@pexip/peer-connection-stats 17.8.0 → 17.9.0

package/CHANGELOG.md

@@ -1,5 +1,19 @@
 # Changelog
 
+## 17.9.0
+
+### Minor Changes
+
+- 4a1908a: Use tsc to build packages
+
+### Patch Changes
+
+- Updated dependencies [4a1908a]
+- Updated dependencies [ec134d1]
+- Updated dependencies [24fe7e1]
+  - @pexip/signal@16.7.0
+  - @pexip/utils@16.10.0
+
 ## 17.8.0
 
 ### Patch Changes

package/dist/index.d.ts

@@ -1,325 +1,3 @@
-import { Signal } from '@pexip/signal';
-
-interface RTCStats {
-    id: string;
-    type: string;
-    kind?: 'audio' | 'video';
-    timestamp?: number;
-    [key: string]: unknown;
-}
-interface Remote extends RTCStats {
-    type: 'remote-inbound-rtp';
-    jitter?: number;
-    packetsLost?: number;
-    roundTripTime?: number;
-    codec?: Codec;
-    transport?: Transport;
-}
-interface CandidatePair extends RTCStats {
-    type: 'candidate-pair';
-    availableIncomingBitrate?: number;
-    availableOutgoingBitrate?: number;
-    currentRoundTripTime?: number;
-    transport?: Transport;
-    localCandidate?: CandidatePair;
-    remoteCandidate?: CandidatePair;
-}
-interface Transport extends RTCStats {
-    type: 'transport';
-    selectedCandidatePair?: CandidatePair;
-}
-interface Codec extends RTCStats {
-    type: 'codec';
-    mimeType?: string;
-}
-interface InboundAudio extends RTCStats {
-    type: 'inbound-rtp';
-    kind: 'audio';
-    jitter?: number;
-    packetsReceived?: number;
-    packetsLost?: number;
-    bytesReceived?: number;
-    track?: Track;
-    transport?: Transport;
-    codec?: Codec;
-}
-interface OutboundAudio extends RTCStats {
-    type: 'outbound-rtp';
-    kind: 'audio';
-    packetsSent?: number;
-    bytesSent?: number;
-    remote?: Remote;
-    transport?: Transport;
-    codec?: Codec;
-}
-interface InboundVideo extends RTCStats {
-    type: 'inbound-rtp';
-    kind: 'video';
-    packetsReceived?: number;
-    packetsLost?: number;
-    bytesReceived?: number;
-    transport?: Transport;
-    codec?: Codec;
-    frameWidth?: number;
-    frameHeight?: number;
-    framesPerSecond?: number;
-    bitrateMean?: number;
-    framerateMean?: number;
-}
-interface OutboundVideo extends RTCStats {
-    type: 'outbound-rtp';
-    kind: 'video';
-    packetsSent?: number;
-    packetsLost?: number;
-    bytesSent?: number;
-    totalPacketSendDelay?: number;
-    frameWidth?: number;
-    frameHeight?: number;
-    framesPerSecond?: number;
-    codec?: Codec;
-    mediaSource?: MediaSource;
-    remote?: Remote;
-    track?: Track;
-    transport?: Transport;
-    bitrateMean?: number;
-    framerateMean?: number;
-}
-interface MediaSource extends RTCStats {
-    type: 'media-source';
-    kind: 'audio' | 'video';
-}
-interface Track extends RTCStats {
-    type: 'track';
-    kind: 'audio' | 'video';
-    mediaSource?: MediaSource;
-}
-interface Stream extends RTCStats {
-    type: 'stream';
-    tracks?: Track[];
-}
-type AnyStats = CandidatePair | Codec | InboundAudio | InboundVideo | OutboundAudio | OutboundVideo | RTCStats | Remote | Stream | Track | Transport;
-interface Metrics {
-    type: 'inbound-rtp' | 'outbound-rtp';
-    kind: 'audio' | 'video';
-    bitrate?: number;
-    bytesTransmitted?: number;
-    codec?: string;
-    jitter?: number;
-    packetsLost: number;
-    packetsTransmitted: number;
-    recentPercentageLost?: number;
-    roundTripTime?: number;
-    timestamp?: number;
-    totalPercentageLost?: number;
-}
-interface VideoMetrics extends Metrics {
-    framesPerSecond?: number;
-    resolution?: string;
-    resolutionHeight?: number;
-    resolutionWidth?: number;
-}
-type InboundAudioMetrics = Metrics;
-type InboundVideoMetrics = VideoMetrics;
-type OutboundAudioMetrics = Metrics;
-type OutboundVideoMetrics = VideoMetrics & {
-    totalPacketSendDelay?: number;
-    averagePacketSendDelay?: number;
