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Use decoder information for resampler sample rate, fix Opus sample rate to 48 kHz everywhere

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This commit is contained in:
Vanilagy
2025-09-10 23:03:20 +02:00
parent f314f04037
commit 6f06631355
12 changed files with 52 additions and 40 deletions
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dev/convert.html
+6 -3
View File
@@ -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
+5 -3
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@@ -14,10 +14,12 @@
source: new Mediabunny.BlobSource(file),
});
const videoTrack = await input.getPrimaryVideoTrack();
const sink = new Mediabunny.VideoSampleSink(videoTrack);
const audioTrack = await input.getPrimaryAudioTrack();
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
Generated
+6 -6
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@@ -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",
"resolved": "https://registry.npmjs.org/mediabunny/-/mediabunny-1.14.2.tgz",
"integrity": "sha512-JCup3rpsJGIw3I7E7bZSmVEANCr/OwfwkH3g/Wq1TcHEJ7ymLLw+jC3+/Nnn/JLXzFnw8/I6fV3xECLnFedfHg==",
"version": "1.14.3",
"resolved": "https://registry.npmjs.org/mediabunny/-/mediabunny-1.14.3.tgz",
"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 -1
View File
@@ -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 -1
View File
@@ -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
+1 -1
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@@ -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
+17 -17
View File
@@ -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) {
return;
if (this.sourceSampleRate === null) {
// 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
+6 -1
View File
@@ -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
+2
View File
@@ -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
+2 -2
View File
@@ -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
+3 -3
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@@ -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
+2 -2
View File
@@ -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,