Improve HomeKit TLV format parser

Security Patch: Sanitize credentials on websocket error messages

.github/workflows/build.yml

@@ -19,7 +19,7 @@ jobs:
 
       - name: Setup Go
         uses: actions/setup-go@v5
-        with: { go-version: '1.21' }
+        with: { go-version: '1.24' }
 
       - name: Build go2rtc_win64
         env: { GOOS: windows, GOARCH: amd64 }
@@ -102,14 +102,14 @@ jobs:
         env: { GOOS: freebsd, GOARCH: amd64 }
         run: go build -ldflags "-s -w" -trimpath
       - name: Upload go2rtc_freebsd_amd64
-        uses: actions/upload-artifact@v3
+        uses: actions/upload-artifact@v4
         with: { name: go2rtc_freebsd_amd64, path: go2rtc }
 
       - name: Build go2rtc_freebsd_arm64
         env: { GOOS: freebsd, GOARCH: arm64 }
         run: go build -ldflags "-s -w" -trimpath
       - name: Upload go2rtc_freebsd_arm64
-        uses: actions/upload-artifact@v3
+        uses: actions/upload-artifact@v4
         with: { name: go2rtc_freebsd_arm64, path: go2rtc }
 
   docker-master:
@@ -123,7 +123,9 @@ jobs:
         id: meta
         uses: docker/metadata-action@v5
         with:
-          images: ${{ github.repository }}
+          images: |
+            ${{ github.repository }}
+            ghcr.io/${{ github.repository }}
           tags: |
             type=ref,event=branch
             type=semver,pattern={{version}},enable=false
@@ -142,13 +144,23 @@ jobs:
           username: ${{ secrets.DOCKERHUB_USERNAME }}
           password: ${{ secrets.DOCKERHUB_TOKEN }}
 
+      - name: Login to GitHub Container Registry
+        if: github.event_name != 'pull_request'
+        uses: docker/login-action@v3
+        with:
+          registry: ghcr.io
+          username: ${{ github.actor }}
+          password: ${{ secrets.GITHUB_TOKEN }}
+
       - name: Build and push
         uses: docker/build-push-action@v5
         with:
           context: .
+          file: docker/Dockerfile
           platforms: |
             linux/amd64
             linux/386
+            linux/arm/v6
             linux/arm/v7
             linux/arm64/v8
           push: ${{ github.event_name != 'pull_request' }}
@@ -168,7 +180,9 @@ jobs:
         id: meta-hw
         uses: docker/metadata-action@v5
         with:
-          images: ${{ github.repository }}
+          images: |
+            ${{ github.repository }}
+            ghcr.io/${{ github.repository }}
           flavor: |
             suffix=-hardware,onlatest=true
             latest=auto
@@ -189,15 +203,78 @@ jobs:
         with:
           username: ${{ secrets.DOCKERHUB_USERNAME }}
           password: ${{ secrets.DOCKERHUB_TOKEN }}
+      
+      - name: Login to GitHub Container Registry
+        if: github.event_name != 'pull_request'
+        uses: docker/login-action@v3
+        with:
+          registry: ghcr.io
+          username: ${{ github.actor }}
+          password: ${{ secrets.GITHUB_TOKEN }}
 
       - name: Build and push
         uses: docker/build-push-action@v5
         with:
           context: .
-          file: hardware.Dockerfile
+          file: docker/hardware.Dockerfile
           platforms: linux/amd64
           push: ${{ github.event_name != 'pull_request' }}
           tags: ${{ steps.meta-hw.outputs.tags }}
           labels: ${{ steps.meta-hw.outputs.labels }}
           cache-from: type=gha
           cache-to: type=gha,mode=max
+
+  docker-rockchip:
+    name: Build docker rockchip
+    runs-on: ubuntu-latest
+    steps:
+      - name: Checkout
+        uses: actions/checkout@v4
+
+      - name: Docker meta
+        id: meta-rk
+        uses: docker/metadata-action@v5
+        with:
+          images: |
+            ${{ github.repository }}
+            ghcr.io/${{ github.repository }}
+          flavor: |
+            suffix=-rockchip,onlatest=true
+            latest=auto
+          tags: |
+            type=ref,event=branch
+            type=semver,pattern={{version}},enable=false
+            type=match,pattern=v(.*),group=1
+
+      - name: Set up QEMU
+        uses: docker/setup-qemu-action@v3
+
+      - name: Set up Docker Buildx
+        uses: docker/setup-buildx-action@v3
+
+      - name: Login to DockerHub
+        if: github.event_name != 'pull_request'
+        uses: docker/login-action@v3
+        with:
+          username: ${{ secrets.DOCKERHUB_USERNAME }}
+          password: ${{ secrets.DOCKERHUB_TOKEN }}
+
+      - name: Login to GitHub Container Registry
+        if: github.event_name != 'pull_request'
+        uses: docker/login-action@v3
+        with:
+          registry: ghcr.io
+          username: ${{ github.actor }}
+          password: ${{ secrets.GITHUB_TOKEN }}
+
+      - name: Build and push
+        uses: docker/build-push-action@v5
+        with:
+          context: .
+          file: docker/rockchip.Dockerfile
+          platforms: linux/arm64
+          push: ${{ github.event_name != 'pull_request' }}
+          tags: ${{ steps.meta-rk.outputs.tags }}
+          labels: ${{ steps.meta-rk.outputs.labels }}
+          cache-from: type=gha
+          cache-to: type=gha,mode=max

.github/workflows/test.yml

@@ -26,7 +26,7 @@ jobs:
       - name: Setup Go
         uses: actions/setup-go@v5
         with:
-          go-version: '1.21'
+          go-version: '1.24'
 
       - name: Build Go binary
         run: go build -ldflags "-s -w" -trimpath -o ./go2rtc
@@ -79,6 +79,7 @@ jobs:
         uses: docker/build-push-action@v5
         with:
           context: .
+          file: docker/Dockerfile
           platforms: linux/${{ matrix.platform }}
           push: false
           load: true
@@ -92,7 +93,7 @@ jobs:
         uses: docker/build-push-action@v5
         with:
           context: .
-          file: hardware.Dockerfile
+          file: docker/hardware.Dockerfile
           platforms: linux/amd64
           push: false
           load: true

.gitignore

@@ -4,6 +4,11 @@
 go2rtc.yaml
 go2rtc.json
 
+go2rtc_freebsd*
+go2rtc_linux*
+go2rtc_mac*
+go2rtc_win*
+
 0_test.go
 
 .DS_Store

README.md

@@ -8,7 +8,7 @@
   [![goreport](https://goreportcard.com/badge/github.com/AlexxIT/go2rtc)](https://goreportcard.com/report/github.com/AlexxIT/go2rtc)
 </h1>
 
-Ultimate camera streaming application with support RTSP, WebRTC, HomeKit, FFmpeg, RTMP, etc.
+Ultimate camera streaming application with support for RTSP, WebRTC, HomeKit, FFmpeg, RTMP, etc.
 
 ![](assets/go2rtc.png)
 
@@ -20,11 +20,11 @@ Ultimate camera streaming application with support RTSP, WebRTC, HomeKit, FFmpeg
 - [publish](#publish-stream) any source to popular streaming services (YouTube, Telegram, etc.)
 - first project in the World with support streaming from [HomeKit Cameras](#source-homekit)
 - support H265 for WebRTC in browser (Safari only, [read more](https://github.com/AlexxIT/Blog/issues/5))
-- on the fly transcoding for unsupported codecs via [FFmpeg](#source-ffmpeg)
+- on-the-fly transcoding for unsupported codecs via [FFmpeg](#source-ffmpeg)
 - play audio files and live streams on some cameras with [speaker](#stream-to-camera)
 - multi-source 2-way [codecs negotiation](#codecs-negotiation)
    - mixing tracks from different sources to single stream
-   - auto match client supported codecs
+   - auto-match client-supported codecs
    - [2-way audio](#two-way-audio) for some cameras
 - streaming from private networks via [ngrok](#module-ngrok)
 - can be [integrated to](#module-api) any smart home platform or be used as [standalone app](#go2rtc-binary)
@@ -115,8 +115,8 @@ Ultimate camera streaming application with support RTSP, WebRTC, HomeKit, FFmpeg
 
 Download binary for your OS from [latest release](https://github.com/AlexxIT/go2rtc/releases/):
 
-- `go2rtc_win64.zip` - Windows 64-bit
-- `go2rtc_win32.zip` - Windows 32-bit
+- `go2rtc_win64.zip` - Windows 10+ 64-bit
+- `go2rtc_win32.zip` - Windows 7+ 32-bit
 - `go2rtc_win_arm64.zip` - Windows ARM 64-bit
 - `go2rtc_linux_amd64` - Linux 64-bit
 - `go2rtc_linux_i386` - Linux 32-bit
@@ -124,8 +124,10 @@ Download binary for your OS from [latest release](https://github.com/AlexxIT/go2
 - `go2rtc_linux_arm` - Linux ARM 32-bit (ex. Raspberry 32-bit OS)
 - `go2rtc_linux_armv6` - Linux ARMv6 (for old Raspberry 1 and Zero)
 - `go2rtc_linux_mipsel` - Linux MIPS (ex. [Xiaomi Gateway 3](https://github.com/AlexxIT/XiaomiGateway3), [Wyze cameras](https://github.com/gtxaspec/wz_mini_hacks))
-- `go2rtc_mac_amd64.zip` - Mac Intel 64-bit
-- `go2rtc_mac_arm64.zip` - Mac ARM 64-bit
+- `go2rtc_mac_amd64.zip` - macOS 10.13+ Intel 64-bit
+- `go2rtc_mac_arm64.zip` - macOS ARM 64-bit
+- `go2rtc_freebsd_amd64.zip` - FreeBSD 64-bit
+- `go2rtc_freebsd_arm64.zip` - FreeBSD ARM 64-bit
 
 Don't forget to fix the rights `chmod +x go2rtc_xxx_xxx` on Linux and Mac.
 
@@ -170,7 +172,7 @@ Available modules:
 - [api](#module-api) - HTTP API (important for WebRTC support)
 - [rtsp](#module-rtsp) - RTSP Server (important for FFmpeg support)
 - [webrtc](#module-webrtc) - WebRTC Server
-- [mp4](#module-mp4) - MSE, MP4 stream and MP4 shapshot Server
+- [mp4](#module-mp4) - MSE, MP4 stream and MP4 snapshot Server
 - [hls](#module-hls) - HLS TS or fMP4 stream Server
 - [mjpeg](#module-mjpeg) - MJPEG Server
 - [ffmpeg](#source-ffmpeg) - FFmpeg integration
@@ -180,11 +182,11 @@ Available modules:
 
 ### Module: Streams
 
-**go2rtc** support different stream source types. You can config one or multiple links of any type as stream source.
+**go2rtc** supports different stream source types. You can config one or multiple links of any type as a stream source.
 
 Available source types:
 
-- [rtsp](#source-rtsp) - `RTSP` and `RTSPS` cameras with [two way audio](#two-way-audio) support
+- [rtsp](#source-rtsp) - `RTSP` and `RTSPS` cameras with [two-way audio](#two-way-audio) support
 - [rtmp](#source-rtmp) - `RTMP` streams
 - [http](#source-http) - `HTTP-FLV`, `MPEG-TS`, `JPEG` (snapshots), `MJPEG` streams
 - [onvif](#source-onvif) - get camera `RTSP` link and snapshot link using `ONVIF` protocol
@@ -201,16 +203,16 @@ Available source types:
 - [gopro](#source-gopro) - GoPro cameras
 - [ivideon](#source-ivideon) - public cameras from [Ivideon](https://tv.ivideon.com/) service
 - [hass](#source-hass) - Home Assistant integration
-- [isapi](#source-isapi) - two way audio for Hikvision (ISAPI) cameras
+- [isapi](#source-isapi) - two-way audio for Hikvision (ISAPI) cameras
 - [roborock](#source-roborock) - Roborock vacuums with cameras
 - [webrtc](#source-webrtc) - WebRTC/WHEP sources
 - [webtorrent](#source-webtorrent) - WebTorrent source from another go2rtc
 
 Read more about [incoming sources](#incoming-sources)
 
-#### Two way audio
+#### Two-way audio
 
-Supported for sources:
+Supported sources:
 
 - [RTSP cameras](#source-rtsp) with [ONVIF Profile T](https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf) (back channel connection)
 - [DVRIP](#source-dvrip) cameras
@@ -220,9 +222,9 @@ Supported for sources:
 - [Exec](#source-exec) audio on server
 - [Any Browser](#incoming-browser) as IP-camera
 
-Two way audio can be used in browser with [WebRTC](#module-webrtc) technology. The browser will give access to the microphone only for HTTPS sites ([read more](https://stackoverflow.com/questions/52759992/how-to-access-camera-and-microphone-in-chrome-without-https)).
+Two-way audio can be used in browser with [WebRTC](#module-webrtc) technology. The browser will give access to the microphone only for HTTPS sites ([read more](https://stackoverflow.com/questions/52759992/how-to-access-camera-and-microphone-in-chrome-without-https)).
 
-go2rtc also support [play audio](#stream-to-camera) files and live streams on this cameras.
+go2rtc also supports [play audio](#stream-to-camera) files and live streams on this cameras.
 
 #### Source: RTSP
 
@@ -231,7 +233,7 @@ streams:
   sonoff_camera: rtsp://rtsp:12345678@192.168.1.123/av_stream/ch0
   dahua_camera:
     - rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif
-    - rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=1
+    - rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=1#backchannel=0
   amcrest_doorbell:
     - rtsp://username:password@192.168.1.123:554/cam/realmonitor?channel=1&subtype=0#backchannel=0
   unifi_camera: rtspx://192.168.1.123:7441/fD6ouM72bWoFijxK
@@ -240,13 +242,13 @@ streams:
 
 **Recommendations**
 
-- **Amcrest Doorbell** users may want to disable two way audio, because with an active stream you won't have a call button working. You need to add `#backchannel=0` to the end of your RTSP link in YAML config file
-- **Dahua Doorbell** users may want to change backchannel [audio codec](https://github.com/AlexxIT/go2rtc/issues/52)
-- **Reolink** users may want NOT to use RTSP protocol at all, some camera models have a very awful unusable stream implementation
+- **Amcrest Doorbell** users may want to disable two-way audio, because with an active stream, you won't have a working call button. You need to add `#backchannel=0` to the end of your RTSP link in YAML config file
+- **Dahua Doorbell** users may want to change [audio codec](https://github.com/AlexxIT/go2rtc/issues/49#issuecomment-2127107379) for proper 2-way audio. Make sure not to request backchannel multiple times by adding `#backchannel=0` to other stream sources of the same doorbell. The `unicast=true&proto=Onvif` is preferred for 2-way audio as this makes the doorbell accept multiple codecs for the incoming audio
+- **Reolink** users may want NOT to use RTSP protocol at all, some camera models have a very awful, unusable stream implementation
 - **Ubiquiti UniFi** users may want to disable HTTPS verification. Use `rtspx://` prefix instead of `rtsps://`. And don't use `?enableSrtp` [suffix](https://github.com/AlexxIT/go2rtc/issues/81)
 - **TP-Link Tapo** users may skip login and password, because go2rtc support login [without them](https://drmnsamoliu.github.io/video.html)
-- If your camera has two RTSP links - you can add both of them as sources. This is useful when streams has different codecs, as example AAC audio with main stream and PCMU/PCMA audio with second stream
-- If the stream from your camera is glitchy, try using [ffmpeg source](#source-ffmpeg). It will not add CPU load if you won't use transcoding
+- If your camera has two RTSP links, you can add both as sources. This is useful when streams have different codecs, for example AAC audio with main stream and PCMU/PCMA audio with second stream
+- If the stream from your camera is glitchy, try using [ffmpeg source](#source-ffmpeg). It will not add CPU load if you don't use transcoding
 - If the stream from your camera is very glitchy, try to use transcoding with [ffmpeg source](#source-ffmpeg)
 
 **Other options**
@@ -255,7 +257,7 @@ Format: `rtsp...#{param1}#{param2}#{param3}`
 
 - Add custom timeout `#timeout=30` (in seconds)
 - Ignore audio - `#media=video` or ignore video - `#media=audio` 
-- Ignore two way audio API `#backchannel=0` - important for some glitchy cameras
+- Ignore two-way audio API `#backchannel=0` - important for some glitchy cameras
 - Use WebSocket transport `#transport=ws...`
 
 **RTSP over WebSocket**
@@ -270,7 +272,7 @@ streams:
 
 #### Source: RTMP
 
-You can get stream from RTMP server, for example [Nginx with nginx-rtmp-module](https://github.com/arut/nginx-rtmp-module).
+You can get a stream from an RTMP server, for example [Nginx with nginx-rtmp-module](https://github.com/arut/nginx-rtmp-module).
 
 ```yaml
 streams:
@@ -286,7 +288,7 @@ Support Content-Type:
 - **HTTP-MJPEG** (`multipart/x`) - simple MJPEG stream over HTTP
 - **MPEG-TS** (`video/mpeg`) - legacy [streaming format](https://en.wikipedia.org/wiki/MPEG_transport_stream)
 
-Source also support HTTP and TCP streams with autodetection for different formats: **MJPEG**, **H.264/H.265 bitstream**, **MPEG-TS**.
+Source also supports HTTP and TCP streams with autodetection for different formats: **MJPEG**, **H.264/H.265 bitstream**, **MPEG-TS**.
 
