Implement custom audio resampler for conversion

dev/convert.html

@@ -11,9 +11,275 @@
 	progress.max = 1;
 	document.body.append(progress);
 
+
+
+
+
+	/**
+ * Naive audio resampler using linear interpolation between samples
+ * Much faster but lower quality than windowed sinc - causes aliasing and imaging artifacts
+ * @param {AudioBuffer} inputBuffer - The input audio buffer
+ * @param {number} targetSampleRate - The desired output sample rate
+ * @returns {AudioBuffer} - New resampled audio buffer
+ */
+function naiveResample(inputBuffer, targetSampleRate) {
+    const inputSampleRate = inputBuffer.sampleRate;
+    const ratio = targetSampleRate / inputSampleRate;
+    const inputLength = inputBuffer.length;
+    const outputLength = Math.floor(inputLength * ratio);
+    
+    // Create output buffer
+    const audioContext = new (window.AudioContext || window.webkitAudioContext)();
+    const outputBuffer = audioContext.createBuffer(
+        inputBuffer.numberOfChannels,
+        outputLength,
+        targetSampleRate
+    );
+    
+    // Process each channel independently
+    for (let channel = 0; channel < inputBuffer.numberOfChannels; channel++) {
+        const inputData = inputBuffer.getChannelData(channel);
+        const outputData = outputBuffer.getChannelData(channel);
+        
+        naiveResampleChannel(inputData, outputData, inputSampleRate, targetSampleRate);
+    }
+    
+    return outputBuffer;
+}
+
+/**
+ * Naive resample a single channel using linear interpolation
+ */
+function naiveResampleChannel(inputData, outputData, inputSampleRate, targetSampleRate) {
+    const inputLength = inputData.length;
+    const outputLength = outputData.length;
+    
+    for (let n = 0; n < outputLength; n++) {
+        // Calculate the corresponding position in the input signal
+        const inputPosition = n * inputSampleRate / targetSampleRate;
+        
+        // Get the floor and ceiling indices
+        const lowerIndex = Math.floor(inputPosition);
+        const upperIndex = Math.ceil(inputPosition);
+        
+        // Handle edge cases
+        if (lowerIndex >= inputLength - 1) {
+            // At or past the end - just use the last sample
+            outputData[n] = inputData[inputLength - 1];
+        } else if (lowerIndex < 0) {
+            // Before the start - use first sample (shouldn't happen with our calculation)
+            outputData[n] = inputData[0];
+        } else if (lowerIndex === upperIndex) {
+            // Exact sample alignment - no interpolation needed
+            outputData[n] = inputData[lowerIndex];
+        } else {
+            // Linear interpolation between floor and ceil samples
+            const fraction = inputPosition - lowerIndex;
+            const lowerSample = inputData[lowerIndex];
+            const upperSample = inputData[upperIndex];
+            
+            // Linear interpolation: lerp(a, b, t) = a + t * (b - a)
+            outputData[n] = lowerSample + fraction * (upperSample - lowerSample);
+        }
+    }
+}
+
+
+
+
+
+
+
+/**
+ * Resample an AudioBuffer to a new sample rate using windowed sinc interpolation
+ * @param {AudioBuffer} inputBuffer - The input audio buffer
+ * @param {number} targetSampleRate - The desired output sample rate
+ * @param {number} windowSize - Half-width of the sinc window (default: 6)
+ * @returns {AudioBuffer} - New resampled audio buffer
+ */
+function resampleAudioBuffer(inputBuffer, targetSampleRate, windowSize = 1) {
+    const inputSampleRate = inputBuffer.sampleRate;
+    const ratio = targetSampleRate / inputSampleRate;
+    const inputLength = inputBuffer.length;
+    const outputLength = Math.floor(inputLength * ratio);
+    
+    // Scale window size for anti-aliasing when downsampling
+    const effectiveWindowSize = windowSize * Math.max(1, inputSampleRate / targetSampleRate);
+    
+    // Create output buffer
+    const audioContext = new (window.AudioContext || window.webkitAudioContext)();
+    const outputBuffer = audioContext.createBuffer(
+        inputBuffer.numberOfChannels,
+        outputLength,
+        targetSampleRate
+    );
+    
+    // Process each channel independently
