npm - werift - Versions diffs - 0.18.17 → 0.19.1 - Mend

werift 0.18.17 → 0.19.1

This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.

Files changed (400) hide show

package/lib/webrtc/src/nonstandard/recorder/writer/webm.js CHANGED Viewed

@@ -3,8 +3,9 @@ Object.defineProperty(exports, "__esModule", { value: true });
 exports.WebmFactory = void 0;
 const promises_1 = require("fs/promises");
 const rx_mini_1 = require("rx.mini");
-const __1 = require("../../..");
 const _1 = require(".");
+const __1 = require("../../..");
+const extra_1 = require("../../../../../rtp/src/extra");
 const sourcePath = "packages/webrtc/src/nonstandard/recorder/writer/webm.ts";
 class WebmFactory extends _1.MediaWriter {
     constructor() {
@@ -67,13 +68,13 @@ class WebmFactory extends _1.MediaWriter {
                 };
             }
         });
-        const webm = new __1.WebmCallback(inputTracks, {
+        const webm = new extra_1.WebmCallback(inputTracks, {
             duration: this.options.defaultDuration ?? 1000 * 60 * 60 * 24,
         });
-        const lipsync = new __1.LipsyncCallback();
+        const lipsync = new extra_1.LipsyncCallback();
         this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {
-            const rtpSource = new __1.RtpSourceCallback();
-            const rtcpSource = new __1.RtcpSourceCallback();
+            const rtpSource = new extra_1.RtpSourceCallback();
+            const rtcpSource = new extra_1.RtcpSourceCallback();
             track.onReceiveRtp
                 .subscribe((rtp) => {
                 rtpSource.input(rtp.clone());
@@ -84,12 +85,12 @@ class WebmFactory extends _1.MediaWriter {
                 rtcpSource.input(rtcp);
             })
                 .disposer(this.unSubscribers);
-            const ntpTime = new __1.NtpTimeCallback(clockRate);
+            const ntpTime = new extra_1.NtpTimeCallback(clockRate);
             if (track.kind === "video") {
-                const depacketizer = new __1.DepacketizeCallback(codec, {
+                const depacketizer = new extra_1.DepacketizeCallback(codec, {
                     isFinalPacketInSequence: (h) => h.marker,
                 });
-                const jitterBuffer = new __1.JitterBufferCallback(clockRate);
+                const jitterBuffer = new extra_1.JitterBufferCallback(clockRate);
                 rtpSource.pipe(jitterBuffer.input);
                 rtcpSource.pipe(ntpTime.input);
                 jitterBuffer.pipe(ntpTime.input);
@@ -98,7 +99,7 @@ class WebmFactory extends _1.MediaWriter {
                 lipsync.pipeVideo(webm.inputVideo);
             }
             else {
-                const depacketizer = new __1.DepacketizeCallback(codec);
+                const depacketizer = new extra_1.DepacketizeCallback(codec);
                 rtpSource.pipe(ntpTime.input);
                 rtcpSource.pipe(ntpTime.input);
                 ntpTime.pipe(depacketizer.input);
@@ -107,7 +108,7 @@ class WebmFactory extends _1.MediaWriter {
             }
             return rtpSource;
         });
-        webm.pipe((0, __1.saveToFileSystem)(this.path));
+        webm.pipe((0, extra_1.saveToFileSystem)(this.path));
     }
     async stop() {
         await Promise.all(this.rtpSources.map((r) => r.stop()));

