Merge main into release for tag v1.14.4

dev/convert.html

@@ -24,7 +24,7 @@
 			chunked: true,
 			chunkSize: 2**20
 		});
-		const outputFormat = new Mediabunny.Mp4OutputFormat({});
+		const outputFormat = new Mediabunny.WavOutputFormat({});
 
 		const button = document.createElement('button');
 		button.textContent = 'Cancel';
@@ -72,6 +72,9 @@
 			}),
 			output,
 			audio: {
+				codec: 'pcm-s16',
+				//sampleRate: 16000,
+				//numberOfChannels: 1,
 				//discard: true,
 				//codec: 'opus',
 				//bitrate: 128000,
@@ -106,7 +109,7 @@
 			*/
 			video: () => ({
 				//discard: true,
-				forceTranscode: true,
+				//forceTranscode: true,
 				//codec: 'avc',
 				//fit: 'contain',
 				//frameRate: 27.123,
@@ -129,7 +132,7 @@
 				//height: 100,
 			}),
 			trim: {
-				start: 1,
+				start: 0,
 				end: 10
 			},
 		});

dev/demux.html

@@ -14,10 +14,12 @@
 			source: new Mediabunny.BlobSource(file),
 		});
 
-		const videoTrack = await input.getPrimaryVideoTrack();
+		const audioTrack = await input.getPrimaryAudioTrack();
-		const sink = new Mediabunny.VideoSampleSink(videoTrack);
+		const sink = new Mediabunny.EncodedPacketSink(audioTrack);
 
-		console.log(await sink.getSample(2.131875));
+		for await (const packet of sink.packets()) {
+			console.log(packet);
+		}
 
 		/*
 		const sink = new Mediabunny.EncodedPacketSink(videoTrack);

package-lock.json

@@ -1,12 +1,12 @@
 {
   "name": "mediabunny",
-  "version": "1.14.3",
+  "version": "1.14.4",
   "lockfileVersion": 3,
   "requires": true,
   "packages": {
     "": {
       "name": "mediabunny",
-      "version": "1.14.3",
+      "version": "1.14.4",
       "license": "MPL-2.0",
       "workspaces": [
         "packages/*"
@@ -7749,9 +7749,9 @@
       }
     },
     "node_modules/mediabunny": {
-      "version": "1.14.2",
+      "version": "1.14.3",
-      "resolved": "https://registry.npmjs.org/mediabunny/-/mediabunny-1.14.2.tgz",
+      "resolved": "https://registry.npmjs.org/mediabunny/-/mediabunny-1.14.3.tgz",
-      "integrity": "sha512-JCup3rpsJGIw3I7E7bZSmVEANCr/OwfwkH3g/Wq1TcHEJ7ymLLw+jC3+/Nnn/JLXzFnw8/I6fV3xECLnFedfHg==",
+      "integrity": "sha512-kCvieRo6X1QDcdWLjn7o2BY/VCDeyU9nNGBVjOIOiWPoTtIekHR+viKAYaZafLEm0poBv+O2PwttL37PaSo/kA==",
       "license": "MPL-2.0",
       "peer": true,
       "workspaces": [
@@ -12242,7 +12242,7 @@
     },
     "packages/mp3-encoder": {
       "name": "@mediabunny/mp3-encoder",
-      "version": "1.14.3",
+      "version": "1.14.4",
       "license": "MPL-2.0",
       "devDependencies": {
         "@types/emscripten": "^1.40.1"

package.json

@@ -1,7 +1,7 @@
 {
   "name": "mediabunny",
   "author": "Vanilagy",
-  "version": "1.14.3",
+  "version": "1.14.4",
   "description": "Pure TypeScript media toolkit for reading, writing, and converting media files, directly in the browser.",
   "type": "module",
   "workspaces": [

packages/mp3-encoder/package.json

@@ -1,7 +1,7 @@
 {
   "name": "@mediabunny/mp3-encoder",
   "author": "Vanilagy",
-  "version": "1.14.3",
+  "version": "1.14.4",
   "description": "MP3 encoder extension for Mediabunny, based on LAME.",
   "main": "./dist/bundles/mediabunny-mp3-encoder.mjs",
   "module": "./dist/bundles/mediabunny-mp3-encoder.mjs",

src/codec.ts

@@ -621,7 +621,7 @@ export const parseAacAudioSpecificConfig = (bytes: Uint8Array | null): AacAudioS
 	};
 };
 
