@homebridge/camera-utils/dist/return-audio-transcoder.js

Utilities to simplify homebridge camera plugin development

import { FfmpegProcess } from './ffmpeg-process.js';
import { reservePorts } from './ports.js';
import { getSsrc } from './rtp.js';
import { RtpSplitter } from './rtp-splitter.js';
import { createCryptoLine } from './srtp.js';
const defaultStartStreamReqeuest = {
    audio: {
        codec: 'AAC_eld',
        channel: 1,
        sample_rate: 16,
        pt: 110,
    },
};
export class ReturnAudioTranscoder {
    options;
    returnRtpSplitter;
    startStreamRequest;
    ffmpegProcess;
    constructor(options) {
        this.options = options;
        this.startStreamRequest = this.options.startStreamRequest || defaultStartStreamReqeuest;
        this.ffmpegProcess = new FfmpegProcess({
            ffmpegArgs: [
                '-hide_banner',
                '-protocol_whitelist',
                'pipe,udp,rtp,file,crypto',
                '-f',
                'sdp',
                '-acodec',
                this.startStreamRequest.audio.codec === 'OPUS' ? 'libopus' : 'libfdk_aac',
                '-i',
                'pipe:',
                '-map',
                '0:0',
                ...this.options.outputArgs,
            ],
            ...this.options,
        });
        // allow return audio splitter to be passed in if you want to create one in the prepare stream phase, and create the transcoder in the stream request phase
        this.returnRtpSplitter = options.returnAudioSplitter || new RtpSplitter();
    }
    reservedPortsPromise = reservePorts({ count: 2 });
    // Removed duplicate constructor
    async start() {
        const [rtpPort, rtcpPort] = await this.reservedPortsPromise;
        const { targetAddress, addressVersion, audio: { srtp_key: srtpKey, srtp_salt: srtpSalt }, } = this.options.prepareStreamRequest;
        const { ssrc: incomingAudioSsrc, rtcpPort: incomingAudioRtcpPort } = this.options.incomingAudioOptions;
        const { codec, sample_rate, channel, pt: packetType, } = this.startStreamRequest.audio;
        this.ffmpegProcess.writeStdin(
        // This SDP was generated using ffmpeg, and describes the type of packets we expect to receive from HomeKit.
        [
            'v=0',
            'o=- 0 0 IN IP4 127.0.0.1',
            's=Talk',
            `c=IN ${addressVersion
                .replace('v', '')
                .toUpperCase()} ${targetAddress}`,
            't=0 0',
            'a=tool:libavformat 58.38.100',
            `m=audio ${rtpPort} RTP/AVP ${packetType}`,
            'b=AS:24',
            ...(codec === 'OPUS'
                ? [
                    `a=rtpmap:${packetType} opus/${sample_rate}000/${channel}`,
                    `a=fmtp:${packetType} minptime=10;useinbandfec=1`,
                ]
                : [
                    `a=rtpmap:${packetType} MPEG4-GENERIC/${sample_rate}000/${channel}`,
                    `a=fmtp:${packetType} profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3; config=F8F0212C00BC00`,
                ]),
            createCryptoLine({
                srtpKey,
                srtpSalt,
            }),
        ].join('\n'));
        this.returnRtpSplitter.addMessageHandler(({ isRtpMessage, message }) => {
            // This splitter will receive all audio-related packets from HomeKit.
            // This includes RTP + RTCP for return audio, as well as RTCP for incoming audio
            if (!isRtpMessage && getSsrc(message) === incomingAudioSsrc) {
                return {
                    port: incomingAudioRtcpPort,
                };
            }
            return {
                port: isRtpMessage ? rtpPort : rtcpPort,
            };
        });
        // this is the port that needs to be passed to homebridge as audio.port
        return this.returnRtpSplitter.portPromise;
    }
    stop() {
        this.ffmpegProcess.stop();
        this.returnRtpSplitter.close();
    }
}
//# sourceMappingURL=return-audio-transcoder.js.map