+    for (let channel = 0; channel < inputBuffer.numberOfChannels; channel++) {
+        const inputData = inputBuffer.getChannelData(channel);
+        const outputData = outputBuffer.getChannelData(channel);
+        
+        resampleChannel(inputData, outputData, inputSampleRate, targetSampleRate, effectiveWindowSize);
+    }
+    
+    return outputBuffer;
+}
+
+/**
+ * Resample a single channel of audio data
+ */
+function resampleChannel(inputData, outputData, inputSampleRate, targetSampleRate, windowSize) {
+    const inputLength = inputData.length;
+    const outputLength = outputData.length;
+    
+    for (let n = 0; n < outputLength; n++) {
+        // Current output time in input sample units
+        const inputTime = n * inputSampleRate / targetSampleRate;
+        
+        let sum = 0;
+        const windowRadius = Math.ceil(windowSize);
+        
+        // Convolve with windowed sinc kernel
+        for (let k = -windowRadius; k <= windowRadius; k++) {
+            const inputIndex = Math.floor(inputTime) + k;
+            
+            // Handle edges with zero padding
+            if (inputIndex < 0 || inputIndex >= inputLength) {
+                continue;
+            }
+            
+            // Time difference for sinc calculation
+            const timeDiff = inputTime - inputIndex;
+            
+            // Calculate windowed sinc weight
+            const weight = windowedSinc(timeDiff, windowSize);
+            
+            sum += inputData[inputIndex] * weight;
+        }
+        
+        outputData[n] = sum;
+    }
+}
+
+/**
+ * Windowed sinc function using Kaiser window
+ * @param {number} x - Input value
+ * @param {number} windowSize - Window size parameter
+ * @returns {number} - Windowed sinc value
+ */
+function windowedSinc(x, windowSize) {
+    if (Math.abs(x) > windowSize) {
+        return 0;
+    }
+    
+    // Sinc function
+    let sincValue;
+    if (Math.abs(x) < 1e-10) {
+        sincValue = 1; // lim(x->0) sinc(x) = 1
+    } else {
+        const piX = Math.PI * x;
+        sincValue = Math.sin(piX) / piX;
+    }
+    
+    // Kaiser window (beta = 8 for good balance of main lobe width vs side lobe suppression)
+    const beta = 8;
+    const windowValue = kaiserWindow(x / windowSize, beta);
+    
+    return sincValue * windowValue;
+}
+
+/**
+ * Kaiser window function
+ * @param {number} n - Normalized position (-1 to 1)
+ * @param {number} beta - Kaiser beta parameter
+ * @returns {number} - Window value
+ */
+function kaiserWindow(n, beta) {
+    if (Math.abs(n) > 1) {
+        return 0;
+    }
+    
+    const arg = beta * Math.sqrt(1 - n * n);
+    return modifiedBesselI0(arg) / modifiedBesselI0(beta);
+}
+
+/**
+ * Modified Bessel function of the first kind, order 0
+ * Using series approximation
+ */
+function modifiedBesselI0(x) {
+    let sum = 1;
+    let term = 1;
+    const xSquaredOver4 = (x * x) / 4;
+    
+    for (let k = 1; k < 50; k++) {
+        term *= xSquaredOver4 / (k * k);
+        sum += term;
+        
+        if (term < 1e-12) break; // Convergence check
+    }
+    
+    return sum;
+}
+
+// Example usage:
+// const resampledBuffer = resampleAudioBuffer(originalBuffer, 44100);
+
+// For testing - create a simple test signal
+function createTestBuffer(sampleRate = 48000, duration = 1, frequency = 440) {
+    const audioContext = new (window.AudioContext || window.webkitAudioContext)();
+    const length = Math.floor(sampleRate * duration);
+    const buffer = audioContext.createBuffer(1, length, sampleRate);
+    const data = buffer.getChannelData(0);
+    
+    for (let i = 0; i < length; i++) {
+        data[i] = Math.sin(2 * Math.PI * frequency * i / sampleRate) * 0.5;
+    }
+    
+    return buffer;
+}
+
+// Test example:
+// const testBuffer = createTestBuffer(48000, 1, 440);
+// const resampled = resampleAudioBuffer(testBuffer, 44100);
+// console.log(`Original: ${testBuffer.sampleRate}Hz, ${testBuffer.length} samples`);
+// console.log(`Resampled: ${resampled.sampleRate}Hz, ${resampled.length} samples`);
+
 	fileInput.addEventListener('change', async () => {
 		const file = fileInput.files[0];
 