 ```yaml
 streams:
@@ -306,7 +308,7 @@ streams:
   custom_header: "https://mjpeg.sanford.io/count.mjpeg#header=Authorization: Bearer XXX"
 ```
 
-**PS.** Dahua camera has bug: if you select MJPEG codec for RTSP second stream - snapshot won't work.
+**PS.** Dahua camera has a bug: if you select MJPEG codec for RTSP second stream, snapshot won't work.
 
 #### Source: ONVIF
 
@@ -314,7 +316,7 @@ streams:
 
 The source is not very useful if you already know RTSP and snapshot links for your camera. But it can be useful if you don't.
 
-**WebUI > Add** webpage support ONVIF autodiscovery. Your server must be on the same subnet as the camera. If you use docker, you must use "network host".
+**WebUI > Add** webpage support ONVIF autodiscovery. Your server must be on the same subnet as the camera. If you use Docker, you must use "network host".
 
 ```yaml
 streams:
@@ -325,7 +327,7 @@ streams:
 
 #### Source: FFmpeg
 
-You can get any stream or file or device via FFmpeg and push it to go2rtc. The app will automatically start FFmpeg with the proper arguments when someone starts watching the stream.
+You can get any stream, file or device via FFmpeg and push it to go2rtc. The app will automatically start FFmpeg with the proper arguments when someone starts watching the stream.
 
 - FFmpeg preistalled for **Docker** and **Hass Add-on** users
 - **Hass Add-on** users can target files from [/media](https://www.home-assistant.io/more-info/local-media/setup-media/) folder
@@ -340,7 +342,7 @@ streams:
   # [FILE] video will be transcoded to H264, audio will be skipped
   file2: ffmpeg:/media/BigBuckBunny.mp4#video=h264
 
-  # [FILE] video will be copied, audio will be transcoded to pcmu
+  # [FILE] video will be copied, audio will be transcoded to PCMU
   file3: ffmpeg:/media/BigBuckBunny.mp4#video=copy#audio=pcmu
 
   # [HLS] video will be copied, audio will be skipped
@@ -350,12 +352,12 @@ streams:
   mjpeg: ffmpeg:http://185.97.122.128/cgi-bin/faststream.jpg#video=h264
 
   # [RTSP] video with rotation, should be transcoded, so select H264
-  rotate: ffmpeg:rtsp://rtsp:12345678@192.168.1.123/av_stream/ch0#video=h264#rotate=90
+  rotate: ffmpeg:rtsp://12345678@192.168.1.123/av_stream/ch0#video=h264#rotate=90
 ```
 
-All trascoding formats has [built-in templates](https://github.com/AlexxIT/go2rtc/blob/master/internal/ffmpeg/ffmpeg.go): `h264`, `h265`, `opus`, `pcmu`, `pcmu/16000`, `pcmu/48000`, `pcma`, `pcma/16000`, `pcma/48000`, `aac`, `aac/16000`.
+All transcoding formats have [built-in templates](https://github.com/AlexxIT/go2rtc/blob/master/internal/ffmpeg/ffmpeg.go): `h264`, `h265`, `opus`, `pcmu`, `pcmu/16000`, `pcmu/48000`, `pcma`, `pcma/16000`, `pcma/48000`, `aac`, `aac/16000`.
 
-But you can override them via YAML config. You can also add your own formats to config and use them with source params.
+But you can override them via YAML config. You can also add your own formats to the config and use them with source params.
 
 ```yaml
 ffmpeg:
@@ -383,12 +385,12 @@ Read more about [hardware acceleration](https://github.com/AlexxIT/go2rtc/wiki/H
 
 #### Source: FFmpeg Device
 
-You can get video from any USB-camera or Webcam as RTSP or WebRTC stream. This is part of FFmpeg integration.
+You can get video from any USB camera or Webcam as RTSP or WebRTC stream. This is part of FFmpeg integration.
 
-- check available devices in Web interface
+- check available devices in web interface
 - `video_size` and `framerate` must be supported by your camera!
 - for Linux supported only video for now
-- for macOS you can stream Facetime camera or whole Desktop!
+- for macOS you can stream FaceTime camera or whole desktop!
 - for macOS important to set right framerate
 
 Format: `ffmpeg:device?{input-params}#{param1}#{param2}#{param3}`
@@ -406,7 +408,7 @@ streams:
 
 Exec source can run any external application and expect data from it. Two transports are supported - **pipe** (*from [v1.5.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.5.0)*) and **RTSP**.
 
-If you want to use **RTSP** transport - the command must contain the `{output}` argument in any place. On launch, it will be replaced by the local address of the RTSP server.
+If you want to use **RTSP** transport, the command must contain the `{output}` argument in any place. On launch, it will be replaced by the local address of the RTSP server.
 
 **pipe** reads data from app stdout in different formats: **MJPEG**, **H.264/H.265 bitstream**, **MPEG-TS**. Also pipe can write data to app stdin in two formats: **PCMA** and **PCM/48000**.
 
@@ -416,11 +418,11 @@ The source can be used with:
 - [FFplay](https://ffmpeg.org/ffplay.html) - play audio on your server
 - [GStreamer](https://gstreamer.freedesktop.org/)
 - [Raspberry Pi Cameras](https://www.raspberrypi.com/documentation/computers/camera_software.html)
-- any your own software
+- any of your own software
 
 Pipe commands support parameters (format: `exec:{command}#{param1}#{param2}`):
 
-- `killsignal` - signal which will be send to stop the process (numeric form)
+- `killsignal` - signal which will be sent to stop the process (numeric form)
 - `killtimeout` - time in seconds for forced termination with sigkill
 - `backchannel` - enable backchannel for two-way audio
 
@@ -437,7 +439,7 @@ streams:
 
 #### Source: Echo
 
-Some sources may have a dynamic link. And you will need to get it using a bash or python script. Your script should echo a link to the source. RTSP, FFmpeg or any of the [supported sources](#module-streams).
+Some sources may have a dynamic link. And you will need to get it using a Bash or Python script. Your script should echo a link to the source. RTSP, FFmpeg or any of the [supported sources](#module-streams).
 
 **Docker** and **Hass Add-on** users has preinstalled `python3`, `curl`, `jq`.
 
@@ -459,20 +461,20 @@ Like `echo` source, but uses the built-in [expr](https://github.com/antonmedv/ex
 **Important:**
 
 - You can use HomeKit Cameras **without Apple devices** (iPhone, iPad, etc.), it's just a yet another protocol
-- HomeKit device can be paired with only one ecosystem. So, if you have paired it to an iPhone (Apple Home) - you can't pair it with Home Assistant or go2rtc. Or if you have paired it to go2rtc - you can't pair it with iPhone
-- HomeKit device should be in same network with working [mDNS](https://en.wikipedia.org/wiki/Multicast_DNS) between device and go2rtc
+- HomeKit device can be paired with only one ecosystem. So, if you have paired it to an iPhone (Apple Home), you can't pair it with Home Assistant or go2rtc. Or if you have paired it to go2rtc, you can't pair it with an iPhone
+- HomeKit device should be on the same network with working [mDNS](https://en.wikipedia.org/wiki/Multicast_DNS) between the device and go2rtc
 
-go2rtc support import paired HomeKit devices from [Home Assistant](#source-hass). So you can use HomeKit camera with Hass and go2rtc simultaneously. If you using Hass, I recommend pairing devices with it, it will give you more options.
+go2rtc supports importing paired HomeKit devices from [Home Assistant](#source-hass). So you can use HomeKit camera with Hass and go2rtc simultaneously. If you are using Hass, I recommend pairing devices with it; it will give you more options.
 
-You can pair device with go2rtc on the HomeKit page. If you can't see your devices - reload the page. Also try reboot your HomeKit device (power off). If you still can't see it - you have a problems with mDNS.
+You can pair device with go2rtc on the HomeKit page. If you can't see your devices, reload the page. Also, try rebooting your HomeKit device (power off). If you still can't see it, you have a problem with mDNS.
 
-If you see a device but it does not have a pair button - it is paired to some ecosystem (Apple Home, Home Assistant, HomeBridge etc). You need to delete device from that ecosystem, and it will be available for pairing. If you cannot unpair device, you will have to reset it.
+If you see a device but it does not have a pairing button, it is paired to some ecosystem (Apple Home, Home Assistant, HomeBridge etc). You need to delete the device from that ecosystem, and it will be available for pairing. If you cannot unpair the device, you will have to reset it.
 
 **Important:**
 
-- HomeKit audio uses very non-standard **AAC-ELD** codec with very non-standard params and specification violation
+- HomeKit audio uses very non-standard **AAC-ELD** codec with very non-standard params and specification violations
 - Audio can't be played in `VLC` and probably any other player
-- Audio should be transcoded for using with MSE, WebRTC, etc.
+- Audio should be transcoded for use with MSE, WebRTC, etc.
 
 Recommended settings for using HomeKit Camera with WebRTC, MSE, MP4, RTSP:
 
@@ -494,7 +496,7 @@ RTSP link with "normal" audio for any player: `rtsp://192.168.1.123:8554/aqara_g
 Other names: [ESeeCloud](http://www.eseecloud.com/), [dvr163](http://help.dvr163.com/).
 
 - you can skip `username`, `password`, `port`, `ch` and `stream` if they are default
-- setup separate streams for different channels and streams
+- set up separate streams for different channels and streams
 
 ```yaml
 streams:
@@ -508,7 +510,7 @@ streams:
 Other names: DVR-IP, NetSurveillance, Sofia protocol (NETsurveillance ActiveX plugin XMeye SDK).
 
 - you can skip `username`, `password`, `port`, `channel` and `subtype` if they are default
-- setup separate streams for different channels
+- set up separate streams for different channels
 - use `subtype=0` for Main stream, and `subtype=1` for Extra1 stream
 - only the TCP protocol is supported
 
@@ -530,7 +532,7 @@ streams:
 - stream quality is the same as [RTSP protocol](https://www.tapo.com/en/faq/34/)
 - use the **cloud password**, this is not the RTSP password! you do not need to add a login!
 - you can also use UPPERCASE MD5 hash from your cloud password with `admin` username
-- some new camera firmwares requires SHA256 instead of MD5
+- some new camera firmwares require SHA256 instead of MD5
 
 ```yaml
 streams:
@@ -571,7 +573,7 @@ Support streaming from [GoPro](https://gopro.com/) cameras, connected via USB or
 
 #### Source: Ivideon
 
-Support public cameras from service [Ivideon](https://tv.ivideon.com/).
+Support public cameras from the service [Ivideon](https://tv.ivideon.com/).
 
 ```yaml
 streams:
@@ -589,7 +591,7 @@ Support import camera links from [Home Assistant](https://www.home-assistant.io/
 
 ```yaml
 hass:
-  config: "/config"  # skip this setting if you Hass Add-on user
+  config: "/config"  # skip this setting if you Hass add-on user
 
 streams:
   generic_camera: hass:Camera1  # Settings > Integrations > Integration Name
@@ -598,9 +600,9 @@ streams:
 
 **WebRTC Cameras** (*from [v1.6.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.6.0)*)
 
-Any cameras in WebRTC format are supported. But at the moment Home Assistant only supports some [Nest](https://www.home-assistant.io/integrations/nest/) cameras in this fomat.
+Any cameras in WebRTC format are supported. But at the moment Home Assistant only supports some [Nest](https://www.home-assistant.io/integrations/nest/) cameras in this format.
 
-**Important.** The Nest API only allows you to get a link to a stream for 5 minutes. Do not use this with Frigate! If the stream expires, Frigate will consume all available ram on your machine within seconds. It's recommended to use [Nest source](#source-nest) - it supports extending the stream.
+**Important.** The Nest API only allows you to get a link to a stream for 5 minutes. Do not use this with Frigate! If the stream expires, Frigate will consume all available RAM on your machine within seconds. It's recommended to use [Nest source](#source-nest) - it supports extending the stream.
 
 ```yaml
 streams:
@@ -612,13 +614,13 @@ streams:
 
 **RTSP Cameras**
 
-By default, the Home Assistant API does not allow you to get dynamic RTSP link to a camera stream. So more cameras, like [Tuya](https://www.home-assistant.io/integrations/tuya/), and possibly others can also be imported by using [this method](https://github.com/felipecrs/hass-expose-camera-stream-source#importing-home-assistant-cameras-to-go2rtc-andor-frigate).
+By default, the Home Assistant API does not allow you to get a dynamic RTSP link to a camera stream. So more cameras, like [Tuya](https://www.home-assistant.io/integrations/tuya/), and possibly others, can also be imported using [this method](https://github.com/felipecrs/hass-expose-camera-stream-source#importing-home-assistant-cameras-to-go2rtc-andor-frigate).
 
 #### Source: ISAPI
 
 *[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)*
 
-This source type support only backchannel audio for Hikvision ISAPI protocol. So it should be used as second source in addition to the RTSP protocol.
+This source type supports only backchannel audio for the Hikvision ISAPI protocol. So it should be used as a second source in addition to the RTSP protocol.
 
 ```yaml
 streams:
@@ -631,9 +633,9 @@ streams:
 
 *[New in v1.6.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.6.0)*
 
-Currently only WebRTC cameras are supported.
+Currently, only WebRTC cameras are supported.
 
