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

* Fix typos in source, tests, and docs

* Fix introduction guide typos

CLAUDE.md

@@ -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

@@ -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.
@@ -74,7 +74,7 @@ For more, see [the corresponding guide](./extensions/server).
 
 ## Motivation
 
-Mediabunny is the evolution of my previous libraries, [mp4-muxer](https://github.com/Vanilagy/mp4-muxer) and [webm-muxer](https://github.com/Vanilagy/webm-muxer), which were both created due to the advent of the WebCodecs API. While they fulfilled their job just fine, I saw a few painpoints:
+Mediabunny is the evolution of my previous libraries, [mp4-muxer](https://github.com/Vanilagy/mp4-muxer) and [webm-muxer](https://github.com/Vanilagy/webm-muxer), which were both created due to the advent of the WebCodecs API. While they fulfilled their job just fine, I saw a few pain points:
 - Lots of duplicated code between the two libraries, otherwise very similar API.
 - No help with the difficulties of navigating the WebCodecs API & related browser APIs.
 - "mp4-demuxer when??"

docs/guide/writing-media-files.md

@@ -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

@@ -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

@@ -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

@@ -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

@@ -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

@@ -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

@@ -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

@@ -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')));
+	const masterPlaylist = sanitizeMasterPlaylist(new TextDecoder().decode(files.get('master.m3u8')));
-	expect(masterPlayist.match(/\.m3u8/g)?.length).toBe(1);
+	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')));
+	const newMasterPlaylist = sanitizeMasterPlaylist(new TextDecoder().decode(files.get('new/master.m3u8')));
-	expect(newMasterPlayist).toBe(masterPlayist);
+	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')));
+	const masterPlaylist = sanitizeMasterPlaylist(new TextDecoder().decode(files.get('master.m3u8')));
-	expect(masterPlayist.match(/\.m3u8/g)?.length).toBe(2);
+	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')));
+	const newMasterPlaylist = sanitizeMasterPlaylist(new TextDecoder().decode(files.get('new/master.m3u8')));
-	expect(newMasterPlayist).toBe(masterPlayist);
+	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')));
+	const masterPlaylist = sanitizeMasterPlaylist(new TextDecoder().decode(files.get('master.m3u8')));
-	expect(masterPlayist.match(/\.m3u8/g)?.length).toBe(2);
+	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')));
+	const newMasterPlaylist = sanitizeMasterPlaylist(new TextDecoder().decode(files.get('new/master.m3u8')));
-	expect(newMasterPlayist).not.toBe(masterPlayist);
+	expect(newMasterPlaylist).not.toBe(masterPlaylist);
-	expect(newMasterPlayist.match(/\.m3u8/g)?.length).toBe(1);
+	expect(newMasterPlaylist.match(/\.m3u8/g)?.length).toBe(1);
 });
 
 test('Fractional audio sample boundary', async () => {