old App, and extract the new package. Windows remains x64-only. macOS packages support Apple silicon and Intel; native capture requires macOS 13 or newer. Linux supports x64 and ARM64 and requires additional [system components](https://github.com/TNTcraftHIM/Piik/blob/main/native/capture/linux/README.md) for native capture. Physical-device coverage on macOS and Linux remains limited; feedback is welcome. ## Attribution Co-authored-by: Pedro Antonio <wrenchwastaken@gmail.com>
241 lines
8.9 KiB
Go
241 lines
8.9 KiB
Go
package mediaedge
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import (
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"errors"
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"slices"
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"sync"
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"time"
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"github.com/TNTcraftHIM/Piik/internal/media/forwarding"
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"github.com/pion/webrtc/v4"
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)
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type PublicationMedia struct {
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Codec string `json:"codec"`
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Layers []PublicationLayer `json:"layers"`
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Audio bool `json:"audio"`
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AudioBitrate uint32 `json:"audioBitrate"`
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}
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type PublicationLayer struct {
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RID string `json:"rid"`
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Width uint32 `json:"width"`
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Height uint32 `json:"height"`
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Bitrate uint32 `json:"bitrate"`
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}
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type Publication struct {
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source *Source
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audio *AudioSource
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transport *forwarding.Publication
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// Reuse the existing bounded SDP/candidate owner without registering a P2P edge.
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signaling Edge
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closeOnce sync.Once
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closeErr error
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qualityMu sync.Mutex
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qualityAt time.Time
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qualityCounters forwarding.PublicationCounters
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}
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func (engine *Engine) NewPublication(source *Source, options EdgeOptions) (*Publication, error) {
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if source == nil || source.engine != engine || options.Local || options.ConnectionID == "" ||
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len(options.ConnectionID) > 256 || options.Audio != nil && options.Audio.engine != engine {
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return nil, errors.New("native publication input is invalid")
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}
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// A new publication starts on the base source. Independent lower-output
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// groups are not part of its descriptor and own their failures separately.
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outputsUnavailable := func() bool {
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return slices.Contains(source.outputBitrates[:min(len(source.outputBitrates), len(source.formats))], uint32(0))
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}
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source.mu.Lock()
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unavailable := outputsUnavailable()
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source.mu.Unlock()
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if unavailable {
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return nil, errors.New("native publication output is unavailable")
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}
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if err := source.reserve(false); err != nil {
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return nil, err
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}
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if options.Audio != nil {
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if err := options.Audio.reserve(false); err != nil {
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source.releaseReservation(false)
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return nil, err
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}
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}
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publication := &Publication{source: source, audio: options.Audio}
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socket, err := engine.newICESocket(options.ICEServers, false)
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if err != nil {
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_ = publication.Close()
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return nil, err
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}
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publication.signaling.socket = socket
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var audio webrtc.TrackLocal
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if options.Audio != nil {
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audio = options.Audio.track
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}
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transport, err := forwarding.NewPublication(forwarding.TransportOptions{
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Source: source.media.Source, Settings: socket.settings(), ConnectionID: options.ConnectionID,
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InitialBitrate: engine.initialBitrate, Audio: audio,
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})
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if err != nil {
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_ = publication.Close()
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return nil, err
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}
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publication.transport = transport
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transport.SetAudioBitrate(options.Audio.configuredBitrate())
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publication.signaling.connection = transport.PC
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publication.signaling.localCandidates = newLocalCandidateGathering(socket, options.ICEServers, nil, options.Events.LocalCandidate)
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transport.PC.OnICECandidate(publication.signaling.localCandidates.addPion)
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transport.PC.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
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if state == webrtc.PeerConnectionStateConnected {
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if connectErr := transport.SetConnected(); connectErr != nil {
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_ = transport.PC.Close()
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return
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}
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source.RequestRecoveryFrame()
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}
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if options.Events.ConnectionState != nil {
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options.Events.ConnectionState(state, nil)
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}
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})
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engine.mu.Lock()
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source.mu.Lock()
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if engine.closed || source.closed || len(source.publications) >= 2 || outputsUnavailable() {
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source.mu.Unlock()
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engine.mu.Unlock()
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_ = publication.Close()
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return nil, errors.New("native publication is unavailable")
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}
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source.publications[publication] = struct{}{}
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engine.publications[publication] = struct{}{}
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source.mu.Unlock()
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engine.mu.Unlock()
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return publication, nil
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}
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func (publication *Publication) CreateOffer() (webrtc.SessionDescription, error) {
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return publication.signaling.CreateOffer()
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}
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func (publication *Publication) SetAnswer(answer webrtc.SessionDescription) error {
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return publication.signaling.SetAnswer(answer)
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}
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func (publication *Publication) AddRemoteCandidate(candidate *webrtc.ICECandidateInit) error {
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return publication.signaling.AddRemoteCandidate(candidate)
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}