-export const OPUS_INTERNAL_SAMPLE_RATE = 48000;
+export const OPUS_SAMPLE_RATE = 48_000;
 
 const PCM_CODEC_REGEX = /^pcm-([usf])(\d+)+(be)?$/;
 

src/conversion.ts

@@ -1125,8 +1125,6 @@ export class Conversion {
 			await this._started;
 
 			const resampler = new AudioResampler({
-				sourceNumberOfChannels: track.numberOfChannels,
-				sourceSampleRate: track.sampleRate,
 				targetNumberOfChannels,
 				targetSampleRate,
 				startTime: this._startTimestamp,
@@ -1246,9 +1244,9 @@ class TrackSynchronizer {
  * OfflineAudioContext.
  */
 export class AudioResampler {
-	sourceSampleRate: number;
+	sourceSampleRate: number | null = null;
 	targetSampleRate: number;
-	sourceNumberOfChannels: number;
+	sourceNumberOfChannels: number | null = null;
 	targetNumberOfChannels: number;
 	startTime: number;
 	endTime: number;
@@ -1262,20 +1260,16 @@ export class AudioResampler {
 	/** The highest index written to in the current buffer */
 	maxWrittenFrame: number;
 	channelMixer!: (sourceData: Float32Array, sourceFrameIndex: number, targetChannelIndex: number) => number;
-	tempSourceBuffer: Float32Array;
+	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;
@@ -1287,17 +1281,14 @@ export class AudioResampler {
 		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 {
+	doChannelMixerSetup(): void {
+		assert(this.sourceNumberOfChannels !== null);
+
 		const sourceNum = this.sourceNumberOfChannels;
 		const targetNum = this.targetNumberOfChannels;
 
@@ -1415,8 +1406,17 @@ export class AudioResampler {
 	}
 
 	async add(audioSample: AudioSample) {
-		if (!audioSample || audioSample._closed) {
+		if (this.sourceSampleRate === null) {
-			return;
+			// This is the first sample, so let's init the missing data. Initting the sample rate from the decoded
+			// sample is more reliable than using the file's metadata, because decoders are free to emit any sample rate
+			// they see fit.
+			this.sourceSampleRate = audioSample.sampleRate;
+			this.sourceNumberOfChannels = audioSample.numberOfChannels;
+
+			// Pre-allocate temporary buffer for source data
+			this.tempSourceBuffer = new Float32Array(this.sourceSampleRate * this.sourceNumberOfChannels);
+
+			this.doChannelMixerSetup();
 		}
 
 		const requiredSamples = audioSample.numberOfFrames * audioSample.numberOfChannels;

src/isobmff/isobmff-demuxer.ts

@@ -12,6 +12,7 @@ import {
 	extractAudioCodecString,
 	extractVideoCodecString,
 	MediaCodec,
+	OPUS_SAMPLE_RATE,
 	parseAacAudioSpecificConfig,
 	parsePcmCodec,
 	PCM_AUDIO_CODECS,
@@ -1049,6 +1050,10 @@ export class IsobmffDemuxer extends Demuxer {
 							}
 						}
 
+						if (track.info.codec === 'opus') {
+							sampleRate = OPUS_SAMPLE_RATE; // Always the same
+						}
+
 						track.info.numberOfChannels = channelCount;
 						track.info.sampleRate = sampleRate;
 
