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Fix typos in source, tests, and docs (#430)

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* Fix typos in source, tests, and docs

* Fix introduction guide typos
This commit is contained in:
Viktor Szépe
2026-07-02 09:33:43 +02:00
committed by GitHub
parent c9a828c272
commit 6812cf1643
10 changed files with 32 additions and 32 deletions
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CLAUDE.md
+1 -1
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@@ -1,6 +1,6 @@
- Prefer functions declared using const, not using the function keyword
- Code style is tab indent with semicolons
- Mediabunny core code is contained in src/, extensions are in packages/*/, website is in docs/
- Tests: Prefer fewer, longer test files over many small ones. Test files should be named after the general catergory of thing that is being tested, not after any individual single test.
- Tests: Prefer fewer, longer test files over many small ones. Test files should be named after the general category of thing that is being tested, not after any individual single test.
- Avoid ifs without a {} block. So no if (cond) return;, always do if (cond) { return; }
- `type` instead of `interface` for object types
docs/guide/introduction.md
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@@ -60,7 +60,7 @@ To get going with Mediabunny, here are some starting points:
## Server-side usage
Mediabunny's simple yet flexible API provides a modern alternative to traditional server-side media processing pipelines which may involve calling out to FFmpeg's CLI manually. Mediabunny was primary built for client-side environments, but when combined with the [`@mediabunny/server`](./extensions/server) extension, the full Mediabunny feature set is available in server-side environments such as Node, Bun, and Deno.
Mediabunny's simple yet flexible API provides a modern alternative to traditional server-side media processing pipelines which may involve calling out to FFmpeg's CLI manually. Mediabunny was primarily built for client-side environments, but when combined with the [`@mediabunny/server`](./extensions/server) extension, the full Mediabunny feature set is available in server-side environments such as Node, Bun, and Deno.
The extension enables:
- Video decoders and encoders for AVC (H.264), HEVC (H.265), VP8, VP9, and AV1. Supports both length-prefixed and Annex B AVC/HEVC as well as transparent video via VP9.
docs/guide/writing-media-files.md
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@@ -342,7 +342,7 @@ Each chunk written to the `WritableStream` represents a contiguous chunk of byte
::: warning
Note that some byte regions in the output file may be written to multiple times. It is therefore **incorrect** to construct the final file by simply concatenating all `Uint8Array`s together - you **must** write each chunk of data at the specified byte offset position _in the order_ in which the chunks arrived. If you don't do this, your output file will likely be invalid or corrupted.
Some [output formats](./output-formats) have *append-only* writing modes in which the byte offset of a written chunk will always be equal to the total number of bytes in all previously written chunks. In other words, when writing is append-only, simply concatening all `Uint8Array`s yields the correct result. Some APIs (like `appendBuffer` of Media Source Extensions) require this, so make sure to configure your output format accordingly for those cases.
Some [output formats](./output-formats) have *append-only* writing modes in which the byte offset of a written chunk will always be equal to the total number of bytes in all previously written chunks. In other words, when writing is append-only, simply concatenating all `Uint8Array`s yields the correct result. Some APIs (like `appendBuffer` of Media Source Extensions) require this, so make sure to configure your output format accordingly for those cases.
