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Use edit lists to model non-zero start timestamps for regular MP4 (fixes #336); fix MP4 PCM logic breaking down with approximate packet timestamps

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Vanilagy
2026-04-30 10:32:42 +02:00
parent 9224fb886c
commit 782d3e134b
4 changed files with 398 additions and 41 deletions
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src/isobmff/isobmff-muxer.ts
+71 -28
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@@ -52,7 +52,7 @@ import {
import { buildIsobmffMimeType } from './isobmff-misc';
import { MAX_BOX_HEADER_SIZE, MIN_BOX_HEADER_SIZE } from './isobmff-reader';
export const GLOBAL_TIMESCALE = 1000;
export const GLOBAL_TIMESCALE = 57600; // LCM of a bunch of common frame rates (24, 25, 30, 60, 144, ...)
const TIMESTAMP_OFFSET = 2_082_844_800; // Seconds between Jan 1 1904 and Jan 1 1970
export type Sample = {
@@ -85,6 +85,7 @@ export type IsobmffTrackData = {
compositionTimeOffsetTable: { sampleCount: number; sampleCompositionTimeOffset: number }[];
lastTimescaleUnits: number | null;
lastSample: Sample | null;
startTimestampOffset: number | null;
finalizedChunks: Chunk[];
currentChunk: Chunk | null;
@@ -120,6 +121,7 @@ export type IsobmffTrackData = {
* Some players expect this for PCM audio.
*/
requiresPcmTransformation: boolean;
expectedNextPcmPacketTimestamp: number | null;
/**
* The "ADTS stripping" involves removing the ADTS header from each AAC packet. SOBMFF stores raw AAC data, not
* ADTS-wrapped data.
@@ -395,7 +397,10 @@ export class IsobmffMuxer extends Muxer {
// The frame rate set by the user may not be an integer. Since timescale is an integer, we'll approximate the
// frame time (inverse of frame rate) with a rational number, then use that approximation's denominator
// as the timescale.
const timescale = computeRationalApproximation(1 / (track.metadata.frameRate ?? 57600), 1e6).denominator;
const timescale = computeRationalApproximation(
1 / (track.metadata.frameRate ?? GLOBAL_TIMESCALE),
1e6,
).denominator;
const displayAspectWidth = decoderConfig.displayAspectWidth;
const displayAspectHeight = decoderConfig.displayAspectHeight;
@@ -425,6 +430,7 @@ export class IsobmffMuxer extends Muxer {
compositionTimeOffsetTable: [],
lastTimescaleUnits: null,
lastSample: null,
startTimestampOffset: null,
finalizedChunks: [],
currentChunk: null,
compactlyCodedChunkTable: [],
@@ -494,6 +500,7 @@ export class IsobmffMuxer extends Muxer {
requiresPcmTransformation:
!this.isFragmented
&& (PCM_AUDIO_CODECS as readonly string[]).includes(track.source._codec),
expectedNextPcmPacketTimestamp: null,
requiresAdtsStripping,
firstPacket: packet,
},
@@ -505,6 +512,7 @@ export class IsobmffMuxer extends Muxer {
compositionTimeOffsetTable: [],
lastTimescaleUnits: null,
lastSample: null,
startTimestampOffset: null,
finalizedChunks: [],
currentChunk: null,
compactlyCodedChunkTable: [],
@@ -547,6 +555,7 @@ export class IsobmffMuxer extends Muxer {
compositionTimeOffsetTable: [],
lastTimescaleUnits: null,
lastSample: null,
startTimestampOffset: null,
finalizedChunks: [],
currentChunk: null,
compactlyCodedChunkTable: [],
@@ -629,41 +638,59 @@ export class IsobmffMuxer extends Muxer {
packetData = packetData.subarray(headerLength);
}
const timestamp = this.validateAndNormalizeTimestamp(
let timestamp = this.validateAndNormalizeTimestamp(
trackData.track,
packet.timestamp,
packet.type === 'key',
);
let duration = packet.duration;
if (trackData.info.requiresPcmTransformation) {
// Packets may have only approximate timestamp/duration information, but for our PCM logic, we need it
// to be precise. So here, we refine the values.