-For simplicity, it is recommended to connect the Nest/WebRTC camera to the [Home Assistant](#source-hass). But if you can somehow get the below parameters - Nest/WebRTC source will work without Hass.
+For simplicity, it is recommended to connect the Nest/WebRTC camera to the [Home Assistant](#source-hass). But if you can somehow get the below parameters, Nest/WebRTC source will work without Hass.
 
 ```yaml
 streams:
@@ -644,28 +646,29 @@ streams:
 
 *[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)*
 
-This source type support Roborock vacuums with cameras. Known working models:
+This source type supports Roborock vacuums with cameras. Known working models:
 
 - Roborock S6 MaxV - only video (the vacuum has no microphone)
-- Roborock S7 MaxV - video and two way audio
+- Roborock S7 MaxV - video and two-way audio
+- Roborock Qrevo MaxV - video and two-way audio
 
-Source support load Roborock credentials from Home Assistant [custom integration](https://github.com/humbertogontijo/homeassistant-roborock). Otherwise, you need to log in to your Roborock account (MiHome account is not supported). Go to: go2rtc WebUI > Add webpage. Copy `roborock://...` source for your vacuum and paste it to `go2rtc.yaml` config.
+Source supports loading Roborock credentials from Home Assistant [custom integration](https://github.com/humbertogontijo/homeassistant-roborock) or the [core integration](https://www.home-assistant.io/integrations/roborock). Otherwise, you need to log in to your Roborock account (MiHome account is not supported). Go to: go2rtc WebUI > Add webpage. Copy `roborock://...` source for your vacuum and paste it to `go2rtc.yaml` config.
 
-If you have graphic pin for your vacuum - add it as numeric pin (lines: 123, 456, 678) to the end of the roborock-link.
+If you have a graphic PIN for your vacuum, add it as a numeric PIN (lines: 123, 456, 789) to the end of the `roborock` link.
 
 #### Source: WebRTC
 
 *[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)*
 
-This source type support four connection formats.
+This source type supports four connection formats.
 
 **whep**
 
-[WebRTC/WHEP](https://datatracker.ietf.org/doc/draft-murillo-whep/) - is replaced by [WebRTC/WISH](https://datatracker.ietf.org/doc/charter-ietf-wish/02/) standard for WebRTC video/audio viewers. But it may already be supported in some third-party software. It is supported in go2rtc.
+[WebRTC/WHEP](https://datatracker.ietf.org/doc/draft-murillo-whep/) is replaced by [WebRTC/WISH](https://datatracker.ietf.org/doc/charter-ietf-wish/02/) standard for WebRTC video/audio viewers. But it may already be supported in some third-party software. It is supported in go2rtc.
 
 **go2rtc**
 
-This format is only supported in go2rtc. Unlike WHEP it supports asynchronous WebRTC connection and two way audio.
+This format is only supported in go2rtc. Unlike WHEP, it supports asynchronous WebRTC connections and two-way audio.
 
 **openipc** (*from [v1.7.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.7.0)*)
 
@@ -673,22 +676,27 @@ Support connection to [OpenIPC](https://openipc.org/) cameras.
 
 **wyze** (*from [v1.6.1](https://github.com/AlexxIT/go2rtc/releases/tag/v1.6.1)*)
 
-Supports connection to [Wyze](https://www.wyze.com/) cameras, using WebRTC protocol. You can use [docker-wyze-bridge](https://github.com/mrlt8/docker-wyze-bridge) project to get connection credentials.
+Supports connection to [Wyze](https://www.wyze.com/) cameras, using WebRTC protocol. You can use the [docker-wyze-bridge](https://github.com/mrlt8/docker-wyze-bridge) project to get connection credentials.
 
 **kinesis** (*from [v1.6.1](https://github.com/AlexxIT/go2rtc/releases/tag/v1.6.1)*)
 
-Supports [Amazon Kinesis Video Streams](https://aws.amazon.com/kinesis/video-streams/), using WebRTC protocol. You need to specify signalling WebSocket URL with all credentials in query params, `client_id` and `ice_servers` list in [JSON format](https://developer.mozilla.org/en-US/docs/Web/API/RTCIceServer).
+Supports [Amazon Kinesis Video Streams](https://aws.amazon.com/kinesis/video-streams/), using WebRTC protocol. You need to specify the signalling WebSocket URL with all credentials in query params, `client_id` and `ice_servers` list in [JSON format](https://developer.mozilla.org/en-US/docs/Web/API/RTCIceServer).
+
+**switchbot**
+
+Support connection to [SwitchBot](https://us.switch-bot.com/) cameras that are based on Kinesis Video Streams. Specifically, this includes [Pan/Tilt Cam Plus 2K](https://us.switch-bot.com/pages/switchbot-pan-tilt-cam-plus-2k) and [Pan/Tilt Cam Plus 3K](https://us.switch-bot.com/pages/switchbot-pan-tilt-cam-plus-3k). `Outdoor Spotlight Cam 1080P`, `Outdoor Spotlight Cam 2K`, `Pan/Tilt Cam`, `Pan/Tilt Cam 2K`, `Indoor Cam` are based on Tuya, so this feature is not available.
 
 ```yaml
 streams:
-  webrtc-whep:    webrtc:http://192.168.1.123:1984/api/webrtc?src=camera1
-  webrtc-go2rtc:  webrtc:ws://192.168.1.123:1984/api/ws?src=camera1
-  webrtc-openipc: webrtc:ws://192.168.1.123/webrtc_ws#format=openipc#ice_servers=[{"urls":"stun:stun.kinesisvideo.eu-north-1.amazonaws.com:443"}]
-  webrtc-wyze:    webrtc:http://192.168.1.123:5000/signaling/camera1?kvs#format=wyze
-  webrtc-kinesis: webrtc:wss://...amazonaws.com/?...#format=kinesis#client_id=...#ice_servers=[{...},{...}]
+  webrtc-whep:      webrtc:http://192.168.1.123:1984/api/webrtc?src=camera1
+  webrtc-go2rtc:    webrtc:ws://192.168.1.123:1984/api/ws?src=camera1
+  webrtc-openipc:   webrtc:ws://192.168.1.123/webrtc_ws#format=openipc#ice_servers=[{"urls":"stun:stun.kinesisvideo.eu-north-1.amazonaws.com:443"}]
+  webrtc-wyze:      webrtc:http://192.168.1.123:5000/signaling/camera1?kvs#format=wyze
+  webrtc-kinesis:   webrtc:wss://...amazonaws.com/?...#format=kinesis#client_id=...#ice_servers=[{...},{...}]
+  webrtc-switchbot: webrtc:wss://...amazonaws.com/?...#format=switchbot#resolution=hd#client_id=...#ice_servers=[{...},{...}]
 ```
 
-**PS.** For `kinesis` sources you can use [echo](#source-echo) to get connection params using `bash`/`python` or any other script language.
+**PS.** For `kinesis` sources, you can use [echo](#source-echo) to get connection params using `bash`, `python` or any other script language.
 
 #### Source: WebTorrent
 
@@ -707,9 +715,9 @@ By default, go2rtc establishes a connection to the source when any client reques
 
 - Go2rtc also can accepts incoming sources in [RTSP](#module-rtsp), [RTMP](#module-rtmp), [HTTP](#source-http) and **WebRTC/WHIP** formats
 - Go2rtc won't stop such a source if it has no clients
-- You can push data only to existing stream (create stream with empty source in config)
-- You can push multiple incoming sources to same stream
-- You can push data to non empty stream, so it will have additional codecs inside
+- You can push data only to an existing stream (create a stream with empty source in config)
+- You can push multiple incoming sources to the same stream
+- You can push data to a non-empty stream, so it will have additional codecs inside
 
 **Examples**
 
@@ -734,11 +742,11 @@ By default, go2rtc establishes a connection to the source when any client reques
 
 *[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)*
 
-You can turn the browser of any PC or mobile into an IP-camera with support video and two way audio. Or even broadcast your PC screen:
+You can turn the browser of any PC or mobile into an IP camera with support for video and two-way audio. Or even broadcast your PC screen:
 
 1. Create empty stream in the `go2rtc.yaml`
 2. Go to go2rtc WebUI
-3. Open `links` page for you stream
+3. Open `links` page for your stream
 4. Select `camera+microphone` or `display+speaker` option
 5. Open `webrtc` local page (your go2rtc **should work over HTTPS!**) or `share link` via [WebTorrent](#module-webtorrent) technology (work over HTTPS by default)
 
@@ -754,7 +762,7 @@ You can use **OBS Studio** or any other broadcast software with [WHIP](https://w
 
 *[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)*
 
-go2rtc support play audio files (ex. music or [TTS](https://www.home-assistant.io/integrations/#text-to-speech)) and live streams (ex. radio) on cameras with [two way audio](#two-way-audio) support (RTSP/ONVIF cameras, TP-Link Tapo, Hikvision ISAPI, Roborock vacuums, any Browser).
+go2rtc supports playing audio files (ex. music or [TTS](https://www.home-assistant.io/integrations/#text-to-speech)) and live streams (ex. radio) on cameras with [two-way audio](#two-way-audio) support (RTSP/ONVIF cameras, TP-Link Tapo, Hikvision ISAPI, Roborock vacuums, any Browser).
 
 API example:
 
@@ -767,7 +775,7 @@ POST http://localhost:1984/api/streams?dst=camera1&src=ffmpeg:http://example.com
 - you can check camera codecs on the go2rtc WebUI info page when the stream is active
 - some cameras support only low quality `PCMA/8000` codec (ex. [Tapo](#source-tapo))
 - it is recommended to choose higher quality formats if your camera supports them (ex. `PCMA/48000` for some Dahua cameras)
-- if you play files over http-link, you need to add `#input=file` params for transcoding, so file will be transcoded and played in real time
+- if you play files over `http` link, you need to add `#input=file` params for transcoding, so the file will be transcoded and played in real time
 - if you play live streams, you should skip `#input` param, because it is already in real time
 - you can stop active playback by calling the API with the empty `src` parameter
 - you will see one active producer and one active consumer in go2rtc WebUI info page during streaming
@@ -779,10 +787,10 @@ POST http://localhost:1984/api/streams?dst=camera1&src=ffmpeg:http://example.com
 You can publish any stream to streaming services (YouTube, Telegram, etc.) via RTMP/RTMPS. Important:
 
 - Supported codecs: H264 for video and AAC for audio
-- Pixel format should be `yuv420p`, for cameras with `yuvj420p` format you SHOULD use [transcoding](#source-ffmpeg)
+- AAC audio is required for YouTube; videos without audio will not work
 - You don't need to enable [RTMP module](#module-rtmp) listening for this task
 
-You can use API:
+You can use the API:
 
 ```
 POST http://localhost:1984/api/streams?src=camera1&dst=rtmps://...
@@ -792,16 +800,19 @@ Or config file:
 
 ```yaml
 publish:
-  # publish stream "tplink_tapo" to Telegram
-  tplink_tapo: rtmps://xxx-x.rtmp.t.me/s/xxxxxxxxxx:xxxxxxxxxxxxxxxxxxxxxx
-  # publish stream "other_camera" to Telegram and YouTube
-  other_camera:
+  # publish stream "video_audio_transcode" to Telegram
+  video_audio_transcode:
     - rtmps://xxx-x.rtmp.t.me/s/xxxxxxxxxx:xxxxxxxxxxxxxxxxxxxxxx
-    - rtmps://xxx.rtmp.youtube.com/live2/xxxx-xxxx-xxxx-xxxx-xxxx
+  # publish stream "audio_transcode" to Telegram and YouTube
+  audio_transcode:
+    - rtmps://xxx-x.rtmp.t.me/s/xxxxxxxxxx:xxxxxxxxxxxxxxxxxxxxxx
+    - rtmp://xxx.rtmp.youtube.com/live2/xxxx-xxxx-xxxx-xxxx-xxxx
 
 streams:
-  # for TP-Link cameras it's important to use transcoding because of wrong pixel format
-  tplink_tapo: ffmpeg:rtsp://user:pass@192.168.1.123/stream1#video=h264#hardware#audio=aac
+  video_audio_transcode:
+    - ffmpeg:rtsp://user:pass@192.168.1.123/stream1#video=h264#hardware#audio=aac
+  audio_transcode:
+    - ffmpeg:rtsp://user:pass@192.168.1.123/stream1#video=copy#audio=aac
 ```
 
 - **Telegram Desktop App** > Any public or private channel or group (where you admin) > Live stream > Start with... > Start streaming.
@@ -811,7 +822,7 @@ streams:
 
 The HTTP API is the main part for interacting with the application. Default address: `http://localhost:1984/`.
 
-**Important!** go2rtc passes requests from localhost and from unix socket without HTTP authorisation, even if you have it configured! It is your responsibility to set up secure external access to API. If not properly configured, an attacker can gain access to your cameras and even your server.
+**Important!** go2rtc passes requests from localhost and from Unix sockets without HTTP authorisation, even if you have it configured! It is your responsibility to set up secure external access to the API. If not properly configured, an attacker can gain access to your cameras and even your server.
 
 [API description](https://github.com/AlexxIT/go2rtc/tree/master/api).
 
@@ -819,7 +830,7 @@ The HTTP API is the main part for interacting with the application. Default addr
 
 - you can disable HTTP API with `listen: ""` and use, for example, only RTSP client/server protocol
 - you can enable HTTP API only on localhost with `listen: "127.0.0.1:1984"` setting
-- you can change API `base_path` and host go2rtc on your main app webserver suburl
+- you can change the API `base_path` and host go2rtc on your main app webserver suburl
 - all files from `static_dir` hosted on root path: `/`
 - you can use raw TLS cert/key content or path to files
 
@@ -828,7 +839,7 @@ api:
   listen: ":1984"    # default ":1984", HTTP API port ("" - disabled)
   username: "admin"  # default "", Basic auth for WebUI
   password: "pass"   # default "", Basic auth for WebUI
-  base_path: "/rtc"  # default "", API prefix for serve on suburl (/api => /rtc/api)
+  base_path: "/rtc"  # default "", API prefix for serving on suburl (/api => /rtc/api)
   static_dir: "www"  # default "", folder for static files (custom web interface)
   origin: "*"        # default "", allow CORS requests (only * supported)
   tls_listen: ":443" # default "", enable HTTPS server
@@ -852,7 +863,7 @@ api:
 
 You can get any stream as RTSP-stream: `rtsp://192.168.1.123:8554/{stream_name}`
 
-You can enable external password protection for your RTSP streams. Password protection always disabled for localhost calls (ex. FFmpeg or Hass on same server).
+You can enable external password protection for your RTSP streams. Password protection is always disabled for localhost calls (ex. FFmpeg or Hass on the same server).
 
 ```yaml
 rtsp:
@@ -877,7 +888,7 @@ Read more about [codecs filters](#codecs-filters).
 
 You can get any stream as RTMP-stream: `rtmp://192.168.1.123/{stream_name}`. Only H264/AAC codecs supported right now.
 
-[Incoming stream](#incoming-sources) in RTMP-format tested only with [OBS Studio](https://obsproject.com/) and Dahua camera. Different FFmpeg versions has differnt problems with this format. 
+[Incoming stream](#incoming-sources) in RTMP format tested only with [OBS Studio](https://obsproject.com/) and a Dahua camera. Different FFmpeg versions have different problems with this format. 
 
 ```yaml
 rtmp:
@@ -886,12 +897,12 @@ rtmp:
 
 ### Module: WebRTC
 
-In most cases [WebRTC](https://en.wikipedia.org/wiki/WebRTC) uses direct peer-to-peer connection from your browser to go2rtc and sends media data via UDP.
+In most cases, [WebRTC](https://en.wikipedia.org/wiki/WebRTC) uses a direct peer-to-peer connection from your browser to go2rtc and sends media data via UDP.
 It **can't pass** media data through your Nginx or Cloudflare or [Nabu Casa](https://www.nabucasa.com/) HTTP TCP connection!
-It can automatically detects your external IP via public [STUN](https://en.wikipedia.org/wiki/STUN) server.
-It can establish a external direct connection via [UDP hole punching](https://en.wikipedia.org/wiki/UDP_hole_punching) technology even if you not open your server to the World.
+It can automatically detect your external IP via a public [STUN](https://en.wikipedia.org/wiki/STUN) server.
+It can establish an external direct connection via [UDP hole punching](https://en.wikipedia.org/wiki/UDP_hole_punching) technology even if you do not open your server to the World.
 