:::
#### Chunked mode
scripts/bundle.ts
+5 -5
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@@ -103,7 +103,7 @@ const mp3EncoderVariants = await createVariants(
'packages/mp3-encoder/src/index.ts',
'MediabunnyMp3Encoder',
'packages/mp3-encoder/dist/bundles/mediabunny-mp3-encoder',
'js', // The bundles are purely for the browser, not for Node (due to the peer dependecy)
'js', // The bundles are purely for the browser, not for Node (due to the peer dependency)
{
plugins: [
PluginExternalGlobal.externalGlobalPlugin({
@@ -134,7 +134,7 @@ const ac3Variants = await createVariants(
'packages/ac3/src/index.ts',
'MediabunnyAc3',
'packages/ac3/dist/bundles/mediabunny-ac3',
'js', // The bundles are purely for the browser, not for Node (due to the peer dependecy)
'js', // The bundles are purely for the browser, not for Node (due to the peer dependency)
{
plugins: [
PluginExternalGlobal.externalGlobalPlugin({
@@ -165,7 +165,7 @@ const aacEncoderVariants = await createVariants(
'packages/aac-encoder/src/index.ts',
'MediabunnyAacEncoder',
'packages/aac-encoder/dist/bundles/mediabunny-aac-encoder',
'js', // The bundles are purely for the browser, not for Node (due to the peer dependecy)
'js', // The bundles are purely for the browser, not for Node (due to the peer dependency)
{
plugins: [
PluginExternalGlobal.externalGlobalPlugin({
@@ -196,7 +196,7 @@ const flacEncoderVariants = await createVariants(
'packages/flac-encoder/src/index.ts',
'MediabunnyFlacEncoder',
'packages/flac-encoder/dist/bundles/mediabunny-flac-encoder',
'js', // The bundles are purely for the browser, not for Node (due to the peer dependecy)
'js', // The bundles are purely for the browser, not for Node (due to the peer dependency)
{
plugins: [
PluginExternalGlobal.externalGlobalPlugin({
@@ -227,7 +227,7 @@ const proresVariants = await createVariants(
'packages/prores/src/index.ts',
'MediabunnyProres',
'packages/prores/dist/bundles/mediabunny-prores',
'js', // The bundles are purely for the browser, not for Node (due to the peer dependecy)
'js', // The bundles are purely for the browser, not for Node (due to the peer dependency)
{
plugins: [
PluginExternalGlobal.externalGlobalPlugin({
src/codec-data.ts
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@@ -1878,7 +1878,7 @@ export type Av1CodecInfo = {
chromaSamplePosition: number;
};
/** Iterates over all OBUs in an AV1 packet bistream. */
/** Iterates over all OBUs in an AV1 packet bitstream. */
export const iterateAv1PacketObus = function* (packet: Uint8Array) {
// https://aomediacodec.github.io/av1-spec/av1-spec.pdf
src/conversion.ts
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@@ -838,7 +838,7 @@ export class Conversion {
if (this._options.trim?.start !== undefined) {
this._startTimestamp = this._options.trim.start;
} else {
// Compute the start timestamp from the set of filtered tracks. Techncially these can still be narrowed
// Compute the start timestamp from the set of filtered tracks. Technically these can still be narrowed
// down later due to discarded tracks, but we need to fix the start timestamp now due to track processing
// depending on it.
this._startTimestamp = Math.max(
src/flac/flac-demuxer.ts
+2 -2
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@@ -186,7 +186,7 @@ export class FlacDemuxer extends Demuxer {
// description is required, and has to be the following:
// 1. The bytes 0x66 0x4C 0x61 0x43 ("fLaC" in ASCII)
// 2. A metadata block (called the STREAMINFO block) as described in section 7 of [FLAC]
// 3. Optionaly (sic) other metadata blocks, that are not used by the specification
// 3. Optionally (sic) other metadata blocks, that are not used by the specification
bitstream.skipBits(16 * 8); // md5 hash
@@ -296,7 +296,7 @@ export class FlacDemuxer extends Demuxer {
// we expect that there are at least `minimumFrameSize` bytes left in the file
// Ideally we also want to validate the next header is valid
// to throw out an accidential sync word
// to throw out an accidental sync word
// The shortest valid FLAC header I can think of, based off the code
// of readFlacFrameHeader:
src/ogg/ogg-demuxer.ts
+2 -2
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@@ -638,11 +638,11 @@ class OggAudioTrackBacking implements InputAudioTrackBacking {
const lowPages: Page[] = [lowPage];
// First, let's perform a binary serach (bisection search) on the file to find the approximate page where
// First, let's perform a binary search (bisection search) on the file to find the approximate page where
// we'll find the packet. We want to find a page whose end packet position is less than or equal to the
// packet position we're searching for.
