From 7aa2cda44930344816005c1d8a91e565f792f214 Mon Sep 17 00:00:00 2001 From: Koushik Dutta Date: Wed, 18 Jan 2023 17:47:49 -0800 Subject: [PATCH] homekit: fix audio cuts --- plugins/homekit/package-lock.json | 4 ++-- plugins/homekit/package.json | 2 +- .../homekit/src/types/camera/camera-streaming-srtp-sender.ts | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) diff --git a/plugins/homekit/package-lock.json b/plugins/homekit/package-lock.json index 21a101b14..cc5caad12 100644 --- a/plugins/homekit/package-lock.json +++ b/plugins/homekit/package-lock.json @@ -1,12 +1,12 @@ { "name": "@scrypted/homekit", - "version": "1.1.93", + "version": "1.1.94", "lockfileVersion": 2, "requires": true, "packages": { "": { "name": "@scrypted/homekit", - "version": "1.1.93", + "version": "1.1.94", "dependencies": { "@koush/werift-src": "file:../../external/werift", "check-disk-space": "^3.3.0", diff --git a/plugins/homekit/package.json b/plugins/homekit/package.json index b67812d7f..d9ac66c0b 100644 --- a/plugins/homekit/package.json +++ b/plugins/homekit/package.json @@ -1,6 +1,6 @@ { "name": "@scrypted/homekit", - "version": "1.1.93", + "version": "1.1.94", "description": "HomeKit Plugin for Scrypted", "scripts": { "scrypted-setup-project": "scrypted-setup-project", diff --git a/plugins/homekit/src/types/camera/camera-streaming-srtp-sender.ts b/plugins/homekit/src/types/camera/camera-streaming-srtp-sender.ts index 83e1230d2..9a2f66f9d 100644 --- a/plugins/homekit/src/types/camera/camera-streaming-srtp-sender.ts +++ b/plugins/homekit/src/types/camera/camera-streaming-srtp-sender.ts @@ -138,7 +138,7 @@ export function createCameraStreamSender(console: Console, config: Config, sende // audio will work so long as the rtp timestamps are created properly: which is a construct of the sample rate // HAP requests, and the packet time is respected, // opus 48khz will work just fine. - rtp.header.timestamp = (firstTimestamp + packetCount * 180 * audioIntervalScale) % 0xFFFFFFFF; + rtp.header.timestamp = (firstTimestamp + packetCount * 160 * audioIntervalScale) % 0xFFFFFFFF; sendPacket(rtp); return; }