-But about 10-20% of users may need to configure additional settings for external access if **mobile phone** or **go2rtc server** behing [Symmetric NAT](https://tomchen.github.io/symmetric-nat-test/).
+But about 10-20% of users may need to configure additional settings for external access if **mobile phone** or **go2rtc server** is behind [Symmetric NAT](https://tomchen.github.io/symmetric-nat-test/).
 
 - by default, WebRTC uses both TCP and UDP on port 8555 for connections
 - you can use this port for external access
@@ -904,25 +915,25 @@ webrtc:
 
 **Static public IP**
 
-- forward the port 8555 on your router (you can use same 8555 port or any other as external port)
-- add your external IP-address and external port to YAML config
+- forward the port 8555 on your router (you can use the same 8555 port or any other as external port)
+- add your external IP address and external port to the YAML config
 
 ```yaml
 webrtc:
   candidates:
-    - 216.58.210.174:8555  # if you have static public IP-address
+    - 216.58.210.174:8555  # if you have a static public IP address
 ```
 
 **Dynamic public IP**
 
-- forward the port 8555 on your router (you can use same 8555 port or any other as the external port)
+- forward the port 8555 on your router (you can use the same 8555 port or any other as the external port)
 - add `stun` word and external port to YAML config
-  - go2rtc automatically detects your external address with STUN-server
+  - go2rtc automatically detects your external address with STUN server
 
 ```yaml
 webrtc:
   candidates:
-    - stun:8555  # if you have dynamic public IP-address
+    - stun:8555  # if you have a dynamic public IP address
 ```
 
 **Private IP**
@@ -936,7 +947,7 @@ ngrok:
 
 **Hard tech way 1. Own TCP-tunnel**
 
-If you have personal [VPS](https://en.wikipedia.org/wiki/Virtual_private_server), you can create TCP-tunnel and setup in the same way as "Static public IP". But use your VPS IP-address in YAML config.
+If you have a personal [VPS](https://en.wikipedia.org/wiki/Virtual_private_server), you can create a TCP tunnel and setup in the same way as "Static public IP". But use your VPS IP address in the YAML config.
 
 **Hard tech way 2. Using TURN-server**
 
@@ -962,7 +973,7 @@ HomeKit module can work in two modes:
 
 **Important**
 
-- HomeKit cameras supports only H264 video and OPUS audio
+- HomeKit cameras support only H264 video and OPUS audio
 
 **Minimal config**
 
@@ -1009,17 +1020,17 @@ homekit:
 
 *[New in v1.3.0](https://github.com/AlexxIT/go2rtc/releases/tag/v1.3.0)*
 
-This module support:
+This module supports:
 
 - Share any local stream via [WebTorrent](https://webtorrent.io/) technology
 - Get any [incoming stream](#incoming-browser) from PC or mobile via [WebTorrent](https://webtorrent.io/) technology
 - Get any remote [go2rtc source](#source-webtorrent) via [WebTorrent](https://webtorrent.io/) technology
 
-Securely and free. You do not need to open a public access to the go2rtc server. But in some cases (Symmetric NAT) you may need to set up external access to [WebRTC module](#module-webrtc).
+Securely and freely. You do not need to open a public access to the go2rtc server. But in some cases (Symmetric NAT), you may need to set up external access to [WebRTC module](#module-webrtc).
 
-To generate sharing link or incoming link - goto go2rtc WebUI (stream links page). This link is **temporary** and will stop working after go2rtc is restarted!
+To generate a sharing link or incoming link, go to the go2rtc WebUI (stream links page). This link is **temporary** and will stop working after go2rtc is restarted!
 
-You can create permanent external links in go2rtc config:
+You can create permanent external links in the go2rtc config:
 
 ```yaml
 webtorrent:
@@ -1031,22 +1042,22 @@ webtorrent:
 
 Link example: https://alexxit.github.io/go2rtc/#share=02SNtgjKXY&pwd=wznEQqznxW&media=video+audio
 
-TODO: article how it works...
+TODO: article on how it works...
 
 ### Module: ngrok
 
-With ngrok integration you can get external access to your streams in situations when you have Internet with private IP-address.
+With ngrok integration, you can get external access to your streams in situations when you have Internet with a private IP address.
 
-- ngrok is pre-installed for **Docker** and **Hass Add-on** users
+- ngrok is pre-installed for **Docker** and **Hass add-on** users
 - you may need external access for two different things:
-  - WebRTC stream, so you need tunnel WebRTC TCP port (ex. 8555)
-  - go2rtc web interface, so you need tunnel API HTTP port (ex. 1984)
-- ngrok support authorization for your web interface
+  - WebRTC stream, so you need a tunnel WebRTC TCP port (ex. 8555)
+  - go2rtc web interface, so you need a tunnel API HTTP port (ex. 1984)
+- ngrok supports authorization for your web interface
 - ngrok automatically adds HTTPS to your web interface
 
 The ngrok free subscription has the following limitations:
 
-- You can reserve a free domain for serving the web interface, but the TCP address you get will always be random and change with each restart of the ngrok agent (not a problem for webrtc stream)
+- You can reserve a free domain for serving the web interface, but the TCP address you get will always be random and change with each restart of the ngrok agent (not a problem for WebRTC stream)
 - You can forward multiple ports from a single agent, but you can only run one ngrok agent on the free plan
 
 go2rtc will automatically get your external TCP address (if you enable it in ngrok config) and use it with WebRTC connection (if you enable it in webrtc config).
@@ -1064,7 +1075,7 @@ ngrok:
 
 **Tunnel for WebRTC and Web interface**
 
-You need to create `ngrok.yaml` config file and add it to go2rtc config:
+You need to create `ngrok.yaml` config file and add it to the go2rtc config:
 
 ```yaml
 ngrok:
@@ -1078,12 +1089,12 @@ version: "2"
 authtoken: eW91IHNoYWxsIG5vdCBwYXNzCnlvdSBzaGFsbCBub3QgcGFzcw
 tunnels:
   api:
-    addr: 1984  # use the same port as in go2rtc config
+    addr: 1984  # use the same port as in the go2rtc config
     proto: http
     basic_auth:
       - admin:password  # you can set login/pass for your web interface
   webrtc:
-    addr: 8555  # use the same port as in go2rtc config
+    addr: 8555  # use the same port as in the go2rtc config
     proto: tcp
 ```
 
@@ -1091,9 +1102,9 @@ See the [ngrok agent documentation](https://ngrok.com/docs/agent/config/) for mo
 
 ### Module: Hass
 
-The best and easiest way to use go2rtc inside the Home Assistant is to install the custom integration [WebRTC Camera](#go2rtc-home-assistant-integration) and custom lovelace card.
+The best and easiest way to use go2rtc inside Home Assistant is to install the custom integration [WebRTC Camera](#go2rtc-home-assistant-integration) and custom Lovelace card.
 
-But go2rtc is also compatible and can be used with [RTSPtoWebRTC](https://www.home-assistant.io/integrations/rtsp_to_webrtc/) built-in integration.
+But go2rtc is also compatible and can be used with the [RTSPtoWebRTC](https://www.home-assistant.io/integrations/rtsp_to_webrtc/) built-in integration.
 
 You have several options on how to add a camera to Home Assistant:
 
@@ -1111,10 +1122,10 @@ You have several options on how to watch the stream from the cameras in Home Ass
    - Install any [go2rtc](#fast-start)
    - Hass > Settings > Integrations > Add Integration > [RTSPtoWebRTC](https://my.home-assistant.io/redirect/config_flow_start/?domain=rtsp_to_webrtc) > `http://127.0.0.1:1984/`
    - RTSPtoWebRTC > Configure > STUN server: `stun.l.google.com:19302`
-   - Use Picture Entity or Picture Glance lovelace card
+   - Use Picture Entity or Picture Glance Lovelace card
 3. `Camera Entity` or `Camera URL` => [WebRTC Camera](https://github.com/AlexxIT/WebRTC) => Technology: `WebRTC/MSE/MP4/MJPEG`, codecs: `H264/H265/AAC/PCMU/PCMA/OPUS`, best latency, best compatibility.
    - Install and add [WebRTC Camera](https://github.com/AlexxIT/WebRTC) custom integration
-   - Use WebRTC Camera custom lovelace card
+   - Use WebRTC Camera custom Lovelace card
 
 You can add camera `entity_id` to [go2rtc config](#configuration) if you need transcoding:
 
@@ -1123,7 +1134,7 @@ streams:
   "camera.hall": ffmpeg:{input}#video=copy#audio=opus
 ```
 
-**PS.** Default Home Assistant lovelace cards don't support 2-way audio. You can use 2-way audio from [Add-on Web UI](https://my.home-assistant.io/redirect/supervisor_addon/?addon=a889bffc_go2rtc&repository_url=https%3A%2F%2Fgithub.com%2FAlexxIT%2Fhassio-addons). But you need use HTTPS to access the microphone. This is a browser restriction and cannot be avoided.
+**PS.** Default Home Assistant lovelace cards don't support two-way audio. You can use 2-way audio from [Add-on Web UI](https://my.home-assistant.io/redirect/supervisor_addon/?addon=a889bffc_go2rtc&repository_url=https%3A%2F%2Fgithub.com%2FAlexxIT%2Fhassio-addons), but you need to use HTTPS to access the microphone. This is a browser restriction and cannot be avoided.
 
 **PS.** There is also another nice card with go2rtc support - [Frigate Lovelace Card](https://github.com/dermotduffy/frigate-hass-card).
 
@@ -1133,7 +1144,7 @@ Provides several features:
 
 1. MSE stream (fMP4 over WebSocket)
 2. Camera snapshots in MP4 format (single frame), can be sent to [Telegram](https://github.com/AlexxIT/go2rtc/wiki/Snapshot-to-Telegram)
-3. HTTP progressive streaming (MP4 file stream) - bad format for streaming because of high start delay. This format doesn't work in all Safari browsers, but go2rtc will automatically redirect it to HLS/fMP4 it this case.
+3. HTTP progressive streaming (MP4 file stream) - bad format for streaming because of high start delay. This format doesn't work in all Safari browsers, but go2rtc will automatically redirect it to HLS/fMP4 in this case.
 
 API examples:
 
@@ -1167,13 +1178,13 @@ Read more about [codecs filters](#codecs-filters).
 
 ### Module: MJPEG
 
-**Important.** For stream as MJPEG format, your source MUST contain the MJPEG codec. If your stream has a MJPEG codec - you can receive **MJPEG stream** or **JPEG snapshots** via API.
+**Important.** For stream in MJPEG format, your source MUST contain the MJPEG codec. If your stream has an MJPEG codec, you can receive **MJPEG stream** or **JPEG snapshots** via API.
 
 You can receive an MJPEG stream in several ways:
 
 - some cameras support MJPEG codec inside [RTSP stream](#source-rtsp) (ex. second stream for Dahua cameras)
-- some cameras has HTTP link with [MJPEG stream](#source-http)
-- some cameras has HTTP link with snapshots - go2rtc can convert them to [MJPEG stream](#source-http)
+- some cameras have an HTTP link with [MJPEG stream](#source-http)
+- some cameras have an HTTP link with snapshots - go2rtc can convert them to [MJPEG stream](#source-http)
 - you can convert H264/H265 stream from your camera via [FFmpeg integraion](#source-ffmpeg)
 
 With this example, your stream will have both H264 and MJPEG codecs:
@@ -1214,7 +1225,7 @@ log:
 
 ## Security
 
-By default `go2rtc` starts the Web interface on port `1984` and RTSP on port `8554`, as well as use port `8555` for WebRTC connections. The three ports are accessible from your local network. So anyone on your local network can watch video from your cameras without authorization. The same rule applies to the Home Assistant Add-on.
+By default, `go2rtc` starts the Web interface on port `1984` and RTSP on port `8554`, as well as uses port `8555` for WebRTC connections. The three ports are accessible from your local network. So anyone on your local network can watch video from your cameras without authorization. The same rule applies to the Home Assistant Add-on.
 
 This is not a problem if you trust your local network as much as I do. But you can change this behaviour with a `go2rtc.yaml` config:
 
