Fix AudioResampler emitting timestamps before startTimestamp (fixes #366)

dev/convert.html

@@ -96,7 +96,8 @@
 			input,
 			output,
 			audio: {
-				discard: true,
+				codec: 'aac',
+				forceTranscode: true,
 				//forceTranscode: true,
 				//sampleFormat: 's16',
 			},
@@ -125,7 +126,7 @@
 			},
 			*/
 			video: {
-				//discard: true,
+				discard: true,
 			},
 			tags: {} ?? {
 				title: 'Bigggy',
@@ -145,7 +146,9 @@
 				}
 			},
 			trim: {
-				end: 10,
+				start: 300.14984567374756 - 100,
+				end: 310.1548298151939 - 100,
+				//end: 10,
 				//start: startTime,
 				//end: startTime + 2,
 			},

src/conversion.ts

@@ -1889,6 +1889,7 @@ export class Conversion {
 				startTime: this._startTimestamp,
 				endTime: this._endTimestamp,
 				onSample: async (sample) => {
+					assert(sample.timestamp >= this._startTimestamp);
 					sample.setTimestamp(sample.timestamp - this._startTimestamp);
 
 					await this._registerAudioSample(trackOptions, outputTrackId, source, sample);

src/resample.ts

@@ -20,7 +20,6 @@ export class AudioResampler {
 	targetSampleRate: number;
 	sourceNumberOfChannels: number | null = null;
 	targetNumberOfChannels: number;
-	startTime: number;
 	endTime: number;
 	onSample: (sample: AudioSample) => Promise<void>;
 
@@ -33,6 +32,7 @@ export class AudioResampler {
 	maxWrittenFrame: number | null = null;
 	channelMixer!: (sourceData: Float32Array, sourceFrameIndex: number, targetChannelIndex: number) => number;
 	tempSourceBuffer!: Float32Array;
+	timestampOffset: number;
 
 	constructor(options: {
 		targetSampleRate: number;
@@ -43,7 +43,6 @@ export class AudioResampler {
 	}) {
 		this.targetSampleRate = options.targetSampleRate;
 		this.targetNumberOfChannels = options.targetNumberOfChannels;
-		this.startTime = options.startTime;
 		this.endTime = options.endTime;
 		this.onSample = options.onSample;
 
@@ -51,7 +50,11 @@ export class AudioResampler {
 		this.bufferSizeInSamples = this.bufferSizeInFrames * this.targetNumberOfChannels;
 
 		this.outputBuffer = new Float32Array(this.bufferSizeInSamples);
-		this.bufferStartFrame = Math.floor(this.startTime * this.targetSampleRate);
+
+		this.bufferStartFrame = Math.floor(options.startTime * this.targetSampleRate);
+		// Set to ensure that if the buffer start frame lands on a fractional sample, that the first timestamp still
+		// comes out as exactly startTime
+		this.timestampOffset = options.startTime - this.bufferStartFrame / this.targetSampleRate;
 	}
 
 	/**
@@ -273,7 +276,7 @@ export class AudioResampler {
 			format: 'f32',
 			sampleRate: this.targetSampleRate,
 			numberOfChannels: this.targetNumberOfChannels,
-			timestamp: timestampSeconds,
+			timestamp: timestampSeconds + this.timestampOffset,
 			data: outputData,
 		});
 

test/browser/conversion.test.ts

@@ -1,6 +1,6 @@
 import { ALL_FORMATS } from '../../src/input-format.js';
 import { Input } from '../../src/input.js';
-import { AdtsOutputFormat, HlsOutputFormat, Mp4OutputFormat, MpegTsOutputFormat } from '../../src/output-format.js';
+import { AdtsOutputFormat, HlsOutputFormat, Mp4OutputFormat, MpegTsOutputFormat, WavOutputFormat } from '../../src/output-format.js';
 import { Output, OutputTrackGroup } from '../../src/output.js';
 import { BufferSource, CustomPathedSource, UrlSource } from '../../src/source.js';
 import { expect, test } from 'vitest';
@@ -369,3 +369,30 @@ test('HLS track assignability can be overridden', async () => {
 	expect(newMasterPlayist).not.toBe(masterPlayist);
 	expect(newMasterPlayist.match(/\.m3u8/g)?.length).toBe(1);
 });
+
+test('Fractional audio sample boundary', async () => {
+	using input = new Input({
+		source: new UrlSource('/trim-buck-bunny-ffmpeg.ts'),
+		formats: ALL_FORMATS,
+	});
+
+	const output = new Output({
+		format: new WavOutputFormat(),
+		target: new BufferTarget(),
+	});
+
+	const conversion = await Conversion.init({
+		input,
+		output,
+		video: {
+			discard: true,
+		},
+		audio: {
+			forceTranscode: true,
+		},
+		trim: {
+			start: 0.4 / 48000,
+		},
+	});
+	await conversion.execute();
+});