package/lib/webrtc/src/nonstandard/recorder/writer/webm.js.map CHANGED Viewed

	@@ -1 +1 @@
1	- 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{ unlink } from \"fs/promises\";\nimport { EventDisposer } from \"rx.mini\";\n\nimport { ~~SupportedCodec~~ } from \"~~../../../../../rtp/src/container/webm/container~~\";\nimport {\n DepacketizeCallback,\n JitterBufferCallback,\n LipsyncCallback,\n ~~MediaStreamTrack,\n~~ NtpTimeCallback,\n RtcpSourceCallback,\n RtpSourceCallback,\n ~~saveToFileSystem~~,\n WebmCallback,\n ~~WeriftError~~,\n} from \"~~../../..\";\nimport { MediaWriter } from~~ \"~~.\"~~;\n\nconst sourcePath = \"packages/webrtc/src/nonstandard/recorder/writer/webm.ts\";\n\nexport class WebmFactory extends MediaWriter {\n rtpSources: RtpSourceCallback[] = [];\n\n unSubscribers = new EventDisposer();\n\n async start(tracks: MediaStreamTrack[]) {\n await unlink(this.path).catch((e) => e);\n\n const inputTracks = tracks.map((track, i) => {\n const trackNumber = i + 1;\n const payloadType = track.codec!.payloadType;\n\n if (track.kind === \"video\") {\n const codec = ((): SupportedCodec => {\n switch (track.codec?.name.toLowerCase() as SupportedVideoCodec) {\n case \"vp8\":\n return \"VP8\";\n case \"vp9\":\n return \"VP9\";\n case \"h264\":\n return \"MPEG4/ISO/AVC\";\n case \"av1x\":\n return \"AV1\";\n default:\n throw new WeriftError({\n message: \"unsupported codec\",\n payload: { track, path: sourcePath },\n });\n }\n })();\n return {\n kind: \"video\" as const,\n codec,\n clockRate: 90000,\n trackNumber,\n width: this.options.width,\n height: this.options.height,\n payloadType,\n track,\n };\n } else {\n return {\n kind: \"audio\" as const,\n codec: \"OPUS\" as const,\n clockRate: 48000,\n trackNumber,\n payloadType,\n track,\n };\n }\n });\n\n const webm = new WebmCallback(inputTracks, {\n duration: this.options.defaultDuration ?? 1000 * 60 * 60 * 24,\n });\n const lipsync = new LipsyncCallback();\n\n this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {\n const rtpSource = new RtpSourceCallback();\n const rtcpSource = new RtcpSourceCallback();\n track.onReceiveRtp\n .subscribe((rtp) => {\n rtpSource.input(rtp.clone());\n })\n .disposer(this.unSubscribers);\n track.onReceiveRtcp\n .subscribe((rtcp) => {\n rtcpSource.input(rtcp);\n })\n .disposer(this.unSubscribers);\n const ntpTime = new NtpTimeCallback(clockRate);\n\n if (track.kind === \"video\") {\n const depacketizer = new DepacketizeCallback(codec, {\n isFinalPacketInSequence: (h) => h.marker,\n });\n const jitterBuffer = new JitterBufferCallback(clockRate);\n\n rtpSource.pipe(jitterBuffer.input);\n rtcpSource.pipe(ntpTime.input);\n\n jitterBuffer.pipe(ntpTime.input);\n ntpTime.pipe(depacketizer.input);\n depacketizer.pipe(lipsync.inputVideo);\n lipsync.pipeVideo(webm.inputVideo);\n } else {\n const depacketizer = new DepacketizeCallback(codec);\n\n rtpSource.pipe(ntpTime.input);\n rtcpSource.pipe(ntpTime.input);\n\n ntpTime.pipe(depacketizer.input);\n depacketizer.pipe(lipsync.inputAudio);\n lipsync.pipeAudio(webm.inputAudio);\n }\n\n return rtpSource;\n });\n webm.pipe(saveToFileSystem(this.path));\n }\n\n async stop() {\n await Promise.all(this.rtpSources.map((r) => r.stop()));\n this.unSubscribers.dispose();\n }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = (typeof supportedVideoCodecs)[number];\n"]}
1	+ 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{ unlink } from \"fs/promises\";\nimport { EventDisposer } from \"rx.mini\";\n\nimport { MediaWriter } from \".\";\nimport { MediaStreamTrack, WeriftError } from \"../../..\";\nimport {\n DepacketizeCallback,\n JitterBufferCallback,\n LipsyncCallback,\n NtpTimeCallback,\n RtcpSourceCallback,\n RtpSourceCallback,\n SupportedCodec,\n WebmCallback,\n saveToFileSystem,\n} from \"../../../../../rtp/src/extra\";\n\nconst sourcePath = \"packages/webrtc/src/nonstandard/recorder/writer/webm.ts\";\n\nexport class WebmFactory extends MediaWriter {\n rtpSources: RtpSourceCallback[] = [];\n\n unSubscribers = new EventDisposer();\n\n async start(tracks: MediaStreamTrack[]) {\n await unlink(this.path).catch((e) => e);\n\n const inputTracks = tracks.map((track, i) => {\n const trackNumber = i + 1;\n const payloadType = track.codec!.payloadType;\n\n if (track.kind === \"video\") {\n const codec = ((): SupportedCodec => {\n switch (track.codec?.name.toLowerCase() as SupportedVideoCodec) {\n case \"vp8\":\n return \"VP8\";\n case \"vp9\":\n return \"VP9\";\n case \"h264\":\n return \"MPEG4/ISO/AVC\";\n case \"av1x\":\n return \"AV1\";\n default:\n throw new WeriftError({\n message: \"unsupported codec\",\n payload: { track, path: sourcePath },\n });\n }\n })();\n return {\n kind: \"video\" as const,\n codec,\n clockRate: 90000,\n trackNumber,\n width: this.options.width,\n height: this.options.height,\n payloadType,\n track,\n };\n } else {\n return {\n kind: \"audio\" as const,\n codec: \"OPUS\" as const,\n clockRate: 48000,\n trackNumber,\n payloadType,\n track,\n };\n }\n });\n\n const webm = new WebmCallback(inputTracks, {\n duration: this.options.defaultDuration ?? 1000 * 60 * 60 * 24,\n });\n const lipsync = new LipsyncCallback();\n\n this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {\n const rtpSource = new RtpSourceCallback();\n const rtcpSource = new RtcpSourceCallback();\n track.onReceiveRtp\n .subscribe((rtp) => {\n rtpSource.input(rtp.clone());\n })\n .disposer(this.unSubscribers);\n track.onReceiveRtcp\n .subscribe((rtcp) => {\n rtcpSource.input(rtcp);\n })\n .disposer(this.unSubscribers);\n const ntpTime = new NtpTimeCallback(clockRate);\n\n if (track.kind === \"video\") {\n const depacketizer = new DepacketizeCallback(codec, {\n isFinalPacketInSequence: (h) => h.marker,\n });\n const jitterBuffer = new JitterBufferCallback(clockRate);\n\n rtpSource.pipe(jitterBuffer.input);\n rtcpSource.pipe(ntpTime.input);\n\n jitterBuffer.pipe(ntpTime.input);\n ntpTime.pipe(depacketizer.input);\n depacketizer.pipe(lipsync.inputVideo);\n lipsync.pipeVideo(webm.inputVideo);\n } else {\n const depacketizer = new DepacketizeCallback(codec);\n\n rtpSource.pipe(ntpTime.input);\n rtcpSource.pipe(ntpTime.input);\n\n ntpTime.pipe(depacketizer.input);\n depacketizer.pipe(lipsync.inputAudio);\n lipsync.pipeAudio(webm.inputAudio);\n }\n\n return rtpSource;\n });\n webm.pipe(saveToFileSystem(this.path));\n }\n\n async stop() {\n await Promise.all(this.rtpSources.map((r) => r.stop()));\n this.unSubscribers.dispose();\n }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = (typeof supportedVideoCodecs)[number];\n"]}