+		/*
+		const context = new AudioContext();
+		const buffer = await context.decodeAudioData(await file.arrayBuffer());
+
+		const resampled = naiveResample(buffer, 16000);
+		console.log(resampled)
+
+		const node = context.createBufferSource();
+		node.buffer = resampled;
+		node.connect(context.destination);
+		node.start();
+		*/
+
+		/*
+		const cursedOutput = new Metamuxer.Output({
+			format: new Metamuxer.WavOutputFormat(),
+			target: new Metamuxer.BufferTarget()
+		});
+		const cursedSource = new Metamuxer.AudioBufferSource({codec: 'pcm-s16'});
+		cursedOutput.addAudioTrack(cursedSource);
+
+		await cursedOutput.start();
+
+		await cursedSource.add(resampled);
+		await cursedOutput.finalize();
+
+		console.log(cursedOutput.target.buffer);
+		download(new Blob([cursedOutput.target.buffer]), 'cursed.wav')
+		*/
+
+		//return;
+
 		const source = new Metamuxer.BlobSource(file);
 		const target = new Metamuxer.BufferTarget() ?? new Metamuxer.StreamTarget(new WritableStream({
 			write: console.log
@@ -21,7 +287,7 @@
 			chunked: true,
 			chunkSize: 2**20
 		});
-		const outputFormat = new Metamuxer.Mp4OutputFormat();
+		const outputFormat = new Metamuxer.WavOutputFormat();
 
 		const button = document.createElement('button');
 		button.textContent = 'Cancel';
@@ -38,6 +304,8 @@
 				target
 			}),
 			audio: {
+				numberOfChannels: 1,
+				sampleRate: 16000
 				//discard: true
 				//forceReencode: true,
 			},
@@ -66,7 +334,8 @@
 			},
 			*/
 			video: {
-				width: 640
+				discard: true,
+				//width: 640
 				//forceReencode: true,
 				//rotate: 90
 				//width: 720 ?? 2160,

src/conversion.ts

@@ -12,14 +12,12 @@ import {
 import { Input } from './input';
 import { InputAudioTrack, InputTrack, InputVideoTrack } from './input-track';
 import {
-	AudioBufferSink,
 	AudioSampleSink,
 	CanvasSink,
 	EncodedPacketSink,
 	VideoSampleSink,
 } from './media-sink';
 import {
-	AudioBufferSource,
 	AudioEncodingConfig,
 	AudioSource,
 	EncodedVideoPacketSource,
@@ -31,7 +29,7 @@ import {
 } from './media-source';
 import { assert, clamp, normalizeRotation, promiseWithResolvers, Rotation } from './misc';
 import { Output, TrackType } from './output';
-import { VideoSample } from './sample';
+import { AudioSample, VideoSample } from './sample';
 
 /**
  * The options for media file conversion.
@@ -782,18 +780,14 @@ export class Conversion {
 		this.utilizedTracks.push(track);
 	}
 
-	/**
-	 * Resamples the audio by decoding it, playing it onto an OfflineAudioContext and encoding the
-	 * resulting AudioBuffer.
-	 * @internal
-	 */
+	/** @internal */
 	_resampleAudio(
 		track: InputAudioTrack,
 		codec: AudioCodec,
 		targetNumberOfChannels: number,
 		targetSampleRate: number,
 	) {
-		const source = new AudioBufferSource({
+		const source = new AudioSampleSource({
 			codec,
 			bitrate: this._options.audio?.bitrate ?? QUALITY_HIGH,
 			onEncodedPacket: packet => this._reportProgress(track.id, packet.timestamp + packet.duration),
@@ -802,89 +796,33 @@ export class Conversion {
 		this._trackPromises.push((async () => {
 			await this._started;
 
-			const trackDuration = Math.min(
-				await track.computeDuration() - this._startTimestamp,
-				this._endTimestamp - this._startTimestamp,
-			);
-			const totalFrameCount = Math.round(trackDuration * targetSampleRate);
-			const maxChunkLength = 5 * targetSampleRate;
-
-			let currentContextStartFrame = 0;
-			let currentContext: OfflineAudioContext | null = new OfflineAudioContext({
-				length: Math.min(totalFrameCount - currentContextStartFrame, maxChunkLength),
-				numberOfChannels: targetNumberOfChannels,
-				sampleRate: targetSampleRate,
+			const resampler = new AudioResampler({
+				sourceNumberOfChannels: track.numberOfChannels,
+				sourceSampleRate: track.sampleRate,
+				targetNumberOfChannels,
+				targetSampleRate,
+				startTime: this._startTimestamp,
+				endTime: this._endTimestamp,
+				onSample: sample => source.add(sample),
 			});
 
