@basmilius/apple-raop 0.10.0 → 0.11.0

package/dist/index.d.mts

@@ -663,4 +663,4 @@ declare class RaopClient extends EventEmitter<EventMap> {
   static discover(deviceId: string, timingServer: TimingServer): Promise<RaopClient>;
 }
 //#endregion
-export { AudioPacketHeader, ControlClient, EncryptionType, MediaMetadata, MetadataType, PacketFifo, PlaybackInfo, RaopClient, RaopListener, RetransmitRequest, RaopRtspClient as RtspClient, Settings, Statistics, StreamClient, StreamContext, type StreamOptions, StreamProtocol, SyncPacket, decodeRetransmitRequest, getAudioProperties, getEncryptionTypes, getMetadataTypes, pctToDbfs };
+export { AudioPacketHeader, ControlClient, EncryptionType, type EventMap, type MediaMetadata, MetadataType, PacketFifo, type PlaybackInfo, RaopClient, type RaopListener, type RetransmitRequest, RaopRtspClient as RtspClient, type Settings, Statistics, StreamClient, type StreamContext, type StreamOptions, type StreamProtocol, SyncPacket, decodeRetransmitRequest, getAudioProperties, getEncryptionTypes, getMetadataTypes, pctToDbfs };

package/dist/index.mjs

@@ -2,7 +2,7 @@ import { createSocket } from "node:dgram";
 import { EventEmitter } from "node:events";
 import { DAAP, NTP, Plist } from "@basmilius/apple-encoding";
 import { createHash } from "node:crypto";
-import { Context, Discovery, generateActiveRemoteId, generateDacpId, generateSessionId, waitFor } from "@basmilius/apple-common";
+import { AUDIO_BYTES_PER_CHANNEL, AUDIO_CHANNELS, AUDIO_FRAMES_PER_PACKET, AUDIO_SAMPLE_RATE, Context, Discovery, generateActiveRemoteId, generateDacpId, generateSessionId, waitFor } from "@basmilius/apple-common";
 import { RtspClient } from "@basmilius/apple-rtsp";
 
 //#region src/types.ts
@@ -403,8 +403,6 @@ var ControlClient = class extends EventEmitter {
 //#region src/rtspClient.ts
 /** User-Agent header value sent with all RTSP requests. */
 const USER_AGENT = "AirPlay/550.10";
-/** Number of audio frames per RTP packet, used in SDP ANNOUNCE payload. */
-const FRAMES_PER_PACKET$1 = 352;
 /** Single-byte flag indicating unencrypted auth-setup mode. */
 const AUTH_SETUP_UNENCRYPTED = Buffer.from([1]);
 /**
@@ -475,7 +473,7 @@ function buildAnnouncePayload(options) {
 		"t=0 0",
 		"m=audio 0 RTP/AVP 96",
 		`a=rtpmap:96 L16/${options.sampleRate}/${options.channels}`,
-		`a=fmtp:96 ${FRAMES_PER_PACKET$1} 0 ${options.bitsPerChannel} 40 10 14 ${options.channels} 255 0 0 ${options.sampleRate}`
+		`a=fmtp:96 ${AUDIO_FRAMES_PER_PACKET} 0 ${options.bitsPerChannel} 40 10 14 ${options.channels} 255 0 0 ${options.sampleRate}`
 	].join("\r\n") + "\r\n";
 }
 /**
@@ -620,7 +618,7 @@ var RaopRtspClient = class extends RtspClient {
 				const parts = wwwAuthenticate.split("\"");
 				if (parts.length >= 5) {
 					this.#digestInfo = {
-						username: "pyatv",
+						username: "apple-protocols",
 						realm: parts[1],
 						password,
 						nonce: parts[3]
@@ -941,6 +939,7 @@ var StreamClient = class extends EventEmitter {
 	*/
 	close() {
 		this.#controlClient?.close();
+		this.#controlClient = void 0;
 	}
 	/**
 	* Initializes the streaming session by parsing device capabilities,
@@ -1002,8 +1001,12 @@ var StreamClient = class extends EventEmitter {
 		let transport;
 		try {
 			transport = createSocket("udp4");
-			await new Promise((resolve) => {
-				transport.connect(this.#streamContext.serverPort, this.#rtsp.connection.remoteIp, resolve);
+			await new Promise((resolve, reject) => {
+				transport.once("error", reject);
+				transport.connect(this.#streamContext.serverPort, this.#rtsp.connection.remoteIp, () => {
+					transport.removeListener("error", reject);
+					resolve();
+				});
 			});
 			this.#controlClient.start(this.#rtsp.connection.remoteIp);
 			if ((this.#metadataTypes & MetadataType.Progress) !== 0) {
@@ -1071,8 +1074,8 @@ var StreamClient = class extends EventEmitter {
 			if (numSent === 0) break;
 			stats.tick(numSent);
 			const framesBehind = stats.framesBehind;
-			if (framesBehind >= 352) {
-				const maxPackets = Math.min(Math.floor(framesBehind / 352), 3);
+			if (framesBehind >= AUDIO_FRAMES_PER_PACKET) {
+				const maxPackets = Math.min(Math.floor(framesBehind / AUDIO_FRAMES_PER_PACKET), 3);
 				this.#context.logger.debug(`Compensating with ${maxPackets} packets (${framesBehind} frames behind)`);
 				const [sentFrames, hasMorePackets] = await this.#sendNumberOfPackets(source, transport, maxPackets);
 				stats.tick(sentFrames);
@@ -1111,7 +1114,7 @@ var StreamClient = class extends EventEmitter {
 	*/
 	async #sendPacket(source, firstPacket, transport) {
 		if (this.#streamContext.paddingSent >= this.#streamContext.latency) return 0;
-		let frames = await source.readFrames(352);
+		let frames = await source.readFrames(AUDIO_FRAMES_PER_PACKET);
 		if (!frames) {
 			frames = Buffer.alloc(this.#streamContext.packetSize);
 			this.#streamContext.paddingSent += Math.floor(frames.length / this.#streamContext.frameSize);
@@ -1172,14 +1175,6 @@ var StreamClient = class extends EventEmitter {
 