-};
-type NormalizedRTCStats = InboundAudioMetrics | InboundVideoMetrics | OutboundAudioMetrics | OutboundVideoMetrics;
-type PacketsStats = Array<[number, number]>;
-type AudioQualityStats = Array<[number, number]>;
-type VideoQualityStats = number[];
-type CallQualityStats = AudioQualityStats | VideoQualityStats;
-type CallPacketsStats = PacketsStats;
-interface CacheStats {
-    previous?: NormalizedRTCStats;
-    callPacketsStats: CallPacketsStats;
-    callQuality: Quality;
-    callQualityStats: CallQualityStats;
-}
-declare enum Quality {
-    GOOD = 0,
-    OK = 1,
-    BAD = 2,
-    TERRIBLE = 3
-}
-interface StatsSignals {
-    onRtcStats: Signal<NormalizedRTCStats>;
-    onCallQuality: Signal<Quality>;
-    onCallQualityStats: Signal<CallQualityStats>;
-}
-interface StatsCollectorOptions {
-    input: {
-        getStats: (selector?: MediaStreamTrack | null) => Promise<RTCStatsReport>;
-    };
-    signals: StatsSignals;
-    interval?: number;
-}
-interface StatsCollector {
-    resetStats: () => () => void;
-    cleanup: () => void;
-}
-
-declare const STATS_SIZE = 60;
-/**
- * The WebRTC stats are given to us as a flat structure (an array of objects
- * that contain id-fields pointing to other objects in the same array).
- *
- * This class takes such an array as input and can expand an entry so that
- * references become nested objects.
- *
- * https://www.w3.org/TR/webrtc-stats/#summary
- * https://www.w3.org/TR/webrtc-stats/#rtctatstype-*
- */
-declare const createResolver: (statsReport: AnyStats[]) => {
-    expand: {
-        (audioIn: InboundAudio): InboundAudio;
-        (audioOut: OutboundAudio): OutboundAudio;
-        (videoIn: InboundVideo): InboundVideo;
-        (videoOut: OutboundVideo): OutboundVideo;
-    };
-};
-/**
- * Normalize inbound audio stats
- *
- * @param statsData - Raw inbound audio stats @see InboundAudio
- *
- * @returns normalized stats for inbound audio from PeerConnection
- */
-declare const inboundAudio: (statsData: InboundAudio) => InboundAudioMetrics;
-/**
- * Normalize outbound audio stats
- *
- * @param statsData - Raw outbound audio stats @see OutboundAudio
- *
- * @returns normalized stats for outbound audio from PeerConnection
- */
-declare const outboundAudio: (statsData: OutboundAudio) => OutboundAudioMetrics;
-/**
- * Normalize inbound video stats
- *
- * @param statsData - Raw inbound video stats @see InboundVideo
- *
- * @returns normalized stats for inbound video from PeerConnection
- */
-declare const inboundVideo: (statsData: InboundVideo) => InboundVideoMetrics;
-/**
- * Normalize outbound video stats
- *
- * @param statsData - Raw outbound video stats @see OutboundVideo
- *
- * @returns normalized stats for outbound video from PeerConnection
- */
-declare const outboundVideo: (statsData: OutboundVideo) => OutboundVideoMetrics;
-/**
- * Normalize stats into inbound and outbound video and audio stats
- *
- * @param statsReports - reports to for mapping
- *
- * @returns normalized stats for inbound, outbound audio and video
- */
-declare const statsFrom: (statsReports: AnyStats[]) => Metrics | undefined;
-/**
- * Gets raw stats from peerConnection and normalized them
- *
- * https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_Statistics_API
- *
- * A RTCPeerConnection has getStats()
- * - getStats() returns promise which resolve to a RTCStatsReport
- * - RTCStatsReport behaves like an array of RTCStats objects, or more
- *   specific RTCRtpStreamStats objects
- *
- * https://developer.mozilla.org/en-US/docs/Web/API/RTCStatsReport
- * https://developer.mozilla.org/en-US/docs/Web/API/RTCStats
- * https://developer.mozilla.org/en-US/docs/Web/API/RTCRtpStreamStats
- *
- */
-declare const statsFromRTCPeer: (rtcPeer: {
-    getStats: (selector?: MediaStreamTrack | null) => Promise<RTCStatsReport>;
-}) => Promise<NormalizedRTCStats>;
