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homekit: remove dead code

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This commit is contained in:
Koushik Dutta
2022-08-31 23:56:14 -05:00
parent a98dc36d6d
commit aef62af06b
1 changed files with 38 additions and 92 deletions
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130
plugins/homekit/src/types/camera/camera-streaming-srtp.ts
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@@ -1,8 +1,6 @@
import { RtpPacket } from '@koush/werift-src/packages/rtp/src/rtp/rtp';
import { readLength } from '@scrypted/common/src/read-stream';
import { getSpsPps, parseSdp } from '@scrypted/common/src/sdp-utils';
import { FFmpegInput } from '@scrypted/sdk';
import net from 'net';
import { Readable } from 'stream';
import { RtspClient } from '../../../../../common/src/rtsp-server';
import { CameraStreamingSession, waitForFirstVideoRtcp } from './camera-streaming-session';
@@ -18,7 +16,6 @@ export async function startCameraStreamSrtp(media: FFmpegInput, console: Console
let { sdp } = mediaStreamOptions;
let socket: Readable;
let rtspClient: RtspClient;
const isRtsp = url.startsWith('rtsp');
let videoSender: (rtp: RtpPacket) => void;
let audioSender: (rtp: RtpPacket) => void;
@@ -30,106 +27,58 @@ export async function startCameraStreamSrtp(media: FFmpegInput, console: Console
session.killPromise.finally(cleanup);
if (isRtsp) {
rtspClient = new RtspClient(url);
rtspClient.requestTimeout = 1000;
rtspClient.client.once('close', cleanup);
rtspClient = new RtspClient(url);
rtspClient.requestTimeout = 1000;
rtspClient.client.once('close', cleanup);
try {
await rtspClient.options();
const sdpResponse = await rtspClient.describe();
sdp = sdpResponse.body.toString().trim();
const parsedSdp = parseSdp(sdp);
const video = parsedSdp.msections.find(msection => msection.type === 'video');
const audio = parsedSdp.msections.find(msection => msection.type === 'audio');
let channel = 0;
if (audio && !audioMode.mute) {
await rtspClient.setup({
type: 'tcp',
port: channel,
path: audio.control,
onRtp: (header, data) => {
if (session.killed)
return;
const rtp = RtpPacket.deSerialize(data);
if (audioMode.udpPort) {
session.audioReturn.send(rtp.serialize(), audioMode.udpPort, '127.0.0.1');
}
else {
audioSender(rtp);
}
},
});
channel += 2;
}
try {
await rtspClient.options();
const sdpResponse = await rtspClient.describe();
sdp = sdpResponse.body.toString().trim();
const parsedSdp = parseSdp(sdp);
const video = parsedSdp.msections.find(msection => msection.type === 'video');
const audio = parsedSdp.msections.find(msection => msection.type === 'audio');
let channel = 0;
if (audio && !audioMode.mute) {
await rtspClient.setup({
type: 'tcp',
port: channel,
path: video.control,
path: audio.control,
onRtp: (header, data) => {
if (session.killed)
return;
const rtp = RtpPacket.deSerialize(data);
videoSender(rtp);
if (audioMode.udpPort) {
session.audioReturn.send(rtp.serialize(), audioMode.udpPort, '127.0.0.1');
}
else {
audioSender(rtp);
}
},
});
channel += 2;
}
if (session.killed)
throw new Error('killed');
}
catch (e) {
cleanup();
throw e;
}
await rtspClient.setup({
type: 'tcp',
port: channel,
path: video.control,
onRtp: (header, data) => {
if (session.killed)
return;
const rtp = RtpPacket.deSerialize(data);
videoSender(rtp);
},
});
channel += 2;
if (session.killed)
throw new Error('killed');
}
else {
const u = new URL(url);
socket = net.connect(parseInt(u.port), u.hostname);
const startStreaming = async () => {
try {
await waitForFirstVideoRtcp(console, session);
while (!session.killed) {
const header = await readLength(socket, 2);
const length = header.readUInt16BE(0);
const data = await readLength(socket, length);
const rtp = RtpPacket.deSerialize(data);
if (!session.killed)
break;
const isAudio = audioPayloadTypes.includes(rtp.header.payloadType);
const isVideo = videoPayloadTypes.includes(rtp.header.payloadType);
if (isAudio && isVideo)
throw new Error('audio and video on same channel?');
if (isAudio) {
if (audioMode.udpPort) {
session.audioReturn.send(rtp.serialize(), audioMode.udpPort, '127.0.0.1');
}
else {
audioSender(rtp);
}
}
else if (isVideo) {
videoSender(rtp);
}
else {
console.warn('invalid payload type', rtp.header.payloadType);
}
}
}
catch (e) {
console.error('streaming ended', e);
}
finally {
cleanup();
}
}
process.nextTick(startStreaming);
catch (e) {
cleanup();
throw e;
}
const parsedSdp = parseSdp(sdp);
@@ -161,9 +110,6 @@ export async function startCameraStreamSrtp(media: FFmpegInput, console: Console
audioSender = as.sendRtp;
as.sendRtcp();
const audioPayloadTypes = parsedSdp.msections.find(msection => msection.type === 'audio')?.payloadTypes;
const videoPayloadTypes = video?.payloadTypes;
if (rtspClient) {
waitForFirstVideoRtcp(console, session).then(async () => {
try {