-			const sink = new AudioBufferSink(track);
-			const iterator = sink.buffers(this._startTimestamp, this._endTimestamp);
+			const sink = new AudioSampleSink(track);
+			const iterator = sink.samples(this._startTimestamp, this._endTimestamp); // Todo make sure timestamps work
 
-			for await (const { buffer, timestamp, duration } of iterator) {
-				if (this._synchronizer.shouldWait(track.id, timestamp)) {
-					await this._synchronizer.wait(timestamp);
+			for await (const sample of iterator) {
+				if (this._synchronizer.shouldWait(track.id, sample.timestamp)) {
+					await this._synchronizer.wait(sample.timestamp);
 				}
 
-				const offsetTimestamp = timestamp - this._startTimestamp;
-				const endTimestamp = offsetTimestamp + duration;
-
-				// while loop, as a single source buffer may span multiple audio contexts
-				while (currentContext) {
-					const currentContextStartTime = currentContextStartFrame / targetSampleRate;
-					const currentContextEndTime
-								= (currentContextStartFrame + currentContext.length) / targetSampleRate;
-
-					if (offsetTimestamp < currentContextEndTime) {
-						// The buffer lies within the context, let's play it
-						const node = currentContext.createBufferSource();
-						node.buffer = buffer;
-						node.connect(currentContext.destination);
-
-						if (offsetTimestamp < currentContextStartTime) {
-							node.start(0, currentContextStartTime - offsetTimestamp);
-						} else {
-							node.start(offsetTimestamp - currentContextStartTime);
-						}
-					}
-
-					if (endTimestamp >= currentContextEndTime) {
-						// Render the audio
-						const renderedBuffer = await currentContext.startRendering();
-
-						if (this._canceled) {
-							return;
-						}
-
-						await source.add(renderedBuffer);
-
-						currentContextStartFrame += currentContext.length;
-
-						const newLength = Math.min(
-							totalFrameCount - currentContextStartFrame,
-							maxChunkLength,
-						);
-						currentContext = newLength > 0
-							? new OfflineAudioContext({
-								length: newLength,
-								numberOfChannels: targetNumberOfChannels,
-								sampleRate: targetSampleRate,
-							})
-							: null;
-					} else {
-						break;
-					}
-				}
-			}
-
-			if (currentContext) {
-				const renderedBuffer = await currentContext.startRendering();
-
 				if (this._canceled) {
 					return;
 				}
 