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func (publication *Publication) SetActiveCount(count int) error {
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if err := publication.transport.SetActiveCount(count); err != nil {
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return err
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}
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publication.source.RequestRecoveryFrame()
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return nil
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}
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func (publication *Publication) RequiredActiveCount() int {
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return publication.transport.RequiredActiveCount()
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}
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func (publication *Publication) LowestLayerBudget() int64 {
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return publication.transport.LowestLayerBudget()
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}
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type PublicationQualitySample struct {
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SampleTimestampMs int64 `json:"sampleTimestampMs"`
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SampleWindowMs int64 `json:"sampleWindowMs"`
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RTPStatsID string `json:"rtpStatsId"`
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TrackIdentifier string `json:"trackIdentifier"`
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Codec string `json:"codec"`
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VideoEncodingCount int `json:"videoEncodingCount"`
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ActiveVideoEncodingCount int `json:"activeVideoEncodingCount"`
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RID string `json:"rid"`
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IntervalFramesSent uint64 `json:"intervalFramesSent"`
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FramesPerSecond float64 `json:"framesPerSecond"`
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Width uint32 `json:"width"`
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Height uint32 `json:"height"`
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BitrateKbps float64 `json:"bitrateKbps"`
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AudioBitrateKbps float64 `json:"audioBitrateKbps"`
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AvailableOutgoingKbps *float64 `json:"availableOutgoingKbps"`
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State string `json:"state"`
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Reason *string `json:"reason"`
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}
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func (publication *Publication) QualitySample(now time.Time) (PublicationQualitySample, bool) {
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current := publication.transport.Counters()
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publication.qualityMu.Lock()
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previous, at := publication.qualityCounters, publication.qualityAt
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publication.qualityCounters, publication.qualityAt = current, now
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publication.qualityMu.Unlock()
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window := now.Sub(at)
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if at.IsZero() || window < time.Second || window > 5*time.Second ||
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current.VideoBytes < previous.VideoBytes || current.AudioBytes < previous.AudioBytes {
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return PublicationQualitySample{}, false
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}
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seconds := window.Seconds()
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source := publication.transport.CurrentSource()
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sample := PublicationQualitySample{SampleTimestampMs: now.UnixMilli(), SampleWindowMs: window.Milliseconds(),
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RTPStatsID: publication.signaling.connection.ID(), TrackIdentifier: string(source.TrackID()),
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Codec: publication.source.codec, VideoEncodingCount: len(current.Frames), State: "unknown",
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BitrateKbps: float64((current.VideoBytes-previous.VideoBytes+current.AudioBytes-previous.AudioBytes)*8) / seconds / 1000,
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AudioBitrateKbps: float64((current.AudioBytes-previous.AudioBytes)*8) / seconds / 1000}
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for index, frames := range current.Frames {
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if index >= len(previous.Frames) || frames <= previous.Frames[index] {
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continue
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}
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sample.ActiveVideoEncodingCount++
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sample.IntervalFramesSent = frames - previous.Frames[index]
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sample.RID = forwarding.PublicationRID(index, len(current.Frames))
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sample.Width, sample.Height = sentVideoDimensions(source, index)
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}
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sample.FramesPerSecond = float64(sample.IntervalFramesSent) / seconds
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if current.Observed {
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capacity := float64(current.TargetBitrate) / 1000
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sample.AvailableOutgoingKbps = &capacity
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if publication.signaling.State() == webrtc.PeerConnectionStateConnected && sample.IntervalFramesSent > 0 {
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reason := "none"
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sample.State = "healthy"
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if current.Limited {
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reason, sample.State = "bandwidth", "degraded"
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}
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sample.Reason = &reason
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}
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}
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return sample, true
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}
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func (publication *Publication) Media() PublicationMedia {
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info := publication.transport.CurrentSource().TrackInfo()
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media := PublicationMedia{Codec: publication.source.codec, Audio: publication.audio != nil,
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AudioBitrate: publication.audio.configuredBitrate(), Layers: make([]PublicationLayer, len(info.Layers))}
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for index, layer := range info.Layers {
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media.Layers[index] = PublicationLayer{RID: forwarding.PublicationRID(index, len(info.Layers)),
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Width: layer.Width, Height: layer.Height, Bitrate: layer.Bitrate}
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}
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publication.transport.SetAudioBitrate(media.AudioBitrate)
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return media
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}
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func (publication *Publication) Close() error {
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publication.closeOnce.Do(func() {
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publication.signaling.mu.Lock()
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publication.signaling.closed = true
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if publication.signaling.localCandidates != nil {
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publication.signaling.localCandidates.close()
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}
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publication.signaling.mu.Unlock()
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if publication.transport != nil {
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publication.closeErr = publication.transport.Close()
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}
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publication.signaling.socket.close()
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publication.source.mu.Lock()
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delete(publication.source.publications, publication)
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publication.source.mu.Unlock()
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publication.source.releaseReservation(false)
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if publication.audio != nil {
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publication.audio.releaseReservation(false)
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}
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engine := publication.source.engine
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engine.mu.Lock()
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delete(engine.publications, publication)
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engine.mu.Unlock()
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publication.source.RequestRecoveryFrame()
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})
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return publication.closeErr
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}
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