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homekit: handle all opus sample rate and bitrate options

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This commit is contained in:
Koushik Dutta
2022-03-03 21:39:41 -08:00
parent dc4f21da0f
commit b255801acb
5 changed files with 32 additions and 14 deletions
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4
plugins/homekit/package-lock.json generated
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@@ -1,12 +1,12 @@
{
"name": "@scrypted/homekit",
"version": "0.0.212",
"version": "0.0.213",
"lockfileVersion": 2,
"requires": true,
"packages": {
"": {
"name": "@scrypted/homekit",
"version": "0.0.212",
"version": "0.0.213",
"dependencies": {
"@koush/qrcode-terminal": "^0.12.0",
"hap-nodejs": "file:../../external/HAP-NodeJS",

2
plugins/homekit/package.json
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@@ -39,5 +39,5 @@
"@types/node": "^14.17.9",
"@types/url-parse": "^1.4.3"
},
"version": "0.0.212"
"version": "0.0.213"
}

7
plugins/homekit/src/types/camera/camera-streaming-ffmpeg.ts
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@@ -1,5 +1,5 @@
import { FFMpegInput, ScryptedDevice, ScryptedMimeTypes, VideoCamera, MediaStreamOptions } from '@scrypted/sdk'
import { AudioStreamingCodecType, SRTPCryptoSuites, StartStreamRequest } from '../../hap';
import { AudioStreamingSamplerate, AudioStreamingCodecType, SRTPCryptoSuites, StartStreamRequest } from '../../hap';
import sdk from '@scrypted/sdk';
import child_process from 'child_process';
@@ -180,10 +180,13 @@ export async function startCameraStreamFfmpeg(device: ScryptedDevice & VideoCame
};
const mangler = await createBindZero();
const sender = createCameraStreamSender(aconfig, mangler.server,
session.audiossrc, session.startRequest.audio.pt,
session.prepareRequest.audio.port, session.prepareRequest.targetAddress,
session.startRequest.audio.rtcp_interval, session.startRequest.audio.packet_time,
session.startRequest.audio.rtcp_interval,
session.startRequest.audio.packet_time,
session.startRequest.audio.sample_rate,
);
session.opusMangler = mangler.server;
mangler.server.on('message', data => {

28
plugins/homekit/src/types/camera/camera-streaming-srtp-sender.ts
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@@ -1,5 +1,4 @@
import sdk from '@scrypted/sdk';
import dgram from 'dgram';
@@ -9,8 +8,9 @@ import { RtpPacket } from '../../../../../external/werift/packages/rtp/src/rtp/r
import { RtcpSenderInfo, RtcpSrPacket } from '../../../../../external/werift/packages/rtp/src/rtcp/sr';
import { Config } from '../../../../../external/werift/packages/rtp/src/srtp/session';
import { ntpTime } from './camera-utils';
import { AudioStreamingSamplerate } from '../../hap';
export function createCameraStreamSender(config: Config, sender: dgram.Socket, ssrc: number, payloadType: number, port: number, targetAddress: string, rtcpInterval: number, audioPacketTime?: number) {
export function createCameraStreamSender(config: Config, sender: dgram.Socket, ssrc: number, payloadType: number, port: number, targetAddress: string, rtcpInterval: number, audioPacketTime?: number, audioSampleRate?: AudioStreamingSamplerate) {
const srtpSession = new SrtpSession(config);
const srtcpSession = new SrtcpSession(config);
@@ -20,6 +20,19 @@ export function createCameraStreamSender(config: Config, sender: dgram.Socket, s
let lastRtcp = 0;
let firstSequenceNumber = 0;
let audioIntervalScale = 1;
if (audioPacketTime) {
switch (audioSampleRate) {
case AudioStreamingSamplerate.KHZ_24:
audioIntervalScale = 3;
break;
case AudioStreamingSamplerate.KHZ_16:
audioIntervalScale = 2;
break;
}
audioIntervalScale = audioIntervalScale * audioPacketTime / 20;
}
return (rtp: RtpPacket) => {
const now = Date.now();
@@ -38,14 +51,13 @@ export function createCameraStreamSender(config: Config, sender: dgram.Socket, s
// from HAP spec:
// RTP Payload Format for Opus Speech and Audio Codec RFC 7587 with an exception
// that Opus audio RTP Timestamp shall be based on RFC 3550.
// RFC 3550 indicates that 24k audio (which we advertise to HAP and it requests),
// should have an interval of 480 when the packet time is 20.
// RFC 3550 indicates that PCM audio based with a sample rate of 8k and a packet
// time of 20ms would have a monotonic interval of 8k / (1000 / 20) = 160.
// So 24k audio would have a monotonic interval of (24k / 8k) * 160 = 480.
// HAP spec also states that it may request packet times of 20, 30, 40, or 60.
// In practice, it requests 20 on LAN and 60 over LTE.
// In practice, HAP has been seen to request 20 on LAN and 60 over LTE.
// So the RTP timestamp must scale accordingly.
// TODO: Support more sample rates from Opus besides 24k, to possibly
// codec copy and repacketize?
rtp.header.timestamp = firstTimestamp + packetCount * 480 * audioPacketTime / 20;
rtp.header.timestamp = firstTimestamp + packetCount * 180 * audioIntervalScale;
}
lastTimestamp = rtp.header.timestamp;

5
plugins/homekit/src/types/camera/camera-streaming-srtp.ts
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@@ -80,7 +80,10 @@ export async function startCameraStreamSrtp(device: & VideoCamera, console: Cons
const audioSender = createCameraStreamSender(aconfig, session.audioReturn,
session.audiossrc, session.startRequest.audio.pt,
session.prepareRequest.audio.port, session.prepareRequest.targetAddress,
session.startRequest.audio.rtcp_interval, session.startRequest.audio.packet_time);
session.startRequest.audio.rtcp_interval,
session.startRequest.audio.packet_time,
session.startRequest.audio.sample_rate,
);
while (true) {
// trim the rtsp framing
if (isRtsp)