// Outer loop: Does the binary serach
// Outer loop: Does the binary search
outer:
while (lowPage.headerStartPos + lowPage.totalSize < high) {
const low = lowPage.headerStartPos;
src/resample.ts
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@@ -172,7 +172,7 @@ export class AudioResampler {
async add(audioSample: AudioSample) {
if (this.sourceSampleRate === null) {
// This is the first sample, so let's init the missing data. Initting the sample rate from the decoded
// This is the first sample, so let's init the missing data. Initing 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;
test/browser/conversion.test.ts
+13 -13
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@@ -177,8 +177,8 @@ test('HLS track assignability is kept #1', async () => {
await output.finalize();
const masterPlayist = sanitizeMasterPlaylist(new TextDecoder().decode(files.get('master.m3u8')));
expect(masterPlayist.match(/\.m3u8/g)?.length).toBe(1);
const masterPlaylist = sanitizeMasterPlaylist(new TextDecoder().decode(files.get('master.m3u8')));
expect(masterPlaylist.match(/\.m3u8/g)?.length).toBe(1);
using input = new Input({
formats: ALL_FORMATS,
@@ -208,8 +208,8 @@ test('HLS track assignability is kept #1', async () => {
const conversion = await Conversion.init({ input, output: newOutput });
await conversion.execute();
const newMasterPlayist = sanitizeMasterPlaylist(new TextDecoder().decode(files.get('new/master.m3u8')));
expect(newMasterPlayist).toBe(masterPlayist);
const newMasterPlaylist = sanitizeMasterPlaylist(new TextDecoder().decode(files.get('new/master.m3u8')));
expect(newMasterPlaylist).toBe(masterPlaylist);
});
test('HLS track assignability is kept #2', async () => {
@@ -256,8 +256,8 @@ test('HLS track assignability is kept #2', async () => {
await output.finalize();
const masterPlayist = sanitizeMasterPlaylist(new TextDecoder().decode(files.get('master.m3u8')));
expect(masterPlayist.match(/\.m3u8/g)?.length).toBe(2);
const masterPlaylist = sanitizeMasterPlaylist(new TextDecoder().decode(files.get('master.m3u8')));
expect(masterPlaylist.match(/\.m3u8/g)?.length).toBe(2);
using input = new Input({
formats: ALL_FORMATS,
@@ -287,8 +287,8 @@ test('HLS track assignability is kept #2', async () => {
const conversion = await Conversion.init({ input, output: newOutput });
await conversion.execute();
const newMasterPlayist = sanitizeMasterPlaylist(new TextDecoder().decode(files.get('new/master.m3u8')));
expect(newMasterPlayist).toBe(masterPlayist);
const newMasterPlaylist = sanitizeMasterPlaylist(new TextDecoder().decode(files.get('new/master.m3u8')));
expect(newMasterPlaylist).toBe(masterPlaylist);
});
test('HLS track assignability can be overridden', async () => {
@@ -335,8 +335,8 @@ test('HLS track assignability can be overridden', async () => {
await output.finalize();
const masterPlayist = sanitizeMasterPlaylist(new TextDecoder().decode(files.get('master.m3u8')));
expect(masterPlayist.match(/\.m3u8/g)?.length).toBe(2);
const masterPlaylist = sanitizeMasterPlaylist(new TextDecoder().decode(files.get('master.m3u8')));
expect(masterPlaylist.match(/\.m3u8/g)?.length).toBe(2);
using input = new Input({
formats: ALL_FORMATS,
@@ -371,9 +371,9 @@ test('HLS track assignability can be overridden', async () => {
});
await conversion.execute();
const newMasterPlayist = sanitizeMasterPlaylist(new TextDecoder().decode(files.get('new/master.m3u8')));
expect(newMasterPlayist).not.toBe(masterPlayist);
expect(newMasterPlayist.match(/\.m3u8/g)?.length).toBe(1);
const newMasterPlaylist = sanitizeMasterPlaylist(new TextDecoder().decode(files.get('new/master.m3u8')));
expect(newMasterPlaylist).not.toBe(masterPlaylist);
expect(newMasterPlaylist.match(/\.m3u8/g)?.length).toBe(1);
});
test('Fractional audio sample boundary', async () => {