package/lib/webrtc/src/nonstandard/userMedia.js.map CHANGED Viewed

	@@ -1 +1 @@
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{ ChildProcess, exec } from \"child_process\";\nimport { createSocket } from \"dgram\";\nimport { setImmediate } from \"timers/promises\";\nimport { v4 } from \"uuid\";\n\nimport { randomPort } from \"../../../common/src\";\nimport { RtpPacket } from \"../../../rtp/src\";\nimport { MediaStreamTrack } from \"../media/track\";\n\nexport const getUserMedia = async ({\n path,\n loop,\n width,\n height,\n}: {\n path: string;\n loop?: boolean;\n width?: number;\n height?: number;\n}) => {\n const audioPort = await randomPort();\n const videoPort = await randomPort();\n\n if (path.endsWith(\".mp4\")) {\n return new MediaPlayerMp4({\n audioPort,\n videoPort,\n path,\n loop,\n width,\n height,\n });\n } else {\n return new MediaPlayerWebm({\n audioPort,\n videoPort,\n path,\n loop,\n width,\n height,\n });\n }\n};\n\nabstract class MediaPlayer {\n protected streamId = v4();\n audio = new MediaStreamTrack({ kind: \"audio\", streamId: this.streamId });\n video = new MediaStreamTrack({ kind: \"video\", streamId: this.streamId });\n protected process!: ChildProcess;\n stopped = false;\n\n constructor(\n protected props: {\n videoPort: number;\n audioPort: number;\n path: string;\n loop?: boolean;\n width?: number;\n height?: number;\n },\n ) {\n this.setupTrack(props.audioPort, this.audio);\n this.setupTrack(props.videoPort, this.video);\n }\n\n private setupTrack = (port: number, track: MediaStreamTrack) => {\n let payloadType = 0;\n\n const socket = createSocket(\"udp4\");\n socket.bind(port);\n socket.on(\"message\", async (buf) => {\n const rtp = RtpPacket.deSerialize(buf);\n if (!payloadType) {\n payloadType = rtp.header.payloadType;\n }\n\n // detect gStreamer restarted\n if (payloadType !== rtp.header.payloadType) {\n payloadType = rtp.header.payloadType;\n track.onSourceChanged.execute(rtp.header);\n }\n\n track.writeRtp(buf);\n });\n };\n\n stop() {\n this.stopped = true;\n this.process.kill(\"SIGINT\");\n }\n}\n\nexport class MediaPlayerMp4 extends MediaPlayer {\n async start() {\n let payloadType = 96;\n const run = async () => {\n if (payloadType > 100) payloadType = 96;\n\n let cmd = \"\";\n if (this.props.width && this.props.height) {\n cmd = `gst-launch-1.0 filesrc location= ${this.props.path} ! \\\ndecodebin ! videoscale ! video/x-raw,width=${this.props.width},height=${\n this.props.height\n } ! x264enc ! \\\nh264parse ! rtph264pay config-interval=10 pt=${payloadType++} ! \\\nudpsink host=127.0.0.1 port=${this.props.videoPort}`;\n } else {\n cmd = `gst-launch-1.0 filesrc location= ${this.props.path} ! \\\nqtdemux name=d ! \\\nqueue ! h264parse ! rtph264pay config-interval=10 pt=${payloadType++} ! \\\nudpsink host=127.0.0.1 port=${this.props.videoPort} d. ! \\\nqueue ! aacparse ! avdec_aac ! audioresample ! audioconvert ! opusenc ! rtpopuspay pt=${payloadType++} ! \\\nudpsink host=127.0.0.1 port=${this.props.audioPort}`;\n }\n console.log(cmd);\n this.process = exec(cmd);\n this.process.on(\"error\", (e) => console.error(\"gst error\", e));\n\n if (this.props.loop) {\n await new Promise((r) => this.process.on(\"close\", r));\n if (!this.stopped) {\n run();\n }\n }\n };\n await setImmediate();\n run();\n }\n}\n\nexport class MediaPlayerWebm extends MediaPlayer {\n async start() {\n let payloadType = 96;\n const run = async () => {\n if (payloadType > 100) payloadType = 96;\n\n const cmd = `gst-launch-1.0 filesrc location=${this.props.path} ! matroskademux name=d \\\nd.video_0 ! queue ! rtpvp8pay pt=${payloadType++} ! \\\nudpsink host=127.0.0.1 port=${this.props.videoPort} \\\nd.audio_0 ! queue ! rtpopuspay pt=${payloadType++} ! \\\nudpsink host=127.0.0.1 port=${this.props.audioPort}`;\n this.process = exec(cmd);\n\n if (this.props.loop) {\n await new Promise((r) => this.process.on(\"close\", r));\n if (!this.stopped) {\n run();\n }\n }\n };\n await setImmediate();\n run();\n }\n}\n"]}