@@ -1230,13 +1241,13 @@ webrtc:
 ```
 
 - local access to RTSP is not a problem for [FFmpeg](#source-ffmpeg) integration, because it runs locally on your server
-- local access to API is not a problem for [Home Assistant Add-on](#go2rtc-home-assistant-add-on), because Hass runs locally on same server and Add-on Web UI protected with Hass authorization ([Ingress feature](https://www.home-assistant.io/blog/2019/04/15/hassio-ingress/))
-- external access to WebRTC TCP port is not a problem, because it used only for transmit encrypted media data
-  - anyway you need to open this port to your local network and to the Internet in order for WebRTC to work
+- local access to API is not a problem for the [Home Assistant add-on](#go2rtc-home-assistant-add-on), because Hass runs locally on the same server, and the add-on web UI is protected with Hass authorization ([Ingress feature](https://www.home-assistant.io/blog/2019/04/15/hassio-ingress/))
+- external access to WebRTC TCP port is not a problem, because it is used only for transmitting encrypted media data
+  - anyway you need to open this port to your local network and to the Internet for WebRTC to work
 
-If you need Web interface protection without Home Assistant Add-on - you need to use reverse proxy, like [Nginx](https://nginx.org/), [Caddy](https://caddyserver.com/), [ngrok](https://ngrok.com/), etc.
+If you need web interface protection without the Home Assistant add-on, you need to use a reverse proxy, like [Nginx](https://nginx.org/), [Caddy](https://caddyserver.com/), [ngrok](https://ngrok.com/), etc.
 
-PS. Additionally WebRTC will try to use the 8555 UDP port for transmit encrypted media. It works without problems on the local network. And sometimes also works for external access, even if you haven't opened this port on your router ([read more](https://en.wikipedia.org/wiki/UDP_hole_punching)). But for stable external WebRTC access, you need to open the 8555 port on your router for both TCP and UDP.
+PS. Additionally, WebRTC will try to use the 8555 UDP port to transmit encrypted media. It works without problems on the local network, and sometimes also works for external access, even if you haven't opened this port on your router ([read more](https://en.wikipedia.org/wiki/UDP_hole_punching)). But for stable external WebRTC access, you need to open the 8555 port on your router for both TCP and UDP.
 
 ## Codecs filters
 
@@ -1259,11 +1270,11 @@ Some examples:
 - `http://192.168.1.123:1984/api/stream.m3u8?src=camera1&mp4` - HLS stream with MP4 compatible codecs (HLS/fMP4)
 - `http://192.168.1.123:1984/api/stream.m3u8?src=camera1&mp4=flac` - HLS stream with PCMA/PCMU/PCM audio support (HLS/fMP4), won't work on old devices
 - `http://192.168.1.123:1984/api/stream.mp4?src=camera1&mp4=flac` - MP4 file with PCMA/PCMU/PCM audio support, won't work on old devices (ex. iOS 12)
-- `http://192.168.1.123:1984/api/stream.mp4?src=camera1&mp4=all` - MP4 file with non standard audio codecs, won't work on some players
+- `http://192.168.1.123:1984/api/stream.mp4?src=camera1&mp4=all` - MP4 file with non-standard audio codecs, won't work on some players
 
 ## Codecs madness
 
-`AVC/H.264` video can be played almost anywhere. But `HEVC/H.265` has a lot of limitations in supporting with different devices and browsers. It's all about patents and money, you can't do anything about it.
+`AVC/H.264` video can be played almost anywhere. But `HEVC/H.265` has many limitations in supporting different devices and browsers. It's all about patents and money; you can't do anything about it.
 
 | Device                                                                   | WebRTC                                  | MSE                                     | HTTP*                                        | HLS                         |
 |--------------------------------------------------------------------------|-----------------------------------------|-----------------------------------------|----------------------------------------------|-----------------------------|
@@ -1276,7 +1287,7 @@ Some examples:
 
 [1]: https://apps.apple.com/app/home-assistant/id1099568401
 
-`HTTP*` - HTTP Progressive Streaming, not related with [Progressive download](https://en.wikipedia.org/wiki/Progressive_download), because the file has no size and no end 
+`HTTP*` - HTTP Progressive Streaming, not related to [progressive download](https://en.wikipedia.org/wiki/Progressive_download), because the file has no size and no end 
 
 - Chrome H265: [read this](https://chromestatus.com/feature/5186511939567616) and [read this](https://github.com/StaZhu/enable-chromium-hevc-hardware-decoding)
 - Edge H265: [read this](https://www.reddit.com/r/MicrosoftEdge/comments/v9iw8k/enable_hevc_support_in_edge/)
@@ -1287,7 +1298,7 @@ Some examples:
 
 - Go2rtc support [automatic repack](#built-in-transcoding) `PCMA/PCMU/PCM` codecs to `FLAC` for MSE/MP4/HLS so they will work almost anywhere
 - **WebRTC** audio codecs: `PCMU/8000`, `PCMA/8000`, `OPUS/48000/2`
-- `OPUS` and `MP3` inside **MP4** is part of the standard, but some players do not support them anyway (especially Apple)
+- `OPUS` and `MP3` inside **MP4** are part of the standard, but some players do not support them anyway (especially Apple)
 
 **Apple devices**
 
@@ -1309,7 +1320,7 @@ Some examples:
 
 There are no plans to embed complex transcoding algorithms inside go2rtc. [FFmpeg source](#source-ffmpeg) does a great job with this. Including [hardware acceleration](https://github.com/AlexxIT/go2rtc/wiki/Hardware-acceleration) support.
 
-But go2rtc has some simple algorithms. They are turned on automatically, you do not need to set them up additionally.
+But go2rtc has some simple algorithms. They are turned on automatically; you do not need to set them up additionally.
 
 **PCM for MSE/MP4/HLS**
 
@@ -1321,7 +1332,7 @@ PCMA/PCMU => PCM => FLAC => MSE/MP4/HLS
 
 **Resample PCMA/PCMU for WebRTC**
 
-By default WebRTC support only `PCMA/8000` and `PCMU/8000`. But go2rtc can automatically resample PCMA and PCMU codec with with a different sample rate. Also go2rtc can transcode `PCM` codec to `PCMA/8000`, so WebRTC can play it:
+By default WebRTC supports only `PCMA/8000` and `PCMU/8000`. But go2rtc can automatically resample PCMA and PCMU codecs with a different sample rate. Also, go2rtc can transcode `PCM` codec to `PCMA/8000`, so WebRTC can play it:
 
 ```
 PCM/xxx => PCMA/8000 => WebRTC
@@ -1331,24 +1342,24 @@ PCMU/xxx => PCMU/8000 => WebRTC
 
 **Important**
 
-- FLAC codec not supported in a RTSP stream. If you using Frigate or Hass for recording MP4 files with PCMA/PCMU/PCM audio - you should setup transcoding to AAC codec.
-- PCMA and PCMU are VERY low quality codecs. Them support only 256! different sounds. Use them only when you have no other options.
+- FLAC codec not supported in an RTSP stream. If you are using Frigate or Hass for recording MP4 files with PCMA/PCMU/PCM audio, you should set up transcoding to the AAC codec.
+- PCMA and PCMU are VERY low-quality codecs. They support only 256! different sounds. Use them only when you have no other options.
 
 ## Codecs negotiation
 
 For example, you want to watch RTSP-stream from [Dahua IPC-K42](https://www.dahuasecurity.com/fr/products/All-Products/Network-Cameras/Wireless-Series/Wi-Fi-Series/4MP/IPC-K42) camera in your Chrome browser.
 
-- this camera support 2-way audio standard **ONVIF Profile T**
-- this camera support codecs **H264, H265** for send video, and you select `H264` in camera settings
-- this camera support codecs **AAC, PCMU, PCMA** for send audio (from mic), and you select `AAC/16000` in camera settings
-- this camera support codecs **AAC, PCMU, PCMA** for receive audio (to speaker), you don't need to select them
-- your browser support codecs **H264, VP8, VP9, AV1** for receive video, you don't need to select them
-- your browser support codecs **OPUS, PCMU, PCMA** for send and receive audio, you don't need to select them
-- you can't get camera audio directly, because its audio codecs doesn't match with your browser codecs
-  - so you decide to use transcoding via FFmpeg and add this setting to config YAML file
+- this camera supports two-way audio standard **ONVIF Profile T**
+- this camera supports codecs **H264, H265** for send video, and you select `H264` in camera settings
+- this camera supports codecs **AAC, PCMU, PCMA** for sending audio (from mic), and you select `AAC/16000` in camera settings
+- this camera supports codecs **AAC, PCMU, PCMA** for receiving audio (to speaker), you don't need to select them
+- your browser supports codecs **H264, VP8, VP9, AV1** for receiving video, you don't need to select them
+- your browser supports codecs **OPUS, PCMU, PCMA** for sending and receiving audio, you don't need to select them
+- you can't get camera audio directly, because its audio codecs don't match with your browser codecs
+  - so you decide to use transcoding via FFmpeg and add this setting to the config YAML file
   - you have chosen `OPUS/48000/2` codec, because it is higher quality than the `PCMU/8000` or `PCMA/8000`
 
-Now you have stream with two sources - **RTSP and FFmpeg**:
+Now you have a stream with two sources - **RTSP and FFmpeg**:
 
 ```yaml
 streams:
@@ -1357,22 +1368,23 @@ streams:
     - ffmpeg:rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0#audio=opus
 ```
 
-**go2rtc** automatically match codecs for you browser and all your stream sources. This called **multi-source 2-way codecs negotiation**. And this is one of the main features of this app.
+**go2rtc** automatically matches codecs for your browser and all your stream sources. This is called **multi-source two-way codec negotiation**. And this is one of the main features of this app.
 
 ![](assets/codecs.svg)
 
-**PS.** You can select `PCMU` or `PCMA` codec in camera setting and don't use transcoding at all. Or you can select `AAC` codec for main stream and `PCMU` codec for second stream and add both RTSP to YAML config, this also will work fine.
+**PS.** You can select `PCMU` or `PCMA` codec in camera settings and not use transcoding at all. Or you can select `AAC` codec for main stream and `PCMU` codec for second stream and add both RTSP to YAML config, this also will work fine.
 
 ## Projects using go2rtc
 
-- [Frigate 12+](https://frigate.video/) - open source NVR built around real-time AI object detection
+- [Frigate](https://frigate.video/) 0.12+ - open-source NVR built around real-time AI object detection
 - [Frigate Lovelace Card](https://github.com/dermotduffy/frigate-hass-card) - custom card for Home Assistant
-- [OpenIPC](https://github.com/OpenIPC/firmware/tree/master/general/package/go2rtc) - Alternative IP Camera firmware from an open community
-- [wz_mini_hacks](https://github.com/gtxaspec/wz_mini_hacks) - Custom firmware for Wyze cameras
-- [EufyP2PStream](https://github.com/oischinger/eufyp2pstream) - A small project that provides a Video/Audio Stream from Eufy cameras that don't directly support RTSP
-- [ioBroker.euSec](https://github.com/bropat/ioBroker.eusec) - [ioBroker](https://www.iobroker.net/) adapter for control Eufy security devices
-- [MMM-go2rtc](https://github.com/Anonym-tsk/MMM-go2rtc) - MagicMirror² Module
-- [ring-mqtt](https://github.com/tsightler/ring-mqtt) - Ring devices to MQTT Bridge
+- [OpenIPC](https://github.com/OpenIPC/firmware/tree/master/general/package/go2rtc) - alternative IP camera firmware from an open community
+- [wz_mini_hacks](https://github.com/gtxaspec/wz_mini_hacks) - custom firmware for Wyze cameras
+- [EufyP2PStream](https://github.com/oischinger/eufyp2pstream) - a small project that provides a video/audio stream from Eufy cameras that don't directly support RTSP
+- [ioBroker.euSec](https://github.com/bropat/ioBroker.eusec) - [ioBroker](https://www.iobroker.net/) adapter for controlling Eufy security devices
+- [MMM-go2rtc](https://github.com/Anonym-tsk/MMM-go2rtc) - MagicMirror² module
+- [ring-mqtt](https://github.com/tsightler/ring-mqtt) - Ring-to-MQTT bridge
+- [lightNVR](https://github.com/opensensor/lightNVR)
 
 **Distributions**
 
@@ -1380,20 +1392,20 @@ streams:
 - [Arch User Repository](https://linux-packages.com/aur/package/go2rtc)
 - [Gentoo](https://github.com/inode64/inode64-overlay/tree/main/media-video/go2rtc)
 - [NixOS](https://search.nixos.org/packages?query=go2rtc)
-- [Proxmox Helper Scripts](https://tteck.github.io/Proxmox/)
+- [Proxmox Helper Scripts](https://github.com/community-scripts/ProxmoxVE/)
 - [QNAP](https://www.myqnap.org/product/go2rtc/)
 - [Synology NAS](https://synocommunity.com/package/go2rtc)
 - [Unraid](https://unraid.net/community/apps?q=go2rtc)
 
-## Cameras experience
+## Camera experience
 
 - [Dahua](https://www.dahuasecurity.com/) - reference implementation streaming protocols, a lot of settings, high stream quality, multiple streaming clients
-- [EZVIZ](https://www.ezviz.com/) - awful RTSP protocol realisation, many bugs in SDP
+- [EZVIZ](https://www.ezviz.com/) - awful RTSP protocol implementation, many bugs in SDP
 - [Hikvision](https://www.hikvision.com/) - a lot of proprietary streaming technologies
-- [Reolink](https://reolink.com/) - some models has awful unusable RTSP realisation and not best RTMP alternative (I recommend that you contact Reolink support for new firmware), few settings
-- [Sonoff](https://sonoff.tech/) - very low stream quality, no settings, not best protocol implementation
+- [Reolink](https://reolink.com/) - some models have an awful, unusable RTSP implementation and not the best RTMP alternative (I recommend that you contact Reolink support for new firmware), few settings
+- [Sonoff](https://sonoff.tech/) - very low stream quality, no settings, not the best protocol implementation
 - [TP-Link](https://www.tp-link.com/) - few streaming clients, packet loss?
-- Chinese cheap noname cameras, Wyze Cams, Xiaomi cameras with hacks (usual has `/live/ch00_1` in RTSP URL) - awful but usable RTSP protocol realisation, low stream quality, few settings, packet loss?
+- Chinese cheap noname cameras, Wyze Cams, Xiaomi cameras with hacks (usually have `/live/ch00_1` in RTSP URL) - awful but usable RTSP protocol implementation, low stream quality, few settings, packet loss?
 
 ## TIPS
 
@@ -1410,22 +1422,22 @@ streams:
 
 **Q. What's the difference between go2rtc, WebRTC Camera and RTSPtoWebRTC?**
 
-**go2rtc** is a new version of the server-side [WebRTC Camera](https://github.com/AlexxIT/WebRTC) integration, completely rewritten from scratch, with a number of fixes and a huge number of new features. It is compatible with native Home Assistant [RTSPtoWebRTC](https://www.home-assistant.io/integrations/rtsp_to_webrtc/) integration. So you [can use](#module-hass) default lovelace Picture Entity or Picture Glance.
+**go2rtc** is a new version of the server-side [WebRTC Camera](https://github.com/AlexxIT/WebRTC) integration, completely rewritten from scratch, with a number of fixes and a huge number of new features. It is compatible with native Home Assistant [RTSPtoWebRTC](https://www.home-assistant.io/integrations/rtsp_to_webrtc/) integration. So you [can use](#module-hass) default Lovelace Picture Entity or Picture Glance.
 
-**Q. Should I use go2rtc addon or WebRTC Camera integration?**
+**Q. Should I use the go2rtc add-on or WebRTC Camera integration?**
 
-**go2rtc** is more than just viewing your stream online with WebRTC/MSE/HLS/etc. You can use it all the time for your various tasks. But every time the Hass is rebooted - all integrations are also rebooted. So your streams may be interrupted if you use them in additional tasks.
+**go2rtc** is more than just viewing your stream online with WebRTC/MSE/HLS/etc. You can use it all the time for your various tasks. But every time Hass is rebooted, all integrations are also rebooted. So your streams may be interrupted if you use them in additional tasks.
 
-Basic users can use **WebRTC Camera** integration. Advanced users can use go2rtc addon or Frigate 12+ addon.
+Basic users can use the **WebRTC Camera** integration. Advanced users can use the go2rtc add-on or the Frigate 0.12+ add-on.
 
 **Q. Which RTSP link should I use inside Hass?**
 
-You can use direct link to your cameras there (as you always do). **go2rtc** support zero-config feature. You may leave `streams` config section empty. And your streams will be created on the fly on first start from Hass. And your cameras will have multiple connections. Some from Hass directly and one from **go2rtc**.
+You can use a direct link to your cameras there (as you always do). **go2rtc** supports zero-config feature. You may leave `streams` config section empty. And your streams will be created on the fly on first start from Hass. And your cameras will have multiple connections. Some from Hass directly and one from **go2rtc**.
 
-Also you can specify your streams in **go2rtc** [config file](#configuration) and use RTSP links to this addon. With additional features: multi-source [codecs negotiation](#codecs-negotiation) or FFmpeg [transcoding](#source-ffmpeg) for unsupported codecs. Or use them as source for Frigate. And your cameras will have one connection from **go2rtc**. And **go2rtc** will have multiple connection - some from Hass via RTSP protocol, some from your browser via WebRTC/MSE/HLS protocols.
+Also, you can specify your streams in **go2rtc** [config file](#configuration) and use RTSP links to this add-on with additional features: multi-source [codecs negotiation](#codecs-negotiation) or FFmpeg [transcoding](#source-ffmpeg) for unsupported codecs. Or use them as a source for Frigate. And your cameras will have one connection from **go2rtc**. And **go2rtc** will have multiple connections - some from Hass via RTSP protocol, some from your browser via WebRTC/MSE/HLS protocols.
 
-Use any config what you like.
+Use any config that you like.
 