-				await source.add(renderedBuffer);
+				await resampler.add(sample);
 			}
 
+			await resampler.finalize();
+
 			await source.close();
 			this._synchronizer.closeTrack(track.id);
 		})());
@@ -971,3 +909,278 @@ class TrackSynchronizer {
 		this.computeMinAndMaybeResolve();
 	}
 }
+
+/**
+ * Utility class to handle audio resampling, handling both sample rate resampling as well as channel up/downmixing.
+ * The advantage over doing this manually rather than using OfflineAudioContext to do it for us is the artifact-free
+ * handling of putting multiple resampled audio samples back to back, which produces flaky results using
+ * OfflineAudioContext.
+ */
+export class AudioResampler {
+	sourceSampleRate: number;
+	targetSampleRate: number;
+	sourceNumberOfChannels: number;
+	targetNumberOfChannels: number;
+	startTime: number;
+	endTime: number;
+	onSample: (sample: AudioSample) => Promise<void>;
+
+	bufferSizeInFrames: number;
+	bufferSizeInSamples: number;
+	outputBuffer: Float32Array;
+	/** Start frame of current buffer */
+	bufferStartFrame: number;
+	/** The highest index written to in the current buffer */
+	maxWrittenFrame: number;
+	channelMixer!: (sourceData: Float32Array, sourceFrameIndex: number, targetChannelIndex: number) => number;
+	tempSourceBuffer: Float32Array;
+
+	constructor(options: {
+		sourceSampleRate: number;
+		targetSampleRate: number;
+		sourceNumberOfChannels: number;
+		targetNumberOfChannels: number;
+		startTime: number;
+		endTime: number;
+		onSample: (sample: AudioSample) => Promise<void>;
+	}) {
+		this.sourceSampleRate = options.sourceSampleRate;
+		this.targetSampleRate = options.targetSampleRate;
+		this.sourceNumberOfChannels = options.sourceNumberOfChannels;
+		this.targetNumberOfChannels = options.targetNumberOfChannels;
+		this.startTime = options.startTime;
+		this.endTime = options.endTime;
+		this.onSample = options.onSample;
+
+		this.bufferSizeInFrames = Math.floor(this.targetSampleRate * 5.0); // 5 seconds
+		this.bufferSizeInSamples = this.bufferSizeInFrames * this.targetNumberOfChannels;
+
+		this.outputBuffer = new Float32Array(this.bufferSizeInSamples);
+		this.bufferStartFrame = 0;
+		this.maxWrittenFrame = -1;
+
+		this.setupChannelMixer();
+
+		// Pre-allocate temporary buffer for source data
+		this.tempSourceBuffer = new Float32Array(this.sourceSampleRate * this.sourceNumberOfChannels);
+	}
+
+	/**
+	 * Sets up the channel mixer to handle up/downmixing in the case where input and output channel counts don't match.
+	 */
+	setupChannelMixer(): void {
+		const sourceNum = this.sourceNumberOfChannels;
+		const targetNum = this.targetNumberOfChannels;
+
+		// Logic taken from
+		// https://developer.mozilla.org/en-US/docs/Web/API/Web_Audio_API/Basic_concepts_behind_Web_Audio_API
+		// Most of the mapping functions are branchless.
+
+		if (sourceNum === 1 && targetNum === 2) {
+			// Mono to Stereo: M -> L, M -> R
+			this.channelMixer = (sourceData: Float32Array, sourceFrameIndex: number) => {
+				return sourceData[sourceFrameIndex * sourceNum]!;
+			};
+		} else if (sourceNum === 1 && targetNum === 4) {
+			// Mono to Quad: M -> L, M -> R, 0 -> SL, 0 -> SR
+			this.channelMixer = (sourceData: Float32Array, sourceFrameIndex: number, targetChannelIndex: number) => {
+				return sourceData[sourceFrameIndex * sourceNum]! * +(targetChannelIndex < 2);
+			};
+		} else if (sourceNum === 1 && targetNum === 6) {
+			// Mono to 5.1: 0 -> L, 0 -> R, M -> C, 0 -> LFE, 0 -> SL, 0 -> SR
+			this.channelMixer = (sourceData: Float32Array, sourceFrameIndex: number, targetChannelIndex: number) => {
+				return sourceData[sourceFrameIndex * sourceNum]! * +(targetChannelIndex === 2);
+			};
+		} else if (sourceNum === 2 && targetNum === 1) {
+			// Stereo to Mono: 0.5 * (L + R)
+			this.channelMixer = (sourceData: Float32Array, sourceFrameIndex: number) => {
+				const baseIdx = sourceFrameIndex * sourceNum;
+				return 0.5 * (sourceData[baseIdx]! + sourceData[baseIdx + 1]!);
+			};