-declare const getQuality: (stats: CallQualityStats) => {
-    quality: Quality;
-    goodOrOkQuality: number;
-    qualityOverTime: Quality[];
-};
-declare const addDeltaStats: (newStats: NormalizedRTCStats, cache: CacheStats) => readonly [{
-    type: "inbound-rtp" | "outbound-rtp";
-    kind: "audio" | "video";
-    bitrate?: number | undefined;
-    bytesTransmitted?: number | undefined;
-    codec?: string | undefined;
-    jitter?: number | undefined;
-    packetsLost: number;
-    packetsTransmitted: number;
-    recentPercentageLost?: number | undefined;
-    roundTripTime?: number | undefined;
-    timestamp?: number | undefined;
-    totalPercentageLost?: number | undefined;
-} | {
-    framesPerSecond?: number | undefined;
-    resolution?: string | undefined;
-    resolutionHeight?: number | undefined;
-    resolutionWidth?: number | undefined;
-    type: "inbound-rtp" | "outbound-rtp";
-    kind: "audio" | "video";
-    bitrate?: number | undefined;
-    bytesTransmitted?: number | undefined;
-    codec?: string | undefined;
-    jitter?: number | undefined;
-    packetsLost: number;
-    packetsTransmitted: number;
-    recentPercentageLost?: number | undefined;
-    roundTripTime?: number | undefined;
-    timestamp?: number | undefined;
-    totalPercentageLost?: number | undefined;
-} | {
-    framesPerSecond?: number | undefined;
-    resolution?: string | undefined;
-    resolutionHeight?: number | undefined;
-    resolutionWidth?: number | undefined;
-    type: "inbound-rtp" | "outbound-rtp";
-    kind: "audio" | "video";
-    bitrate?: number | undefined;
-    bytesTransmitted?: number | undefined;
-    codec?: string | undefined;
-    jitter?: number | undefined;
-    packetsLost: number;
-    packetsTransmitted: number;
-    recentPercentageLost?: number | undefined;
-    roundTripTime?: number | undefined;
-    timestamp?: number | undefined;
-    totalPercentageLost?: number | undefined;
-    totalPacketSendDelay?: number | undefined;
-    averagePacketSendDelay?: number | undefined;
-}, Quality, CallQualityStats];
-declare const calculateQuality: (stats: number | [
-    number,
-    number
-]) => Quality;
-/**
- * Creates stats collector
- *
- * @param input - input to get stats from
- * @param signal - Signal which distributes stats
- * @param interval - how often signal with fire with new stats
- *
- * @returns \{function to reset stats window, cleanup func\}
- */
-declare const createStatsCollector: ({ input, signals: { onCallQuality, onCallQualityStats, onRtcStats }, interval, }: StatsCollectorOptions) => StatsCollector;
-
-/**
- * Create and return all required and optional (if specified with `more`),
- * signals for infinity client to work
- *
- * @param scope - any scope prefix for the generated signal name, @see Signal
- * @param more - Keys from `InfinitySignalsOptional`, @see InfinitySignalsOptional
- *
- * The following signals created by default
- * - 'onConnected',
- * - 'onAnswer',
- *
- * @see REQUIRED_STATS_SIGNAL_KEYS
- */
-declare const createStatsSignals: (scope?: string) => StatsSignals;
-
-export { AnyStats, AudioQualityStats, CacheStats, CallPacketsStats, CallQualityStats, CandidatePair, Codec, InboundAudio, InboundAudioMetrics, InboundVideo, InboundVideoMetrics, MediaSource, Metrics, NormalizedRTCStats, OutboundAudio, OutboundAudioMetrics, OutboundVideo, OutboundVideoMetrics, PacketsStats, Quality, RTCStats, Remote, STATS_SIZE, StatsCollector, StatsCollectorOptions, StatsSignals, Stream, Track, Transport, VideoMetrics, VideoQualityStats, addDeltaStats, calculateQuality, createResolver, createStatsCollector, createStatsSignals, getQuality, inboundAudio, inboundVideo, outboundAudio, outboundVideo, statsFrom, statsFromRTCPeer };
+export * from './statsCollector.types';
+export * from './statsCollector';
+export * from './utils';

