Add .gitignore to exclude all node packages and lock files

skills/remotion-prompt-video/node_modules/@remotion/media-utils/dist/get-partial-audio-data.js

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+"use strict";
+Object.defineProperty(exports, "__esModule", { value: true });
+exports.getPartialAudioData = void 0;
+const mediabunny_1 = require("mediabunny");
+// Audio frames might have dependencies on previous and next frames so we need to decode a bit more and then discard it.
+// The worst case seems to be FLAC files with a 65'535 sample window, which would be 1486.0ms at 44.1Khz.
+// So let's set a threshold of 1.5 seconds.
+const EXTRA_THRESHOLD_IN_SECONDS = 1.5;
+const getPartialAudioData = async ({ track, fromSeconds, toSeconds, channelIndex, signal, isMatroska = false, }) => {
+    if (signal.aborted) {
+        throw new Error('Operation was aborted');
+    }
+    const audioSamples = [];
+    // matroska must be decoded from the start due to limitation
+    // https://www.remotion.dev/docs/media/support#matroska-limitation
+    // Also request extra data beforehand to handle audio frame dependencies
+    const actualFromSeconds = isMatroska
+        ? 0
+        : Math.max(0, fromSeconds - EXTRA_THRESHOLD_IN_SECONDS);
+    // mediabunny docs: constructing the sink is virtually free and does not perform any media data reads.
+    const sink = new mediabunny_1.AudioBufferSink(track);
+    const iterator = sink.buffers(actualFromSeconds, toSeconds);
+    for await (const { buffer, timestamp, duration } of iterator) {
+        if (signal.aborted) {
+            break;
+        }
+        const channelData = buffer.getChannelData(channelIndex);
+        const bufferStartSeconds = timestamp;
+        const bufferEndSeconds = timestamp + duration;
+        const overlapStartSecond = Math.max(bufferStartSeconds, fromSeconds);
+        const overlapEndSecond = Math.min(bufferEndSeconds, toSeconds);
+        if (overlapStartSecond >= overlapEndSecond) {
+            continue;
+        }
+        const startSampleInBuffer = Math.floor((overlapStartSecond - bufferStartSeconds) * buffer.sampleRate);
+        const endSampleInBuffer = Math.ceil((overlapEndSecond - bufferStartSeconds) * buffer.sampleRate);
+        const trimmedData = channelData.slice(startSampleInBuffer, endSampleInBuffer);
+        audioSamples.push(trimmedData);
+    }
+    await iterator.return();
+    const totalSamples = audioSamples.reduce((sum, sample) => sum + sample.length, 0);
+    const result = new Float32Array(totalSamples);
+    let offset = 0;
+    for (const audioSample of audioSamples) {
+        result.set(audioSample, offset);
+        offset += audioSample.length;
+    }
+    return result;
+};
+exports.getPartialAudioData = getPartialAudioData;