+		} else if (sourceNum === 2 && targetNum === 4) {
+			// Stereo to Quad: L -> L, R -> R, 0 -> SL, 0 -> SR
+			this.channelMixer = (sourceData: Float32Array, sourceFrameIndex: number, targetChannelIndex: number) => {
+				return sourceData[sourceFrameIndex * sourceNum + targetChannelIndex]! * +(targetChannelIndex < 2);
+			};
+		} else if (sourceNum === 2 && targetNum === 6) {
+			// Stereo to 5.1: L -> L, R -> R, 0 -> C, 0 -> LFE, 0 -> SL, 0 -> SR
+			this.channelMixer = (sourceData: Float32Array, sourceFrameIndex: number, targetChannelIndex: number) => {
+				return sourceData[sourceFrameIndex * sourceNum + targetChannelIndex]! * +(targetChannelIndex < 2);
+			};
+		} else if (sourceNum === 4 && targetNum === 1) {
+			// Quad to Mono: 0.25 * (L + R + SL + SR)
+			this.channelMixer = (sourceData: Float32Array, sourceFrameIndex: number) => {
+				const baseIdx = sourceFrameIndex * sourceNum;
+				return 0.25 * (
+					sourceData[baseIdx]! + sourceData[baseIdx + 1]!
+					+ sourceData[baseIdx + 2]! + sourceData[baseIdx + 3]!
+				);
+			};
+		} else if (sourceNum === 4 && targetNum === 2) {
+			// Quad to Stereo: 0.5 * (L + SL), 0.5 * (R + SR)
+			this.channelMixer = (sourceData: Float32Array, sourceFrameIndex: number, targetChannelIndex: number) => {
+				const baseIdx = sourceFrameIndex * sourceNum;
+				return 0.5 * (
+					sourceData[baseIdx + targetChannelIndex]!
+					+ sourceData[baseIdx + targetChannelIndex + 2]!
+				);
+			};
+		} else if (sourceNum === 4 && targetNum === 6) {
+			// Quad to 5.1: L -> L, R -> R, 0 -> C, 0 -> LFE, SL -> SL, SR -> SR
+			this.channelMixer = (sourceData: Float32Array, sourceFrameIndex: number, targetChannelIndex: number) => {
+				const baseIdx = sourceFrameIndex * sourceNum;
+
+				// It's a bit harder to do this one branchlessly
+				if (targetChannelIndex < 2) return sourceData[baseIdx + targetChannelIndex]!; // L, R
+				if (targetChannelIndex === 2 || targetChannelIndex === 3) return 0; // C, LFE
+				return sourceData[baseIdx + targetChannelIndex - 2]!; // SL, SR
+			};
+		} else if (sourceNum === 6 && targetNum === 1) {
+			// 5.1 to Mono: sqrt(1/2) * (L + R) + C + 0.5 * (SL + SR)
+			this.channelMixer = (sourceData: Float32Array, sourceFrameIndex: number) => {
+				const baseIdx = sourceFrameIndex * sourceNum;
+				return Math.SQRT1_2 * (sourceData[baseIdx]! + sourceData[baseIdx + 1]!)
+					+ sourceData[baseIdx + 2]!
+					+ 0.5 * (sourceData[baseIdx + 4]! + sourceData[baseIdx + 5]!);
+			};
+		} else if (sourceNum === 6 && targetNum === 2) {
+			// 5.1 to Stereo: L + sqrt(1/2) * (C + SL), R + sqrt(1/2) * (C + SR)
+			this.channelMixer = (sourceData: Float32Array, sourceFrameIndex: number, targetChannelIndex: number) => {
+				const baseIdx = sourceFrameIndex * sourceNum;
+				return sourceData[baseIdx + targetChannelIndex]!
+					+ Math.SQRT1_2 * (sourceData[baseIdx + 2]! + sourceData[baseIdx + targetChannelIndex + 4]!);
+			};
+		} else if (sourceNum === 6 && targetNum === 4) {
+			// 5.1 to Quad: L + sqrt(1/2) * C, R + sqrt(1/2) * C, SL, SR
+			this.channelMixer = (sourceData: Float32Array, sourceFrameIndex: number, targetChannelIndex: number) => {
+				const baseIdx = sourceFrameIndex * sourceNum;
+
+				// It's a bit harder to do this one branchlessly
+				if (targetChannelIndex < 2) {
+					return sourceData[baseIdx + targetChannelIndex]! + Math.SQRT1_2 * sourceData[baseIdx + 2]!;
+				}
+				return sourceData[baseIdx + targetChannelIndex + 2]!; // SL, SR
+			};
+		} else {
+			// Discrete fallback: direct mapping with zero-fill or drop
+			this.channelMixer = (sourceData: Float32Array, sourceFrameIndex: number, targetChannelIndex: number) => {
+				return targetChannelIndex < sourceNum
+					? sourceData[sourceFrameIndex * sourceNum + targetChannelIndex]!
+					: 0;
+			};
+		}
+	}
+
+	ensureTempBufferSize(requiredSamples: number): void {
+		let length = this.tempSourceBuffer.length;
+
+		while (length < requiredSamples) {
+			length *= 2;
+		}
+
+		if (length !== this.tempSourceBuffer.length) {