@@ -1421,7 +1426,7 @@ export class IsobmffDemuxer extends Demuxer {
 
 				track.info.codecDescription = description;
 				track.info.numberOfChannels = outputChannelCount;
-				track.info.sampleRate = inputSampleRate;
+				// Don't copy the input sample rate, irrelevant, and output sample rate is fixed
 			}; break;
 
 			case 'dfLa': { // Used for FLAC audio

src/matroska/matroska-demuxer.ts

@@ -18,6 +18,7 @@ import {
 	extractAudioCodecString,
 	extractVideoCodecString,
 	MediaCodec,
+	OPUS_SAMPLE_RATE,
 	VideoCodec,
 } from '../codec';
 import { Demuxer } from '../demuxer';
@@ -988,6 +989,7 @@ export class MatroskaDemuxer extends Demuxer {
 						} else if (codecIdWithoutSuffix === CODEC_STRING_MAP.opus) {
 							this.currentTrack.info.codec = 'opus';
 							this.currentTrack.info.codecDescription = this.currentTrack.codecPrivate;
+							this.currentTrack.info.sampleRate = OPUS_SAMPLE_RATE; // Always the same
 						} else if (codecIdWithoutSuffix === CODEC_STRING_MAP.vorbis) {
 							this.currentTrack.info.codec = 'vorbis';
 							this.currentTrack.info.codecDescription = this.currentTrack.codecPrivate;

src/matroska/matroska-muxer.ts

@@ -47,7 +47,7 @@ import {
 	parseSubtitleTimestamp,
 } from '../subtitles';
 import {
-	OPUS_INTERNAL_SAMPLE_RATE,
+	OPUS_SAMPLE_RATE,
 	PCM_AUDIO_CODECS,
 	PcmAudioCodec,
 	SubtitleCodec,
@@ -293,7 +293,7 @@ export class MatroskaMuxer extends Muxer {
 					const header = parseOpusIdentificationHeader(bytes);
 
 					// Use the preSkip value from the header
-					seekPreRollNs = Math.round(1e9 * (header.preSkip / OPUS_INTERNAL_SAMPLE_RATE));
+					seekPreRollNs = Math.round(1e9 * (header.preSkip / OPUS_SAMPLE_RATE));
 				}
 			}
 

src/ogg/ogg-demuxer.ts

@@ -6,7 +6,7 @@
  * file, You can obtain one at https://mozilla.org/MPL/2.0/.
  */
 
-import { OPUS_INTERNAL_SAMPLE_RATE } from '../codec';
+import { OPUS_SAMPLE_RATE } from '../codec';
 import { parseModesFromVorbisSetupPacket, parseOpusIdentificationHeader } from '../codec-data';
 import { Demuxer } from '../demuxer';
 import { Input } from '../input';
@@ -249,7 +249,7 @@ export class OggDemuxer extends Demuxer {
 
 		const header = parseOpusIdentificationHeader(firstPacket.data);
 		bitstream.numberOfChannels = header.outputChannelCount;
-		bitstream.sampleRate = header.inputSampleRate;
+		bitstream.sampleRate = OPUS_SAMPLE_RATE; // Always the same
 
 		bitstream.codecInfo.opusInfo = {
 			preSkip: header.preSkip,
@@ -574,7 +574,7 @@ class OggAudioTrackBacking implements InputAudioTrackBacking {
 	constructor(public bitstream: LogicalBitstream, public demuxer: OggDemuxer) {
 		// Opus always uses a fixed sample rate for its internal calculations, even if the actual rate is different
 		this.internalSampleRate = bitstream.codecInfo.codec === 'opus'
-			? OPUS_INTERNAL_SAMPLE_RATE
+			? OPUS_SAMPLE_RATE
 			: bitstream.sampleRate;
 	}
 

src/ogg/ogg-muxer.ts

@@ -6,7 +6,7 @@
  * file, You can obtain one at https://mozilla.org/MPL/2.0/.
  */
 
-import { OPUS_INTERNAL_SAMPLE_RATE, validateAudioChunkMetadata } from '../codec';
+import { OPUS_SAMPLE_RATE, validateAudioChunkMetadata } from '../codec';
 import { parseModesFromVorbisSetupPacket, parseOpusIdentificationHeader } from '../codec-data';
 import {
 	assert,
@@ -119,7 +119,7 @@ export class OggMuxer extends Muxer {
 			track,
 			serialNumber,
 			internalSampleRate: track.source._codec === 'opus'
-				? OPUS_INTERNAL_SAMPLE_RATE
+				? OPUS_SAMPLE_RATE
 				: meta.decoderConfig.sampleRate,
 			codecInfo: {
 				codec: track.source._codec,