package/dist/index.js

@@ -0,0 +1,3 @@
+export * from './statsCollector.types';
+export * from './statsCollector';
+export * from './utils';

package/dist/statsCollector.d.ts

@@ -0,0 +1,145 @@
+import type { AnyStats, CacheStats, CallQualityStats, InboundAudio, InboundAudioMetrics, InboundVideo, InboundVideoMetrics, Metrics, NormalizedRTCStats, OutboundAudio, OutboundAudioMetrics, OutboundVideo, OutboundVideoMetrics, StatsCollector, StatsCollectorOptions } from './statsCollector.types';
+import { Quality } from './statsCollector.types';
+export declare const STATS_SIZE = 60;
+/**
+ * The WebRTC stats are given to us as a flat structure (an array of objects
+ * that contain id-fields pointing to other objects in the same array).
+ *
+ * This class takes such an array as input and can expand an entry so that
+ * references become nested objects.
+ *
+ * https://www.w3.org/TR/webrtc-stats/#summary
+ * https://www.w3.org/TR/webrtc-stats/#rtctatstype-*
+ */
+export declare const createResolver: (statsReport: AnyStats[]) => {
+    expand: {
+        (audioIn: InboundAudio): InboundAudio;
+        (audioOut: OutboundAudio): OutboundAudio;
+        (videoIn: InboundVideo): InboundVideo;
+        (videoOut: OutboundVideo): OutboundVideo;
+    };
+};
+/**
+ * Normalize inbound audio stats
+ *
+ * @param statsData - Raw inbound audio stats @see InboundAudio
+ *
+ * @returns normalized stats for inbound audio from PeerConnection
+ */
+export declare const inboundAudio: (statsData: InboundAudio) => InboundAudioMetrics;
+/**
+ * Normalize outbound audio stats
+ *
+ * @param statsData - Raw outbound audio stats @see OutboundAudio
+ *
+ * @returns normalized stats for outbound audio from PeerConnection
+ */
+export declare const outboundAudio: (statsData: OutboundAudio) => OutboundAudioMetrics;
+/**
+ * Normalize inbound video stats
+ *
+ * @param statsData - Raw inbound video stats @see InboundVideo
+ *
+ * @returns normalized stats for inbound video from PeerConnection
+ */
+export declare const inboundVideo: (statsData: InboundVideo) => InboundVideoMetrics;
+/**
+ * Normalize outbound video stats
+ *
+ * @param statsData - Raw outbound video stats @see OutboundVideo
+ *
+ * @returns normalized stats for outbound video from PeerConnection
+ */
+export declare const outboundVideo: (statsData: OutboundVideo) => OutboundVideoMetrics;
+/**
+ * Normalize stats into inbound and outbound video and audio stats
+ *
+ * @param statsReports - reports to for mapping
+ *
+ * @returns normalized stats for inbound, outbound audio and video
+ */
+export declare const statsFrom: (statsReports: AnyStats[]) => Metrics | undefined;
+/**
+ * Gets raw stats from peerConnection and normalized them
+ *
+ * https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_Statistics_API
+ *
+ * A RTCPeerConnection has getStats()
+ * - getStats() returns promise which resolve to a RTCStatsReport
+ * - RTCStatsReport behaves like an array of RTCStats objects, or more
+ *   specific RTCRtpStreamStats objects