package/lib/webrtc/src/peerConnection.d.ts CHANGED Viewed

@@ -1,8 +1,8 @@
 import Event from "rx.mini";
-import { Message } from "../../ice/src/stun/message";
-import { Protocol } from "../../ice/src/types/model";
 import { Address, CandidatePair, InterfaceAddresses } from ".";
 import { DtlsKeys } from ".";
+import { Message } from "../../ice/src/stun/message";
+import { Protocol } from "../../ice/src/types/model";
 import { RTCDataChannel } from "./dataChannel";
 import { EventTarget } from "./helper";
 import { RTCRtpCodecParameters, RTCRtpHeaderExtensionParameters } from "./media/parameters";

package/lib/webrtc/src/peerConnection.js CHANGED Viewed

@@ -372,13 +372,15 @@ class RTCPeerConnection extends helper_1.EventTarget {
         });
     }
     get localDescription() {
-        if (!this._localDescription)
-            return;
+        if (!this._localDescription) {
+            return undefined;
+        }
         return this._localDescription.toJSON();
     }
     get remoteDescription() {
-        if (!this._remoteDescription)
-            return;
+        if (!this._remoteDescription) {
+            return undefined;
+        }
         return this._remoteDescription.toJSON();
     }
     get _localDescription() {
@@ -630,7 +632,7 @@ class RTCPeerConnection extends helper_1.EventTarget {
     }
     createSctpTransport() {
         const dtlsTransport = this.createTransport([
-            const_1.SRTP_PROFILE.SRTP_AEAD_AES_128_GCM,
+            const_1.SRTP_PROFILE.SRTP_AEAD_AES_128_GCM, // prefer
             const_1.SRTP_PROFILE.SRTP_AES128_CM_HMAC_SHA1_80,
         ]);
         const sctp = new sctp_1.RTCSctpTransport();
@@ -852,8 +854,9 @@ class RTCPeerConnection extends helper_1.EventTarget {
     }
     get remoteIsBundled() {
         const remoteSdp = this._remoteDescription;
-        if (!remoteSdp)
-            return;
+        if (!remoteSdp) {
+            return undefined;
+        }
         const bundle = remoteSdp.group.find((g) => g.semantic === "BUNDLE" && this.config.bundlePolicy !== "disable");
         return bundle;
     }
@@ -1128,7 +1131,7 @@ class RTCPeerConnection extends helper_1.EventTarget {
         const kind = typeof trackOrKind === "string" ? trackOrKind : trackOrKind.kind;
         const direction = options.direction || "sendrecv";
         const dtlsTransport = this.createTransport([
-            const_1.SRTP_PROFILE.SRTP_AEAD_AES_128_GCM,
+            const_1.SRTP_PROFILE.SRTP_AEAD_AES_128_GCM, // prefer
             const_1.SRTP_PROFILE.SRTP_AES128_CM_HMAC_SHA1_80,
         ]);
         const sender = new rtpSender_1.RTCRtpSender(trackOrKind);