+			const newBuffer = new Float32Array(length);
+			newBuffer.set(this.tempSourceBuffer);
+			this.tempSourceBuffer = newBuffer;
+		}
+	}
+
+	async add(audioSample: AudioSample) {
+		if (!audioSample || audioSample._closed) {
+			return;
+		}
+
+		const requiredSamples = audioSample.numberOfFrames * audioSample.numberOfChannels;
+		this.ensureTempBufferSize(requiredSamples);
+
+		// Copy the audio data to the temp buffer
+		const sourceDataSize = audioSample.allocationSize({ planeIndex: 0, format: 'f32' });
+		const sourceView = new Float32Array(this.tempSourceBuffer.buffer, 0, sourceDataSize / 4);
+		audioSample.copyTo(sourceView, { planeIndex: 0, format: 'f32' });
+
+		const inputStartTime = audioSample.timestamp - this.startTime;
+		const inputDuration = audioSample.numberOfFrames / this.sourceSampleRate;
+		const inputEndTime = Math.min(inputStartTime + inputDuration, this.endTime - this.startTime);
+
+		// Compute which output frames are affected by this sample
+		const outputStartFrame = Math.floor(inputStartTime * this.targetSampleRate);
+		const outputEndFrame = Math.ceil(inputEndTime * this.targetSampleRate);
+
+		for (let outputFrame = outputStartFrame; outputFrame < outputEndFrame; outputFrame++) {
+			if (outputFrame < this.bufferStartFrame) {
+				continue; // Skip writes to the past
+			}
+
+			while (outputFrame >= this.bufferStartFrame + this.bufferSizeInFrames) {
+				// The write is after the current buffer, so finalize it
+				await this.finalizeCurrentBuffer();
+				this.bufferStartFrame += this.bufferSizeInFrames;
+			}
+
+			const bufferFrameIndex = outputFrame - this.bufferStartFrame;
+			assert(bufferFrameIndex < this.bufferSizeInFrames);
+
+			const outputTime = outputFrame / this.targetSampleRate;
+			const inputTime = outputTime - inputStartTime;
+			const sourcePosition = inputTime * this.sourceSampleRate;
+
+			const sourceLowerFrame = Math.floor(sourcePosition);
+			const sourceUpperFrame = Math.ceil(sourcePosition);
+			const fraction = sourcePosition - sourceLowerFrame;
+
+			// Process each output channel
+			for (let targetChannel = 0; targetChannel < this.targetNumberOfChannels; targetChannel++) {
+				let lowerSample = 0;
+				let upperSample = 0;
+
+				if (sourceLowerFrame >= 0 && sourceLowerFrame < audioSample.numberOfFrames) {
+					lowerSample = this.channelMixer(sourceView, sourceLowerFrame, targetChannel);
+				}
+
+				if (sourceUpperFrame >= 0 && sourceUpperFrame < audioSample.numberOfFrames) {
+					upperSample = this.channelMixer(sourceView, sourceUpperFrame, targetChannel);
+				}
+
+				// For resampling, we do naive linear interpolation to find the in-between sample. This produces
+				// suboptimal results especially for downsampling (for which a low-pass filter would first need to be
+				// applied), but AudioContext doesn't do this either so, whatever.
+				const outputSample = lowerSample + fraction * (upperSample - lowerSample);
+
+				// Write to output buffer (interleaved)
+				const outputIndex = bufferFrameIndex * this.targetNumberOfChannels + targetChannel;
+				this.outputBuffer[outputIndex]! += outputSample; // Add in case of overlapping samples
+			}
+
+			this.maxWrittenFrame = Math.max(this.maxWrittenFrame, bufferFrameIndex);
+		}
+	}
+
+	async finalizeCurrentBuffer() {
+		if (this.maxWrittenFrame < 0) {
+			return; // Nothing to finalize
+		}
+
+		const samplesWritten = (this.maxWrittenFrame + 1) * this.targetNumberOfChannels;
+
+		const outputData = new Float32Array(samplesWritten);
+		outputData.set(this.outputBuffer.subarray(0, samplesWritten));
+
+		const timestampSeconds = this.bufferStartFrame / this.targetSampleRate;
+		const audioSample = new AudioSample({
+			format: 'f32',
+			sampleRate: this.targetSampleRate,
+			numberOfChannels: this.targetNumberOfChannels,
+			timestamp: timestampSeconds,
+			data: outputData,
+		});
+
+		await this.onSample(audioSample);
+
+		this.outputBuffer.fill(0);
+		this.maxWrittenFrame = -1;
+	}
+
+	finalize() {
+		return this.finalizeCurrentBuffer();
+	}
+}