npm - werift - Versions diffs - 0.21.13 → 0.21.14 - Mend

werift 0.21.13 → 0.21.14

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 (4) hide show

package/lib/nonstandard/index.mjs CHANGED Viewed

@@ -9364,9 +9364,6 @@ var WebmFactory = class extends MediaWriter {
       duration: this.props.defaultDuration ?? 1e3 * 60 * 60 * 24
     });
     const lipsync = new LipsyncCallback(this.props.lipsync ?? {});
-    if (inputTracks.length === 1 || this.props.disableNtp) {
-      this.props.disableLipSync = true;
-    }
     this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {
       const rtpSource = new RtpSourceCallback();
       const rtcpSource = new RtcpSourceCallback();

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

@@ -87,9 +87,6 @@ class WebmFactory extends _1.MediaWriter {
             duration: this.props.defaultDuration ?? 1000 * 60 * 60 * 24,
         });
         const lipsync = new rtpExtra_1.LipsyncCallback(this.props.lipsync ?? {});
-        if (inputTracks.length === 1 || this.props.disableNtp) {
-            this.props.disableLipSync = true;
-        }
         this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {
             const rtpSource = new rtpExtra_1.RtpSourceCallback();
             const rtcpSource = new rtpExtra_1.RtcpSourceCallback();

