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Make MPEG-TS muxing tests not rely on AAC encoder

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This commit is contained in:
Vanilagy
2026-01-16 21:02:12 +01:00
parent 3113e449d4
commit 46fbfc8a2c
1 changed files with 78 additions and 33 deletions
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test/browser/mpeg-ts-muxing.test.ts
+78 -33
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@@ -5,7 +5,7 @@ import { ALL_FORMATS, MPEG_TS } from '../../src/input-format.js';
import { Output } from '../../src/output.js';
import { MpegTsOutputFormat } from '../../src/output-format.js';
import { BufferTarget, StreamTarget, StreamTargetChunk } from '../../src/target.js';
import { AudioBufferSource, CanvasSource, EncodedAudioPacketSource } from '../../src/media-source.js';
import { CanvasSource, EncodedAudioPacketSource } from '../../src/media-source.js';
import { QUALITY_HIGH } from '../../src/encode.js';
import { EncodedPacketSink } from '../../src/media-sink.js';
import { assert } from '../../src/misc.js';
@@ -63,10 +63,7 @@ test('MPEG-TS muxing with AVC and AAC', async () => {
});
output.addVideoTrack(videoSource);
const audioSource = new AudioBufferSource({
codec: 'aac',
bitrate: QUALITY_HIGH,
});
const audioSource = new EncodedAudioPacketSource('aac');
output.addAudioTrack(audioSource);
await output.start();
@@ -81,12 +78,27 @@ test('MPEG-TS muxing with AVC and AAC', async () => {
await videoSource.add(i * frameDuration, frameDuration);
}
const audioBuffer = new AudioBuffer({
length: 48000 * duration,
numberOfChannels: 2,
sampleRate: 48000,
using aacInput = new Input({
source: new UrlSource('/video.mp4'),
formats: ALL_FORMATS,
});
await audioSource.add(audioBuffer);
const aacTrack = await aacInput.getPrimaryAudioTrack();
assert(aacTrack);
const aacSink = new EncodedPacketSink(aacTrack);
let isFirst = true;
for await (const packet of aacSink.packets()) {
if (packet.timestamp >= duration) break;
await audioSource.add(packet, {
decoderConfig: isFirst
? (await aacTrack.getDecoderConfig())!
: undefined,
});
isFirst = false;
}
await output.finalize();
finalized = true;
@@ -188,7 +200,7 @@ test('MPEG-TS muxing with AVC and AAC', async () => {
const audioDuration = await audioTrack.computeDuration();
expect(videoDuration).toBeCloseTo(5, 1);
expect(audioDuration).toBeCloseTo(5.077333333333334, 1);
expect(audioDuration).toBeCloseTo(5, 1);
});
test('MPEG-TS muxing with HEVC and MP3', async () => {
@@ -410,21 +422,34 @@ test('MPEG-TS muxing with audio only', async () => {
target: new BufferTarget(),
});
const audioSource = new AudioBufferSource({
codec: 'aac',
bitrate: QUALITY_HIGH,
});
const audioSource = new EncodedAudioPacketSource('aac');
output.addAudioTrack(audioSource);
await output.start();
const duration = 1;
const audioBuffer = new AudioBuffer({
length: 48000 * duration,
numberOfChannels: 2,
sampleRate: 48000,
using aacInput = new Input({
source: new UrlSource('/video.mp4'),
formats: ALL_FORMATS,
});
await audioSource.add(audioBuffer);
const aacTrack = await aacInput.getPrimaryAudioTrack();
assert(aacTrack);
const aacSink = new EncodedPacketSink(aacTrack);
let isFirst = true;
for await (const packet of aacSink.packets()) {
if (packet.timestamp >= duration) break;
await audioSource.add(packet, {
decoderConfig: isFirst
? (await aacTrack.getDecoderConfig())!
: undefined,
});
isFirst = false;
}
await output.finalize();
@@ -510,8 +535,8 @@ test('MPEG-TS muxing with two audio tracks', async () => {
target: new BufferTarget(),
});
const audioSource1 = new AudioBufferSource({ codec: 'aac', bitrate: QUALITY_HIGH });
const audioSource2 = new AudioBufferSource({ codec: 'aac', bitrate: QUALITY_HIGH });
const audioSource1 = new EncodedAudioPacketSource('aac');
const audioSource2 = new EncodedAudioPacketSource('aac');
output.addAudioTrack(audioSource1);
output.addAudioTrack(audioSource2);
@@ -519,19 +544,39 @@ test('MPEG-TS muxing with two audio tracks', async () => {
await output.start();
const duration = 1;
const audioBuffer1 = new AudioBuffer({
length: 48000 * duration,
numberOfChannels: 2,
sampleRate: 48000,
});
const audioBuffer2 = new AudioBuffer({
length: 44100 * duration,
numberOfChannels: 1,
sampleRate: 44100,
using aacInput = new Input({
source: new UrlSource('/video.mp4'),
formats: ALL_FORMATS,
});
await audioSource1.add(audioBuffer1);
await audioSource2.add(audioBuffer2);
const aacTrack = await aacInput.getPrimaryAudioTrack();
assert(aacTrack);
const aacSink = new EncodedPacketSink(aacTrack);
const decoderConfig = await aacTrack.getDecoderConfig();
assert(decoderConfig);
let isFirst1 = true;
for await (const packet of aacSink.packets()) {
if (packet.timestamp >= duration) break;
await audioSource1.add(packet, {
decoderConfig: isFirst1 ? decoderConfig : undefined,
});
isFirst1 = false;
}
let isFirst2 = true;
for await (const packet of aacSink.packets()) {
if (packet.timestamp >= duration) break;
await audioSource2.add(packet, {
decoderConfig: isFirst2 ? decoderConfig : undefined,
});
isFirst2 = false;
}
await output.finalize();