-**Q. What about lovelace card with support 2-way audio?**
+**Q. What about Lovelace card with support for two-way audio?**
 
-At this moment I am focused on improving stability and adding new features to **go2rtc**. Maybe someone could write such a card themselves. It's not difficult, I have [some sketches](https://github.com/AlexxIT/go2rtc/blob/master/www/webrtc.html).
+At this moment, I am focused on improving stability and adding new features to **go2rtc**. Maybe someone could write such a card themselves. It's not difficult, I have [some sketches](https://github.com/AlexxIT/go2rtc/blob/master/www/webrtc.html).

docker/Dockerfile

@@ -2,11 +2,7 @@
 
 # 0. Prepare images
 ARG PYTHON_VERSION="3.11"
-ARG GO_VERSION="1.21"
-ARG NGROK_VERSION="3"
-
-FROM python:${PYTHON_VERSION}-alpine AS base
-FROM ngrok/ngrok:${NGROK_VERSION}-alpine AS ngrok
+ARG GO_VERSION="1.24"
 
 
 # 1. Build go2rtc binary
@@ -20,6 +16,8 @@ ENV GOARCH=${TARGETARCH}
 
 WORKDIR /build
 
+RUN apk add git
+
 # Cache dependencies
 COPY go.mod go.sum ./
 RUN --mount=type=cache,target=/root/.cache/go-build go mod download
@@ -28,21 +26,14 @@ COPY . .
 RUN --mount=type=cache,target=/root/.cache/go-build CGO_ENABLED=0 go build -ldflags "-s -w" -trimpath
 
 
-# 2. Collect all files
-FROM scratch AS rootfs
-
-COPY --from=build /build/go2rtc /usr/local/bin/
-COPY --from=ngrok /bin/ngrok /usr/local/bin/
-
-
-# 3. Final image
-FROM base
+# 2. Final image
+FROM python:${PYTHON_VERSION}-alpine AS base
 
 # Install ffmpeg, tini (for signal handling),
 # and other common tools for the echo source.
 # alsa-plugins-pulse for ALSA support (+0MB)
 # font-droid for FFmpeg drawtext filter (+2MB)
-RUN apk add --no-cache tini ffmpeg bash curl jq alsa-plugins-pulse font-droid
+RUN apk add --no-cache tini ffmpeg ffplay bash curl jq alsa-plugins-pulse font-droid
 
 # Hardware Acceleration for Intel CPU (+50MB)
 ARG TARGETARCH
@@ -54,7 +45,7 @@ RUN if [ "${TARGETARCH}" = "amd64" ]; then apk add --no-cache libva-intel-driver
 # Hardware: AMD and NVidia VDPAU (not sure about this)
 # RUN libva-vdpau-driver mesa-vdpau-gallium (+150MB total)
 
-COPY --from=rootfs / /
+COPY --from=build /build/go2rtc /usr/local/bin/
 
 ENTRYPOINT ["/sbin/tini", "--"]
 VOLUME /config

docker/README.md

@@ -0,0 +1,50 @@
+## Versions
+
+- `alexxit/go2rtc:latest` - latest release based on `alpine` (`amd64`, `386`, `arm/v6`, `arm/v7`, `arm64`) with support hardware transcoding for Intel iGPU and Raspberry
+- `alexxit/go2rtc:latest-hardware` - latest release based on `debian 13` (`amd64`) with support hardware transcoding for Intel iGPU, AMD GPU and NVidia GPU
+- `alexxit/go2rtc:latest-rockchip` - latest release based on `debian 12` (`arm64`) with support hardware transcoding for Rockchip RK35xx
+- `alexxit/go2rtc:master` - latest unstable version based on `alpine`
+- `alexxit/go2rtc:master-hardware` - latest unstable version based on `debian 13` (`amd64`)
+- `alexxit/go2rtc:master-rockchip` - latest unstable version based on `debian 12` (`arm64`)
+
+## Docker compose
+
+```yaml
+services:
+  go2rtc:
+    image: alexxit/go2rtc
+    network_mode: host       # important for WebRTC, HomeKit, UDP cameras
+    privileged: true         # only for FFmpeg hardware transcoding
+    restart: unless-stopped  # autorestart on fail or config change from WebUI
+    environment:
+      - TZ=Atlantic/Bermuda  # timezone in logs
+    volumes:
+      - "~/go2rtc:/config"   # folder for go2rtc.yaml file (edit from WebUI)
+```
+
+## Basic Deployment
+
+```bash
+docker run -d \
+  --name go2rtc \
+  --network host \
+  --privileged \
+  --restart unless-stopped \
+  -e TZ=Atlantic/Bermuda \
+  -v ~/go2rtc:/config \
+  alexxit/go2rtc
+```
+
+## Deployment with GPU Acceleration
+
+```bash
+docker run -d \
+  --name go2rtc \
+  --network host \
+  --privileged \
+  --restart unless-stopped \
+  -e TZ=Atlantic/Bermuda \
+  --gpus all \
+  -v ~/go2rtc:/config \
+  alexxit/go2rtc:latest-hardware
+```

docker/hardware.Dockerfile

@@ -4,16 +4,11 @@
 # only debian 13 (trixie) has latest ffmpeg
 # https://packages.debian.org/trixie/ffmpeg
 ARG DEBIAN_VERSION="trixie-slim"
-ARG GO_VERSION="1.21-bookworm"
-ARG NGROK_VERSION="3"
-
-FROM debian:${DEBIAN_VERSION} AS base
-FROM golang:${GO_VERSION} AS go
-FROM ngrok/ngrok:${NGROK_VERSION} AS ngrok
+ARG GO_VERSION="1.24-bookworm"
 
 
 # 1. Build go2rtc binary
-FROM --platform=$BUILDPLATFORM go AS build
+FROM --platform=$BUILDPLATFORM golang:${GO_VERSION} AS build
 ARG TARGETPLATFORM
 ARG TARGETOS
 ARG TARGETARCH
@@ -31,34 +26,28 @@ COPY . .
 RUN --mount=type=cache,target=/root/.cache/go-build CGO_ENABLED=0 go build -ldflags "-s -w" -trimpath
 
 
-# 2. Collect all files
-FROM scratch AS rootfs
+# 2. Final image
+FROM debian:${DEBIAN_VERSION}
 
-COPY --link --from=build /build/go2rtc /usr/local/bin/
-COPY --link --from=ngrok /bin/ngrok /usr/local/bin/
-
-# 3. Final image
-FROM base
 # Prepare apt for buildkit cache
 RUN rm -f /etc/apt/apt.conf.d/docker-clean \
   && echo 'Binary::apt::APT::Keep-Downloaded-Packages "true";' >/etc/apt/apt.conf.d/keep-cache
-# Install ffmpeg, bash (for run.sh), tini (for signal handling),
+
+# Install ffmpeg, tini (for signal handling),
 # and other common tools for the echo source.
 # non-free for Intel QSV support (not used by go2rtc, just for tests)
 # mesa-va-drivers for AMD APU
 # libasound2-plugins for ALSA support
 RUN --mount=type=cache,target=/var/cache/apt,sharing=locked --mount=type=cache,target=/var/lib/apt,sharing=locked \
     echo 'deb http://deb.debian.org/debian trixie non-free' > /etc/apt/sources.list.d/debian-non-free.list && \
-    apt-get -y update && apt-get -y install tini ffmpeg \
+    apt-get -y update && apt-get -y install ffmpeg tini \
         python3 curl jq \
         intel-media-va-driver-non-free \
         mesa-va-drivers \
         libasound2-plugins && \
     apt-get clean && rm -rf /var/lib/apt/lists/*
 
-COPY --link --from=rootfs / /
-
-
+COPY --from=build /build/go2rtc /usr/local/bin/
 
 ENTRYPOINT ["/usr/bin/tini", "--"]
 VOLUME /config

docker/rockchip.Dockerfile

@@ -0,0 +1,50 @@
+# syntax=docker/dockerfile:labs
+
+# 0. Prepare images
+ARG PYTHON_VERSION="3.13-slim-bookworm"
+ARG GO_VERSION="1.24-bookworm"
+
+
+# 1. Build go2rtc binary
+FROM --platform=$BUILDPLATFORM golang:${GO_VERSION} AS build
+ARG TARGETPLATFORM
+ARG TARGETOS
+ARG TARGETARCH
+
+ENV GOOS=${TARGETOS}
+ENV GOARCH=${TARGETARCH}
+
+WORKDIR /build
+
+# Cache dependencies
+COPY go.mod go.sum ./
+RUN --mount=type=cache,target=/root/.cache/go-build go mod download
+
+COPY . .
+RUN --mount=type=cache,target=/root/.cache/go-build CGO_ENABLED=0 go build -ldflags "-s -w" -trimpath
+
+
+# 2. Final image
+FROM python:${PYTHON_VERSION}
+
+# Prepare apt for buildkit cache
+RUN rm -f /etc/apt/apt.conf.d/docker-clean \
+  && echo 'Binary::apt::APT::Keep-Downloaded-Packages "true";' >/etc/apt/apt.conf.d/keep-cache
+
+# Install ffmpeg, tini (for signal handling),
+# and other common tools for the echo source.
+# libasound2-plugins for ALSA support
+RUN --mount=type=cache,target=/var/cache/apt,sharing=locked --mount=type=cache,target=/var/lib/apt,sharing=locked \
+    apt-get -y update && apt-get -y install tini \
+        curl jq \
+        libasound2-plugins && \
+    apt-get clean && rm -rf /var/lib/apt/lists/*
+
+COPY --from=build /build/go2rtc /usr/local/bin/
+ADD --chmod=755 https://github.com/MarcA711/Rockchip-FFmpeg-Builds/releases/download/6.1-8-no_extra_dump/ffmpeg /usr/local/bin
+
+ENTRYPOINT ["/usr/bin/tini", "--"]
+VOLUME /config
+WORKDIR /config
+
+CMD ["go2rtc", "-config", "/config/go2rtc.yaml"]

examples/go2rtc_mjpeg/main.go

@@ -0,0 +1,26 @@
+package main
+
+import (
+	"github.com/AlexxIT/go2rtc/internal/api"
+	"github.com/AlexxIT/go2rtc/internal/api/ws"
+	"github.com/AlexxIT/go2rtc/internal/app"
+	"github.com/AlexxIT/go2rtc/internal/ffmpeg"
+	"github.com/AlexxIT/go2rtc/internal/mjpeg"
+	"github.com/AlexxIT/go2rtc/internal/streams"
+	"github.com/AlexxIT/go2rtc/internal/v4l2"
+	"github.com/AlexxIT/go2rtc/pkg/shell"
+)
+
+func main() {
+	app.Init()
+	streams.Init()
+
+	api.Init()
+	ws.Init()
+
+	ffmpeg.Init()
+	mjpeg.Init()
+	v4l2.Init()
+
+	shell.RunUntilSignal()
+}

examples/homekit_info/main.go

@@ -54,7 +54,7 @@ var chars = map[string]string{
 	"21C": "Third Party Camera Active",
 	"21D": "Camera Operating Mode Indicator",
 	"11B": "Night Vision",
-	"129": "Supported Data Stream Transport Configuration",
+	//"129": "Supported Data Stream Transport Configuration",
 	"37":  "Version",
 	"131": "Setup Data Stream Transport",
 	"130": "Supported Data Stream Transport Configuration",

examples/onvif_client/README.md

@@ -0,0 +1,5 @@
+## Example
+
+```shell
+go run examples/onvif_client/main.go http://admin:password@192.168.10.90 GetAudioEncoderConfigurations
+```

examples/onvif_client/main.go

@@ -0,0 +1,75 @@
+package main
+
+import (
+	"log"
+	"net/url"
+	"os"
+
+	"github.com/AlexxIT/go2rtc/pkg/onvif"
+)
+
+func main() {
+	var rawURL = os.Args[1]
+	var operation = os.Args[2]
+	var token string
+	if len(os.Args) > 3 {
+		token = os.Args[3]
+	}
+
+	client, err := onvif.NewClient(rawURL)
+	if err != nil {
+		log.Panic(err)
+	}
+
+	var b []byte
+
+	switch operation {
+	case onvif.ServiceGetServiceCapabilities:
+		b, err = client.MediaRequest(operation)
+	case onvif.DeviceGetCapabilities,
+		onvif.DeviceGetDeviceInformation,
+		onvif.DeviceGetDiscoveryMode,
+		onvif.DeviceGetDNS,
+		onvif.DeviceGetHostname,
+		onvif.DeviceGetNetworkDefaultGateway,
+		onvif.DeviceGetNetworkInterfaces,
+		onvif.DeviceGetNetworkProtocols,
+		onvif.DeviceGetNTP,
+		onvif.DeviceGetScopes,
+		onvif.DeviceGetServices,
+		onvif.DeviceGetSystemDateAndTime,
+		onvif.DeviceSystemReboot:
+		b, err = client.DeviceRequest(operation)
+	case onvif.MediaGetProfiles,
+		onvif.MediaGetVideoEncoderConfigurations,
+		onvif.MediaGetVideoSources,
+		onvif.MediaGetVideoSourceConfigurations,
+		onvif.MediaGetAudioEncoderConfigurations,
+		onvif.MediaGetAudioSources,
+		onvif.MediaGetAudioSourceConfigurations:
+		b, err = client.MediaRequest(operation)
+	case onvif.MediaGetProfile:
+		b, err = client.GetProfile(token)
+	case onvif.MediaGetVideoSourceConfiguration:
+		b, err = client.GetVideoSourceConfiguration(token)
+	case onvif.MediaGetStreamUri:
+		b, err = client.GetStreamUri(token)
+	case onvif.MediaGetSnapshotUri:
+		b, err = client.GetSnapshotUri(token)
+	default:
+		log.Printf("unknown action\n")
+	}
+
+	if err != nil {
+		log.Printf("%s\n", err)
+	}
+
+	u, err := url.Parse(rawURL)
+	if err != nil {
+		log.Fatal(err)
+	}
+
+	if err = os.WriteFile(u.Hostname()+"_"+operation+".xml", b, 0644); err != nil {
+		log.Printf("%s\n", err)
+	}
+}

examples/rtsp_client/main.go

@@ -0,0 +1,39 @@
+package main
+
+import (
+	"log"
+	"os"
+
+	"github.com/AlexxIT/go2rtc/pkg/core"
+	"github.com/AlexxIT/go2rtc/pkg/rtsp"
+	"github.com/AlexxIT/go2rtc/pkg/shell"
+)
+
+func main() {
+	client := rtsp.NewClient(os.Args[1])
+	if err := client.Dial(); err != nil {
+		log.Panic(err)
+	}
+
+	client.Medias = []*core.Media{
+		{
+			Kind:      core.KindAudio,
+			Direction: core.DirectionRecvonly,
+			Codecs: []*core.Codec{
+				{Name: core.CodecPCMU, ClockRate: 8000},
+			},
+			ID: "streamid=0",
+		},
+	}
+	if err := client.Announce(); err != nil {
+		log.Panic(err)
+	}
+	if _, err := client.SetupMedia(client.Medias[0]); err != nil {
+		log.Panic(err)
+	}
+	if err := client.Record(); err != nil {
+		log.Panic(err)
+	}
+
+	shell.RunUntilSignal()
+}