 //#endregion
 //#region src/raop.ts
-/** Default audio sample rate in Hz (CD quality). */
-const SAMPLE_RATE = 44100;
-/** Default number of audio channels (stereo). */
-const CHANNELS = 2;
-/** Default bytes per channel sample (16-bit). */
-const BYTES_PER_CHANNEL = 2;
-/** Number of audio frames per RTP packet. */
-const FRAMES_PER_PACKET = 352;
 /**
 * High-level RAOP client for streaming audio to AirPlay receivers.
 * Wraps the RTSP handshake, UDP audio transport, control channel,
@@ -1401,20 +1396,20 @@ var RaopStreamProtocol = class {
 function createStreamContext() {
 	const rtptime = Math.floor(Math.random() * 4294967295);
 	return {
-		sampleRate: SAMPLE_RATE,
-		channels: CHANNELS,
-		bytesPerChannel: BYTES_PER_CHANNEL,
+		sampleRate: AUDIO_SAMPLE_RATE,
+		channels: AUDIO_CHANNELS,
+		bytesPerChannel: AUDIO_BYTES_PER_CHANNEL,
 		rtpseq: Math.floor(Math.random() * 65536),
 		rtptime,
 		headTs: rtptime,
-		latency: Math.floor(SAMPLE_RATE * 2),
+		latency: Math.floor(AUDIO_SAMPLE_RATE * 2),
 		serverPort: 0,
 		controlPort: 0,
 		rtspSession: "",
 		volume: -20,
 		position: 0,
-		packetSize: FRAMES_PER_PACKET * CHANNELS * BYTES_PER_CHANNEL,
-		frameSize: CHANNELS * BYTES_PER_CHANNEL,
+		packetSize: AUDIO_FRAMES_PER_PACKET * AUDIO_CHANNELS * AUDIO_BYTES_PER_CHANNEL,
+		frameSize: AUDIO_CHANNELS * AUDIO_BYTES_PER_CHANNEL,
 		paddingSent: 0,
 		reset() {
 			this.rtpseq = Math.floor(Math.random() * 65536);

package/package.json

@@ -1,7 +1,7 @@
 {
     "name": "@basmilius/apple-raop",
     "description": "Implementation of Apple's RAOP protocol in Node.js.",
-    "version": "0.10.0",
+    "version": "0.11.0",
     "type": "module",
     "license": "MIT",
     "author": {
@@ -46,10 +46,10 @@
         }
     },
     "dependencies": {
-        "@basmilius/apple-common": "0.10.0",
-        "@basmilius/apple-encoding": "0.10.0",
-        "@basmilius/apple-encryption": "0.10.0",
-        "@basmilius/apple-rtsp": "0.10.0"
+        "@basmilius/apple-common": "0.11.0",
+        "@basmilius/apple-encoding": "0.11.0",
+        "@basmilius/apple-encryption": "0.11.0",
+        "@basmilius/apple-rtsp": "0.11.0"
     },
     "devDependencies": {
         "@types/bun": "^1.3.11",