const pcmInfo = parsePcmCodec(
trackData.info.decoderConfig.codec as PcmAudioCodec,
);
const frameSize = pcmInfo.sampleSize * trackData.info.numberOfChannels;
// Compute the precise duration
duration = packetData.byteLength / frameSize / trackData.info.sampleRate;
if (trackData.info.expectedNextPcmPacketTimestamp !== null) {
const diff = timestamp - trackData.info.expectedNextPcmPacketTimestamp;
if (diff < 0.01) {
timestamp = trackData.info.expectedNextPcmPacketTimestamp;
} else {
const paddedDuration = await this.padWithSilence(
trackData,
trackData.info.expectedNextPcmPacketTimestamp,
diff,
);
timestamp = trackData.info.expectedNextPcmPacketTimestamp + paddedDuration;
}
}
trackData.info.expectedNextPcmPacketTimestamp = timestamp + duration;
}
const internalSample = this.createSampleForTrack(
trackData,
packetData,
timestamp,
packet.duration,
duration,
packet.type,
);
if (trackData.info.requiresPcmTransformation) {
await this.maybePadWithSilence(trackData, timestamp);
}
await this.registerSample(trackData, internalSample);
} finally {
release();
}
}
private async maybePadWithSilence(trackData: IsobmffAudioTrackData, untilTimestamp: number) {
// The PCM transformation assumes that all samples are contiguous. This is not something that is enforced, so
// we need to pad the "holes" in between samples (and before the first sample) with additional
// "silence samples".
const lastSample = last(trackData.samples);
const lastEndTimestamp = lastSample
? lastSample.timestamp + lastSample.duration
: 0;
const delta = untilTimestamp - lastEndTimestamp;
const deltaInTimescale = intoTimescale(delta, trackData.timescale);
private async padWithSilence(trackData: IsobmffAudioTrackData, timestamp: number, duration: number) {
const deltaInTimescale = intoTimescale(duration, trackData.timescale);
duration = deltaInTimescale / trackData.timescale;
if (deltaInTimescale > 0) {
const { sampleSize, silentValue } = parsePcmCodec(
@@ -675,12 +702,14 @@ export class IsobmffMuxer extends Muxer {
const paddingSample = this.createSampleForTrack(
trackData,
new Uint8Array(data.buffer),
lastEndTimestamp,
delta,
timestamp,
duration,
'key',
);
await this.registerSample(trackData, paddingSample);
}
return duration;
}
async addSubtitleCue(track: OutputSubtitleTrack, cue: SubtitleCue, meta?: SubtitleMetadata) {
@@ -821,6 +850,11 @@ export class IsobmffMuxer extends Muxer {
}
if (trackData.type === 'audio' && trackData.info.requiresPcmTransformation) {
if (!this.isFragmented) {
// The first timestamp is the lowest
trackData.startTimestampOffset ??= trackData.timestampProcessingQueue[0]!.timestamp;
}
let totalDuration = 0;
// Compute the total duration in the track timescale (which is equal to the amount of PCM audio samples)
@@ -848,6 +882,10 @@ export class IsobmffMuxer extends Muxer {
const sortedTimestamps = trackData.timestampProcessingQueue.map(x => x.timestamp).sort((a, b) => a - b);
if (!this.isFragmented) {
trackData.startTimestampOffset ??= sortedTimestamps[0]!;
}
for (let i = 0; i < trackData.timestampProcessingQueue.length; i++) {
const sample = trackData.timestampProcessingQueue[i]!;
@@ -857,12 +895,6 @@ export class IsobmffMuxer extends Muxer {
// model it.
sample.decodeTimestamp = sortedTimestamps[i]!;
if (!this.isFragmented && trackData.lastTimescaleUnits === null) {
// In non-fragmented files, the first decode timestamp is always zero. If the first presentation
// timestamp isn't zero, we'll simply use the composition time offset to achieve it.
sample.decodeTimestamp = 0;
}
const sampleCompositionTimeOffset
= intoTimescale(sample.timestamp - sample.decodeTimestamp, trackData.timescale);
const durationInTimescale = intoTimescale(sample.duration, trackData.timescale);
@@ -1377,6 +1409,17 @@ export class IsobmffMuxer extends Muxer {
} else {
for (const trackData of this.trackDatas) {
await this.finalizeCurrentChunk(trackData);
// Must hold because we will have processed at least one sample
assert(trackData.startTimestampOffset !== null);
// Shift all of the samples by the start offset. We'll then write out an edit list that will shift them
// back to their proper spot in the composition.
for (let i = 0; i < trackData.samples.length; i++) {
const sample = trackData.samples[i]!;
sample.timestamp -= trackData.startTimestampOffset;
sample.decodeTimestamp -= trackData.startTimestampOffset;
}
}
}