+ *
+ * https://developer.mozilla.org/en-US/docs/Web/API/RTCStatsReport
+ * https://developer.mozilla.org/en-US/docs/Web/API/RTCStats
+ * https://developer.mozilla.org/en-US/docs/Web/API/RTCRtpStreamStats
+ *
+ */
+export declare const statsFromRTCPeer: (rtcPeer: {
+    getStats: (selector?: MediaStreamTrack | null) => Promise<RTCStatsReport>;
+}) => Promise<NormalizedRTCStats>;
+export declare const getQuality: (stats: CallQualityStats) => {
+    quality: Quality;
+    goodOrOkQuality: number;
+    qualityOverTime: Quality[];
+};
+export declare const addDeltaStats: (newStats: NormalizedRTCStats, cache: CacheStats) => readonly [{
+    type: "inbound-rtp" | "outbound-rtp";
+    kind: "audio" | "video";
+    bitrate?: number | undefined;
+    bytesTransmitted?: number | undefined;
+    codec?: string | undefined;
+    jitter?: number | undefined;
+    packetsLost: number;
+    packetsTransmitted: number;
+    recentPercentageLost?: number | undefined;
+    roundTripTime?: number | undefined;
+    timestamp?: number | undefined;
+    totalPercentageLost?: number | undefined;
+} | {
+    framesPerSecond?: number | undefined;
+    resolution?: string | undefined;
+    resolutionHeight?: number | undefined;
+    resolutionWidth?: number | undefined;
+    type: "inbound-rtp" | "outbound-rtp";
+    kind: "audio" | "video";
+    bitrate?: number | undefined;
+    bytesTransmitted?: number | undefined;
+    codec?: string | undefined;
+    jitter?: number | undefined;
+    packetsLost: number;
+    packetsTransmitted: number;
+    recentPercentageLost?: number | undefined;
+    roundTripTime?: number | undefined;
+    timestamp?: number | undefined;
+    totalPercentageLost?: number | undefined;
+} | {
+    framesPerSecond?: number | undefined;
+    resolution?: string | undefined;
+    resolutionHeight?: number | undefined;
+    resolutionWidth?: number | undefined;
+    type: "inbound-rtp" | "outbound-rtp";
+    kind: "audio" | "video";
+    bitrate?: number | undefined;
+    bytesTransmitted?: number | undefined;
+    codec?: string | undefined;
+    jitter?: number | undefined;
+    packetsLost: number;
+    packetsTransmitted: number;
+    recentPercentageLost?: number | undefined;
+    roundTripTime?: number | undefined;
+    timestamp?: number | undefined;
+    totalPercentageLost?: number | undefined;
+    totalPacketSendDelay?: number | undefined;
+    averagePacketSendDelay?: number | undefined;
+}, Quality, CallQualityStats];
+export declare const calculateQuality: (stats: number | [number, number]) => Quality;
+/**
+ * Creates stats collector
+ *
+ * @param input - input to get stats from
+ * @param signal - Signal which distributes stats
+ * @param interval - how often signal with fire with new stats
+ *
+ * @returns \{function to reset stats window, cleanup func\}
+ */
+export declare const createStatsCollector: ({ input, signals: { onCallQuality, onCallQualityStats, onRtcStats }, interval, }: StatsCollectorOptions) => StatsCollector;