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

	@@ -1 +1 @@
1	- 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{ unlink } from \"fs/promises\";\nimport { Event, EventDisposer } from \"../../../imports/common\";\n\nimport { MediaWriter } from \".\";\nimport { type MediaStreamTrack, WeriftError } from \"../../..\";\nimport {\n type ContainerSupportedCodec,\n DepacketizeCallback,\n JitterBufferCallback,\n LipsyncCallback,\n NtpTimeCallback,\n RtcpSourceCallback,\n RtpSourceCallback,\n RtpTimeCallback,\n WebmCallback,\n type WebmTrack,\n saveToFileSystem,\n} from \"../../../imports/rtpExtra\";\n\nconst sourcePath = \"packages/webrtc/src/nonstandard/recorder/writer/webm.ts\";\n\nexport class WebmFactory extends MediaWriter {\n rtpSources: RtpSourceCallback[] = [];\n private onEol = new Event();\n private ended = false;\n unSubscribers = new EventDisposer();\n\n async start(tracks: MediaStreamTrack[]) {\n if (this.props.path) {\n await unlink(this.props.path).catch((e) => e);\n }\n\n const inputTracks: (WebmTrack & { track: MediaStreamTrack })[] = tracks.map(\n (track, i) => {\n const trackNumber = i + 1;\n const payloadType = track.codec!.payloadType;\n\n if (track.kind === \"video\") {\n const codec = ((): ContainerSupportedCodec => {\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.props.width ?? 640,\n height: this.props.height ?? 360,\n roll: this.props.roll,\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\n const webm = new WebmCallback(inputTracks, {\n duration: this.props.defaultDuration ?? 1000 * 60 * 60 * 24,\n });\n const lipsync = new LipsyncCallback(this.props.lipsync ?? {});\n\n if (inputTracks.length === 1 \|\| this.props.disableNtp) {\n this.props.disableLipSync = true;\n }\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 time = this.props.disableNtp\n ? new RtpTimeCallback(clockRate)\n : 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(\n clockRate,\n this.props.jitterBuffer ?? {},\n );\n\n rtpSource.pipe(jitterBuffer.input);\n rtcpSource.pipe(time.input);\n\n jitterBuffer.pipe(time.input);\n time.pipe(depacketizer.input);\n if (this.props.disableLipSync) {\n depacketizer.pipe(webm.inputVideo);\n } else {\n depacketizer.pipe(lipsync.inputVideo);\n lipsync.pipeVideo(webm.inputVideo);\n }\n } else {\n const depacketizer = new DepacketizeCallback(codec);\n\n rtpSource.pipe(time.input);\n rtcpSource.pipe(time.input);\n\n time.pipe(depacketizer.input);\n if (this.props.disableLipSync) {\n depacketizer.pipe(webm.inputAudio);\n } else {\n depacketizer.pipe(lipsync.inputAudio);\n lipsync.pipeAudio(webm.inputAudio);\n }\n }\n\n return rtpSource;\n });\n if (this.props.path) {\n webm.pipe(async (o) => {\n const eol = await saveToFileSystem(this.props.path)(o);\n if (eol) {\n this.onEol.execute();\n this.ended = true;\n }\n });\n } else if (this.props.stream) {\n webm.pipe(async (o) => {\n this.props.stream.execute(o);\n });\n }\n }\n\n async stop() {\n await Promise.all(this.rtpSources.map((r) => r.stop()));\n\n if (!this.ended) {\n await this.onEol.asPromise(5000).catch((e) => e);\n }\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 { Event, EventDisposer } from \"../../../imports/common\";\n\nimport { MediaWriter } from \".\";\nimport { type MediaStreamTrack, WeriftError } from \"../../..\";\nimport {\n type ContainerSupportedCodec,\n DepacketizeCallback,\n JitterBufferCallback,\n LipsyncCallback,\n NtpTimeCallback,\n RtcpSourceCallback,\n RtpSourceCallback,\n RtpTimeCallback,\n WebmCallback,\n type WebmTrack,\n saveToFileSystem,\n} from \"../../../imports/rtpExtra\";\n\nconst sourcePath = \"packages/webrtc/src/nonstandard/recorder/writer/webm.ts\";\n\nexport class WebmFactory extends MediaWriter {\n rtpSources: RtpSourceCallback[] = [];\n private onEol = new Event();\n private ended = false;\n unSubscribers = new EventDisposer();\n\n async start(tracks: MediaStreamTrack[]) {\n if (this.props.path) {\n await unlink(this.props.path).catch((e) => e);\n }\n\n const inputTracks: (WebmTrack & { track: MediaStreamTrack })[] = tracks.map(\n (track, i) => {\n const trackNumber = i + 1;\n const payloadType = track.codec!.payloadType;\n\n if (track.kind === \"video\") {\n const codec = ((): ContainerSupportedCodec => {\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.props.width ?? 640,\n height: this.props.height ?? 360,\n roll: this.props.roll,\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\n const webm = new WebmCallback(inputTracks, {\n duration: this.props.defaultDuration ?? 1000 * 60 * 60 * 24,\n });\n const lipsync = new LipsyncCallback(this.props.lipsync ?? {});\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 time = this.props.disableNtp\n ? new RtpTimeCallback(clockRate)\n : 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(\n clockRate,\n this.props.jitterBuffer ?? {},\n );\n\n rtpSource.pipe(jitterBuffer.input);\n rtcpSource.pipe(time.input);\n\n jitterBuffer.pipe(time.input);\n time.pipe(depacketizer.input);\n if (this.props.disableLipSync) {\n depacketizer.pipe(webm.inputVideo);\n } else {\n depacketizer.pipe(lipsync.inputVideo);\n lipsync.pipeVideo(webm.inputVideo);\n }\n } else {\n const depacketizer = new DepacketizeCallback(codec);\n\n rtpSource.pipe(time.input);\n rtcpSource.pipe(time.input);\n\n time.pipe(depacketizer.input);\n if (this.props.disableLipSync) {\n depacketizer.pipe(webm.inputAudio);\n } else {\n depacketizer.pipe(lipsync.inputAudio);\n lipsync.pipeAudio(webm.inputAudio);\n }\n }\n\n return rtpSource;\n });\n if (this.props.path) {\n webm.pipe(async (o) => {\n const eol = await saveToFileSystem(this.props.path)(o);\n if (eol) {\n this.onEol.execute();\n this.ended = true;\n }\n });\n } else if (this.props.stream) {\n webm.pipe(async (o) => {\n this.props.stream.execute(o);\n });\n }\n }\n\n async stop() {\n await Promise.all(this.rtpSources.map((r) => r.stop()));\n\n if (!this.ended) {\n await this.onEol.asPromise(5000).catch((e) => e);\n }\n this.unSubscribers.dispose();\n }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = (typeof supportedVideoCodecs)[number];\n"]}

package/package.json CHANGED Viewed

@@ -1,6 +1,6 @@
 {
   "name": "werift",
-  "version": "0.21.13",
+  "version": "0.21.14",
   "description": "WebRTC Implementation for TypeScript (Node.js)",
   "keywords": [
     "WebRTC",