go.mod

@@ -1,48 +1,49 @@
 module github.com/AlexxIT/go2rtc
 
-go 1.21
+go 1.23.0
 
 require (
 	github.com/asticode/go-astits v1.13.0
-	github.com/expr-lang/expr v1.16.5
-	github.com/gorilla/websocket v1.5.1
-	github.com/miekg/dns v1.1.59
-	github.com/pion/ice/v2 v2.3.24
-	github.com/pion/interceptor v0.1.29
-	github.com/pion/rtcp v1.2.14
-	github.com/pion/rtp v1.8.6
-	github.com/pion/sdp/v3 v3.0.9
-	github.com/pion/srtp/v2 v2.0.18
-	github.com/pion/stun v0.6.1
-	github.com/pion/webrtc/v3 v3.2.40
-	github.com/rs/zerolog v1.32.0
+	github.com/expr-lang/expr v1.17.5
+	github.com/google/uuid v1.6.0
+	github.com/gorilla/websocket v1.5.3
+	github.com/mattn/go-isatty v0.0.20
+	github.com/miekg/dns v1.1.66
+	github.com/pion/ice/v4 v4.0.10
+	github.com/pion/interceptor v0.1.40
+	github.com/pion/rtcp v1.2.15
+	github.com/pion/rtp v1.8.20
+	github.com/pion/sdp/v3 v3.0.14
+	github.com/pion/srtp/v3 v3.0.6
+	github.com/pion/stun/v3 v3.0.0
+	github.com/pion/webrtc/v4 v4.1.3
+	github.com/rs/zerolog v1.34.0
 	github.com/sigurn/crc16 v0.0.0-20240131213347-83fcde1e29d1
 	github.com/sigurn/crc8 v0.0.0-20220107193325-2243fe600f9f
-	github.com/stretchr/testify v1.9.0
+	github.com/stretchr/testify v1.10.0
 	github.com/tadglines/go-pkgs v0.0.0-20210623144937-b983b20f54f9
-	golang.org/x/crypto v0.23.0
+	golang.org/x/crypto v0.39.0
 	gopkg.in/yaml.v3 v3.0.1
 )
 
 require (
-	github.com/asticode/go-astikit v0.30.0 // indirect
+	github.com/asticode/go-astikit v0.56.0 // indirect
 	github.com/davecgh/go-spew v1.1.1 // indirect
-	github.com/google/uuid v1.6.0 // indirect
-	github.com/kr/pretty v0.2.1 // indirect
-	github.com/mattn/go-colorable v0.1.13 // indirect
-	github.com/mattn/go-isatty v0.0.20 // indirect
-	github.com/pion/datachannel v1.5.6 // indirect
-	github.com/pion/dtls/v2 v2.2.11 // indirect
-	github.com/pion/logging v0.2.2 // indirect
-	github.com/pion/mdns v0.0.12 // indirect
+	github.com/kr/pretty v0.3.1 // indirect
+	github.com/mattn/go-colorable v0.1.14 // indirect
+	github.com/pion/datachannel v1.5.10 // indirect
+	github.com/pion/dtls/v3 v3.0.6 // indirect
+	github.com/pion/logging v0.2.4 // indirect
+	github.com/pion/mdns/v2 v2.0.7 // indirect
 	github.com/pion/randutil v0.1.0 // indirect
-	github.com/pion/sctp v1.8.16 // indirect
-	github.com/pion/transport/v2 v2.2.5 // indirect
-	github.com/pion/turn/v2 v2.1.6 // indirect
+	github.com/pion/sctp v1.8.39 // indirect
+	github.com/pion/transport/v3 v3.0.7 // indirect
+	github.com/pion/turn/v4 v4.0.2 // indirect
 	github.com/pmezard/go-difflib v1.0.0 // indirect
-	golang.org/x/mod v0.17.0 // indirect
-	golang.org/x/net v0.25.0 // indirect
-	golang.org/x/sync v0.7.0 // indirect
-	golang.org/x/sys v0.20.0 // indirect
-	golang.org/x/tools v0.20.0 // indirect
+	github.com/wlynxg/anet v0.0.5 // indirect
+	golang.org/x/mod v0.25.0 // indirect
+	golang.org/x/net v0.41.0 // indirect
+	golang.org/x/sync v0.15.0 // indirect
+	golang.org/x/sys v0.33.0 // indirect
+	golang.org/x/tools v0.34.0 // indirect
 )

go.sum

@@ -1,194 +1,140 @@
-github.com/asticode/go-astikit v0.30.0 h1:DkBkRQRIxYcknlaU7W7ksNfn4gMFsB0tqMJflxkRsZA=
 github.com/asticode/go-astikit v0.30.0/go.mod h1:h4ly7idim1tNhaVkdVBeXQZEE3L0xblP7fCWbgwipF0=
+github.com/asticode/go-astikit v0.54.0 h1:uq9eurgisdkYwJU9vSWIQaPH4MH0cac82sQH00kmSNQ=
+github.com/asticode/go-astikit v0.54.0/go.mod h1:fV43j20UZYfXzP9oBn33udkvCvDvCDhzjVqoLFuuYZE=
+github.com/asticode/go-astikit v0.56.0 h1:DmD2p7YnvxiPdF0h+dRmos3bsejNEXbycENsY5JfBqw=
+github.com/asticode/go-astikit v0.56.0/go.mod h1:fV43j20UZYfXzP9oBn33udkvCvDvCDhzjVqoLFuuYZE=
 github.com/asticode/go-astits v1.13.0 h1:XOgkaadfZODnyZRR5Y0/DWkA9vrkLLPLeeOvDwfKZ1c=
 github.com/asticode/go-astits v1.13.0/go.mod h1:QSHmknZ51pf6KJdHKZHJTLlMegIrhega3LPWz3ND/iI=
 github.com/coreos/go-systemd/v22 v22.5.0/go.mod h1:Y58oyj3AT4RCenI/lSvhwexgC+NSVTIJ3seZv2GcEnc=
+github.com/creack/pty v1.1.9/go.mod h1:oKZEueFk5CKHvIhNR5MUki03XCEU+Q6VDXinZuGJ33E=
 github.com/davecgh/go-spew v1.1.0/go.mod h1:J7Y8YcW2NihsgmVo/mv3lAwl/skON4iLHjSsI+c5H38=
 github.com/davecgh/go-spew v1.1.1 h1:vj9j/u1bqnvCEfJOwUhtlOARqs3+rkHYY13jYWTU97c=
 github.com/davecgh/go-spew v1.1.1/go.mod h1:J7Y8YcW2NihsgmVo/mv3lAwl/skON4iLHjSsI+c5H38=
-github.com/expr-lang/expr v1.16.5 h1:m2hvtguFeVaVNTHj8L7BoAyt7O0PAIBaSVbjdHgRXMs=
-github.com/expr-lang/expr v1.16.5/go.mod h1:uCkhfG+x7fcZ5A5sXHKuQ07jGZRl6J0FCAaf2k4PtVQ=
+github.com/expr-lang/expr v1.17.2 h1:o0A99O/Px+/DTjEnQiodAgOIK9PPxL8DtXhBRKC+Iso=
+github.com/expr-lang/expr v1.17.2/go.mod h1:8/vRC7+7HBzESEqt5kKpYXxrxkr31SaO8r40VO/1IT4=
+github.com/expr-lang/expr v1.17.5 h1:i1WrMvcdLF249nSNlpQZN1S6NXuW9WaOfF5tPi3aw3k=
+github.com/expr-lang/expr v1.17.5/go.mod h1:8/vRC7+7HBzESEqt5kKpYXxrxkr31SaO8r40VO/1IT4=
 github.com/godbus/dbus/v5 v5.0.4/go.mod h1:xhWf0FNVPg57R7Z0UbKHbJfkEywrmjJnf7w5xrFpKfA=
-github.com/google/uuid v1.3.1/go.mod h1:TIyPZe4MgqvfeYDBFedMoGGpEw/LqOeaOT+nhxU+yHo=
 github.com/google/uuid v1.6.0 h1:NIvaJDMOsjHA8n1jAhLSgzrAzy1Hgr+hNrb57e+94F0=
 github.com/google/uuid v1.6.0/go.mod h1:TIyPZe4MgqvfeYDBFedMoGGpEw/LqOeaOT+nhxU+yHo=
-github.com/gorilla/websocket v1.5.1 h1:gmztn0JnHVt9JZquRuzLw3g4wouNVzKL15iLr/zn/QY=
-github.com/gorilla/websocket v1.5.1/go.mod h1:x3kM2JMyaluk02fnUJpQuwD2dCS5NDG2ZHL0uE0tcaY=
-github.com/kr/pretty v0.1.0/go.mod h1:dAy3ld7l9f0ibDNOQOHHMYYIIbhfbHSm3C4ZsoJORNo=
-github.com/kr/pretty v0.2.1 h1:Fmg33tUaq4/8ym9TJN1x7sLJnHVwhP33CNkpYV/7rwI=
-github.com/kr/pretty v0.2.1/go.mod h1:ipq/a2n7PKx3OHsz4KJII5eveXtPO4qwEXGdVfWzfnI=
-github.com/kr/pty v1.1.1/go.mod h1:pFQYn66WHrOpPYNljwOMqo10TkYh1fy3cYio2l3bCsQ=
-github.com/kr/text v0.1.0 h1:45sCR5RtlFHMR4UwH9sdQ5TC8v0qDQCHnXt+kaKSTVE=
-github.com/kr/text v0.1.0/go.mod h1:4Jbv+DJW3UT/LiOwJeYQe1efqtUx/iVham/4vfdArNI=
-github.com/mattn/go-colorable v0.1.13 h1:fFA4WZxdEF4tXPZVKMLwD8oUnCTTo08duU7wxecdEvA=
+github.com/gorilla/websocket v1.5.3 h1:saDtZ6Pbx/0u+bgYQ3q96pZgCzfhKXGPqt7kZ72aNNg=
+github.com/gorilla/websocket v1.5.3/go.mod h1:YR8l580nyteQvAITg2hZ9XVh4b55+EU/adAjf1fMHhE=
+github.com/kr/pretty v0.3.1 h1:flRD4NNwYAUpkphVc1HcthR4KEIFJ65n8Mw5qdRn3LE=
+github.com/kr/pretty v0.3.1/go.mod h1:hoEshYVHaxMs3cyo3Yncou5ZscifuDolrwPKZanG3xk=
+github.com/kr/text v0.2.0 h1:5Nx0Ya0ZqY2ygV366QzturHI13Jq95ApcVaJBhpS+AY=
+github.com/kr/text v0.2.0/go.mod h1:eLer722TekiGuMkidMxC/pM04lWEeraHUUmBw8l2grE=
 github.com/mattn/go-colorable v0.1.13/go.mod h1:7S9/ev0klgBDR4GtXTXX8a3vIGJpMovkB8vQcUbaXHg=
+github.com/mattn/go-colorable v0.1.14 h1:9A9LHSqF/7dyVVX6g0U9cwm9pG3kP9gSzcuIPHPsaIE=
+github.com/mattn/go-colorable v0.1.14/go.mod h1:6LmQG8QLFO4G5z1gPvYEzlUgJ2wF+stgPZH1UqBm1s8=
 github.com/mattn/go-isatty v0.0.16/go.mod h1:kYGgaQfpe5nmfYZH+SKPsOc2e4SrIfOl2e/yFXSvRLM=
 github.com/mattn/go-isatty v0.0.19/go.mod h1:W+V8PltTTMOvKvAeJH7IuucS94S2C6jfK/D7dTCTo3Y=
 github.com/mattn/go-isatty v0.0.20 h1:xfD0iDuEKnDkl03q4limB+vH+GxLEtL/jb4xVJSWWEY=
 github.com/mattn/go-isatty v0.0.20/go.mod h1:W+V8PltTTMOvKvAeJH7IuucS94S2C6jfK/D7dTCTo3Y=
-github.com/miekg/dns v1.1.59 h1:C9EXc/UToRwKLhK5wKU/I4QVsBUc8kE6MkHBkeypWZs=
-github.com/miekg/dns v1.1.59/go.mod h1:nZpewl5p6IvctfgrckopVx2OlSEHPRO/U4SYkRklrEk=
-github.com/pion/datachannel v1.5.6 h1:1IxKJntfSlYkpUj8LlYRSWpYiTTC02nUrOE8T3DqGeg=
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 gopkg.in/yaml.v2 v2.2.2/go.mod h1:hI93XBmqTisBFMUTm0b8Fm+jr3Dg1NNxqwp+5A1VGuI=
-gopkg.in/yaml.v3 v3.0.0-20200313102051-9f266ea9e77c/go.mod h1:K4uyk7z7BCEPqu6E+C64Yfv1cQ7kz7rIZviUmN+EgEM=
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 gopkg.in/yaml.v3 v3.0.1/go.mod h1:K4uyk7z7BCEPqu6E+C64Yfv1cQ7kz7rIZviUmN+EgEM=

internal/alsa/alsa.go

@@ -0,0 +1,7 @@
+//go:build !(linux && (386 || amd64 || arm || arm64 || mipsle))
+
+package alsa
+
+func Init() {
+	// not supported
+}

internal/alsa/alsa_linux.go

@@ -0,0 +1,83 @@
+//go:build linux && (386 || amd64 || arm || arm64 || mipsle)
+
+package alsa
+
+import (
+	"fmt"
+	"net/http"
+	"os"
+	"strconv"
+	"strings"
+
+	"github.com/AlexxIT/go2rtc/internal/api"
+	"github.com/AlexxIT/go2rtc/internal/streams"
+	"github.com/AlexxIT/go2rtc/pkg/alsa"
+	"github.com/AlexxIT/go2rtc/pkg/alsa/device"
+)
+
+func Init() {
+	streams.HandleFunc("alsa", alsa.Open)
+
+	api.HandleFunc("api/alsa", apiAlsa)
+}
+
+func apiAlsa(w http.ResponseWriter, r *http.Request) {
+	files, err := os.ReadDir("/dev/snd/")
+	if err != nil {
+		return
+	}
+
+	var sources []*api.Source
+
+	for _, file := range files {
+		if !strings.HasPrefix(file.Name(), "pcm") {
+			continue
+		}
+
+		path := "/dev/snd/" + file.Name()
+
+		dev, err := device.Open(path)
+		if err != nil {
+			continue
+		}
+
+		info, err := dev.Info()
+		if err == nil {
+			formats := formatsToString(dev.ListFormats())
+			r1, r2 := dev.RangeRates()
+			c1, c2 := dev.RangeChannels()
+			source := &api.Source{
+				Name: info.ID,
+				Info: fmt.Sprintf("Formats: %s, Rates: %d-%d, Channels: %d-%d", formats, r1, r2, c1, c2),
+				URL:  "alsa:device?audio=" + path,
+			}
+			if !strings.Contains(source.Name, info.Name) {
+				source.Name += ", " + info.Name
+			}
+			sources = append(sources, source)
+		}
+
+		_ = dev.Close()
+	}
+
+	api.ResponseSources(w, sources)
+}
+
+func formatsToString(formats []byte) string {
+	var s string
+	for i, format := range formats {
+		if i > 0 {
+			s += " "
+		}
+		switch format {
+		case 2:
+			s += "s16le"
+		case 10:
+			s += "s32le"
+		default:
+			s += strconv.Itoa(int(format))
+		}
+
+	}
+	return s
+}

internal/api/api.go

@@ -69,6 +69,8 @@ func Init() {
 	}
 
 	if cfg.Mod.Listen != "" {
+		_, port, _ := net.SplitHostPort(cfg.Mod.Listen)
+		Port, _ = strconv.Atoi(port)
 		go listen("tcp", cfg.Mod.Listen)
 	}
 
@@ -92,10 +94,6 @@ func listen(network, address string) {
 
 	log.Info().Str("addr", address).Msg("[api] listen")
 
-	if network == "tcp" {
-		Port = ln.Addr().(*net.TCPAddr).Port
-	}
-
 	server := http.Server{
 		Handler:           Handler,
 		ReadHeaderTimeout: 5 * time.Second, // Example: Set to 5 seconds

internal/api/static.go

@@ -1,8 +1,9 @@
 package api
 
 import (
-	"github.com/AlexxIT/go2rtc/www"
 	"net/http"
+
+	"github.com/AlexxIT/go2rtc/www"
 )
 
 func initStatic(staticDir string) {

internal/api/ws/ws.go

@@ -1,6 +1,7 @@
 package ws
 
 import (
+	"encoding/json"
 	"io"
 	"net/http"
 	"net/url"
@@ -10,6 +11,7 @@ import (
 
 	"github.com/AlexxIT/go2rtc/internal/api"
 	"github.com/AlexxIT/go2rtc/internal/app"
+	"github.com/AlexxIT/go2rtc/pkg/core"
 	"github.com/gorilla/websocket"
 	"github.com/rs/zerolog"
 )
@@ -38,20 +40,19 @@ type Message struct {
 	Value any    `json:"value,omitempty"`
 }
 
-func (m *Message) String() string {
+func (m *Message) String() (value string) {
 	if s, ok := m.Value.(string); ok {
 		return s
 	}
-	return ""
+	return
 }
 
-func (m *Message) GetString(key string) string {
-	if v, ok := m.Value.(map[string]any); ok {
-		if s, ok := v[key].(string); ok {
-			return s
-		}
+func (m *Message) Unmarshal(v any) error {
+	b, err := json.Marshal(m.Value)
+	if err != nil {
+		return err
 	}
-	return ""
+	return json.Unmarshal(b, v)
 }
 
 type WSHandler func(tr *Transport, msg *Message) error
@@ -132,7 +133,8 @@ func apiWS(w http.ResponseWriter, r *http.Request) {
 		if handler := wsHandlers[msg.Type]; handler != nil {
 			go func() {
 				if err = handler(tr, msg); err != nil {
-					tr.Write(&Message{Type: "error", Value: msg.Type + ": " + err.Error()})
+					errMsg := core.StripUserinfo(err.Error())
+					tr.Write(&Message{Type: "error", Value: msg.Type + ": " + errMsg})
 				}
 			}()
 		}

internal/app/README.md

@@ -1,6 +1,10 @@
 - By default go2rtc will search config file `go2rtc.yaml` in current work directory
-- go2rtc support multiple config files
-- go2rtc support inline config as `YAML`, `JSON` or `key=value` format from command line
+- go2rtc support multiple config files:
+  - `go2rtc -c config1.yaml -c config2.yaml -c config3.yaml` 
+- go2rtc support inline config as multiple formats from command line:
+  - **YAML**: `go2rtc -c '{log: {format: text}}'`
+  - **JSON**: `go2rtc -c '{"log":{"format":"text"}}'`
+  - **key=value**: `go2rtc -c log.format=text`
 - Every next config will overwrite previous (but only defined params)
 
 ```
@@ -15,21 +19,30 @@ go2rtc -c log.format=text -c /config/go2rtc.yaml -c rtsp.listen='' -c /usr/local
 
 ## Environment variables
 
-Also go2rtc support templates for using environment variables in any part of config:
+There is support for loading external variables into the config. First, they will be attempted to be loaded from [credential files](https://systemd.io/CREDENTIALS). If `CREDENTIALS_DIRECTORY` is not set, then the key will be loaded from an environment variable. If no environment variable is set, then the string will be left as-is.
 
 ```yaml
 streams:
   camera1: rtsp://rtsp:${CAMERA_PASSWORD}@192.168.1.123/av_stream/ch0
 
-  ${LOGS:}  # empty default value
-
 rtsp:
-  username: ${RTSP_USER:admin}   # "admin" if env "RTSP_USER" not set
-  password: ${RTSP_PASS:secret}  # "secret" if env "RTSP_PASS" not set
+  username: ${RTSP_USER:admin}   # "admin" if "RTSP_USER" not set
+  password: ${RTSP_PASS:secret}  # "secret" if "RTSP_PASS" not set
+```
+
+## JSON Schema
+
+Editors like [GoLand](https://www.jetbrains.com/go/) and [VS Code](https://code.visualstudio.com/) supports autocomplete and syntax validation.
+
+```yaml
+# yaml-language-server: $schema=https://raw.githubusercontent.com/AlexxIT/go2rtc/master/website/schema.json
 ```
 
 ## Defaults
 
+- Default values may change in updates
+- FFmpeg module has many presets, they are not listed here because they may also change in updates
+
 ```yaml
 api:
   listen: ":1984"
@@ -38,7 +51,10 @@ ffmpeg:
   bin: "ffmpeg"
 
 log:
+  format: "color"
   level: "info"
+  output: "stdout"
+  time: "UNIXMS"
 
 rtsp:
   listen: ":8554"
@@ -51,4 +67,4 @@ webrtc:
   listen: ":8555/tcp"
   ice_servers:
     - urls: [ "stun:stun.l.google.com:19302" ]
-```
+```

internal/app/app.go

@@ -1,28 +1,20 @@
 package app
 
 import (
-	"errors"
 	"flag"
 	"fmt"
 	"os"
 	"os/exec"
-	"path/filepath"
 	"runtime"
 	"runtime/debug"
-	"strings"
-
-	"github.com/AlexxIT/go2rtc/pkg/shell"
-	"github.com/AlexxIT/go2rtc/pkg/yaml"
-	"github.com/rs/zerolog/log"
 )
 
-var Version = "1.9.2"
-var UserAgent = "go2rtc/" + Version
-
-var ConfigPath string
-var Info = map[string]any{
-	"version": Version,
-}
+var (
+	Version    string
+	UserAgent  string
+	ConfigPath string
+	Info       = make(map[string]any)
+)
 
 const usage = `Usage of go2rtc:
 
@@ -32,12 +24,12 @@ const usage = `Usage of go2rtc:
 `
 
 func Init() {
-	var confs Config
+	var config flagConfig
 	var daemon bool
 	var version bool
 
-	flag.Var(&confs, "config", "")
-	flag.Var(&confs, "c", "")
+	flag.Var(&config, "config", "")
+	flag.Var(&config, "c", "")
 	flag.BoolVar(&daemon, "daemon", false, "")
 	flag.BoolVar(&daemon, "d", false, "")
 	flag.BoolVar(&version, "version", false, "")
@@ -53,126 +45,39 @@ func Init() {
 		os.Exit(0)
 	}
 
-	if daemon {
+	if daemon && os.Getppid() != 1 {
 		if runtime.GOOS == "windows" {
-			fmt.Println("Daemon not supported on Windows")
+			fmt.Println("Daemon mode is not supported on Windows")
 			os.Exit(1)
 		}
 
-		args := os.Args[1:]
-		for i, arg := range args {
-			if arg == "-daemon" {
-				args[i] = ""
-			}
-		}
 		// Re-run the program in background and exit
-		cmd := exec.Command(os.Args[0], args...)
+		cmd := exec.Command(os.Args[0], os.Args[1:]...)
 		if err := cmd.Start(); err != nil {
-			log.Fatal().Err(err).Send()
+			fmt.Println("Failed to start daemon:", err)
+			os.Exit(1)
 		}
 		fmt.Println("Running in daemon mode with PID:", cmd.Process.Pid)
 		os.Exit(0)
 	}
 
-	if confs == nil {
-		confs = []string{"go2rtc.yaml"}
-	}
-
-	for _, conf := range confs {
-		if len(conf) == 0 {
-			continue
-		}
-		if conf[0] == '{' {
-			// config as raw YAML or JSON
-			configs = append(configs, []byte(conf))
-		} else if data := parseConfString(conf); data != nil {
-			configs = append(configs, data)
-		} else {
-			// config as file
-			if ConfigPath == "" {
-				ConfigPath = conf
-			}
-
-			if data, _ = os.ReadFile(conf); data == nil {
-				continue
-			}
-
-			data = []byte(shell.ReplaceEnvVars(string(data)))
-			configs = append(configs, data)
-		}
-	}
-
-	if ConfigPath != "" {
-		if !filepath.IsAbs(ConfigPath) {
-			if cwd, err := os.Getwd(); err == nil {
-				ConfigPath = filepath.Join(cwd, ConfigPath)
-			}
-		}
-		Info["config_path"] = ConfigPath
-	}
+	UserAgent = "go2rtc/" + Version
 
+	Info["version"] = Version
 	Info["revision"] = revision
 
-	var cfg struct {
-		Mod map[string]string `yaml:"log"`
-	}
-
-	LoadConfig(&cfg)
-
-	log.Logger = NewLogger(cfg.Mod["format"], cfg.Mod["level"])
-
-	modules = cfg.Mod
+	initConfig(config)
+	initLogger()
 
 	platform := fmt.Sprintf("%s/%s", runtime.GOOS, runtime.GOARCH)
-	log.Info().Str("version", Version).Str("platform", platform).Str("revision", revision).Msg("go2rtc")
-	log.Debug().Str("version", runtime.Version()).Str("vcs.time", vcsTime).Msg("build")
+	Logger.Info().Str("version", Version).Str("platform", platform).Str("revision", revision).Msg("go2rtc")
+	Logger.Debug().Str("version", runtime.Version()).Str("vcs.time", vcsTime).Msg("build")
 
 	if ConfigPath != "" {
-		log.Info().Str("path", ConfigPath).Msg("config")
-	}
-
-	migrateStore()
-}
-
-func LoadConfig(v any) {
-	for _, data := range configs {
-		if err := yaml.Unmarshal(data, v); err != nil {
-			log.Warn().Err(err).Msg("[app] read config")
-		}
+		Logger.Info().Str("path", ConfigPath).Msg("config")
 	}
 }
 
-func PatchConfig(key string, value any, path ...string) error {
-	if ConfigPath == "" {
-		return errors.New("config file disabled")
-	}
-
-	// empty config is OK
-	b, _ := os.ReadFile(ConfigPath)
-
-	b, err := yaml.Patch(b, key, value, path...)
-	if err != nil {
-		return err
-	}
-
-	return os.WriteFile(ConfigPath, b, 0644)
-}
-
-// internal
-
-type Config []string
-
-func (c *Config) String() string {
-	return strings.Join(*c, " ")
-}
-
-func (c *Config) Set(value string) error {
-	*c = append(*c, value)
-	return nil
-}
-
-var configs [][]byte
-
 func readRevisionTime() (revision, vcsTime string) {
 	if info, ok := debug.ReadBuildInfo(); ok {
 		for _, setting := range info.Settings {
@@ -194,25 +99,3 @@ func readRevisionTime() (revision, vcsTime string) {
 	}
 	return
 }
-
-func parseConfString(s string) []byte {
-	i := strings.IndexByte(s, '=')
-	if i < 0 {
-		return nil
-	}
-
-	items := strings.Split(s[:i], ".")
-	if len(items) < 2 {
-		return nil
-	}
-
-	// `log.level=trace` => `{log: {level: trace}}`
-	var pre string
-	var suf = s[i+1:]
-	for _, item := range items {
-		pre += "{" + item + ": "
-		suf += "}"
-	}
-
-	return []byte(pre + suf)
-}

internal/app/config.go

@@ -0,0 +1,115 @@
+package app
+
+import (
+	"errors"
+	"os"
+	"path/filepath"
+	"strings"
+	"sync"
+
+	"github.com/AlexxIT/go2rtc/pkg/shell"
+	"github.com/AlexxIT/go2rtc/pkg/yaml"
+)
+
+func LoadConfig(v any) {
+	for _, data := range configs {
+		if err := yaml.Unmarshal(data, v); err != nil {
+			Logger.Warn().Err(err).Send()
+		}
+	}
+}
+
+var configMu sync.Mutex
+
+func PatchConfig(path []string, value any) error {
+	if ConfigPath == "" {
+		return errors.New("config file disabled")
+	}
+
+	configMu.Lock()
+	defer configMu.Unlock()
+
+	// empty config is OK
+	b, _ := os.ReadFile(ConfigPath)
+
+	b, err := yaml.Patch(b, path, value)
+	if err != nil {
+		return err
+	}
+
+	return os.WriteFile(ConfigPath, b, 0644)
+}
+
+type flagConfig []string
+
+func (c *flagConfig) String() string {
+	return strings.Join(*c, " ")
+}
+
+func (c *flagConfig) Set(value string) error {
+	*c = append(*c, value)
+	return nil
+}
+
+var configs [][]byte
+
+func initConfig(confs flagConfig) {
+	if confs == nil {
+		confs = []string{"go2rtc.yaml"}
+	}
+
+	for _, conf := range confs {
+		if len(conf) == 0 {
+			continue
+		}
+		if conf[0] == '{' {
+			// config as raw YAML or JSON
+			configs = append(configs, []byte(conf))
+		} else if data := parseConfString(conf); data != nil {
+			configs = append(configs, data)
+		} else {
+			// config as file
+			if ConfigPath == "" {
+				ConfigPath = conf
+			}
+
+			if data, _ = os.ReadFile(conf); data == nil {
+				continue
+			}
+
+			data = []byte(shell.ReplaceEnvVars(string(data)))
+			configs = append(configs, data)
+		}
+	}
+
+	if ConfigPath != "" {
+		if !filepath.IsAbs(ConfigPath) {
+			if cwd, err := os.Getwd(); err == nil {
+				ConfigPath = filepath.Join(cwd, ConfigPath)
+			}
+		}
+		Info["config_path"] = ConfigPath
+	}
+}
+
+func parseConfString(s string) []byte {
+	i := strings.IndexByte(s, '=')
+	if i < 0 {
+		return nil
+	}
+
+	items := strings.Split(s[:i], ".")
+	if len(items) < 2 {
+		return nil
+	}
+
+	// `log.level=trace` => `{log: {level: trace}}`
+	var pre string
+	var suf = s[i+1:]
+	for _, item := range items {
+		pre += "{" + item + ": "
+		suf += "}"
+	}
+
+	return []byte(pre + suf)
+}

internal/app/log.go

@@ -3,60 +3,123 @@ package app
 import (
 	"io"
 	"os"
+	"strings"
+	"sync"
 
+	"github.com/mattn/go-isatty"
 	"github.com/rs/zerolog"
-	"github.com/rs/zerolog/log"
 )
 
-var MemoryLog *circularBuffer
-
-func NewLogger(format string, level string) zerolog.Logger {
-	var writer io.Writer = os.Stdout
-
-	if format != "json" {
-		writer = zerolog.ConsoleWriter{
-			Out: writer, TimeFormat: "15:04:05.000", NoColor: format == "text",
-		}
-	}
-
-	MemoryLog = newBuffer(16)
-
-	writer = zerolog.MultiLevelWriter(writer, MemoryLog)
-
-	zerolog.TimeFieldFormat = zerolog.TimeFormatUnixMs
-
-	lvl, err := zerolog.ParseLevel(level)
-	if err != nil || lvl == zerolog.NoLevel {
-		lvl = zerolog.InfoLevel
-	}
-
-	return zerolog.New(writer).With().Timestamp().Logger().Level(lvl)
-}
+var MemoryLog = newBuffer()
 
 func GetLogger(module string) zerolog.Logger {
 	if s, ok := modules[module]; ok {
 		lvl, err := zerolog.ParseLevel(s)
 		if err == nil {
-			return log.Level(lvl)
+			return Logger.Level(lvl)
 		}
-		log.Warn().Err(err).Caller().Send()
+		Logger.Warn().Err(err).Caller().Send()
 	}
 
-	return log.Logger
+	return Logger
 }
 
-// modules log levels
-var modules map[string]string
+// initLogger support:
+// - output: empty (only to memory), stderr, stdout
+// - format: empty (autodetect color support), color, json, text
+// - time:   empty (disable timestamp), UNIXMS, UNIXMICRO, UNIXNANO
+// - level:  disabled, trace, debug, info, warn, error...
+func initLogger() {
+	var cfg struct {
+		Mod map[string]string `yaml:"log"`
+	}
 
-const chunkSize = 1 << 16
+	cfg.Mod = modules // defaults
+
+	LoadConfig(&cfg)
+
+	var writer io.Writer
+
+	switch output, path, _ := strings.Cut(modules["output"], ":"); output {
+	case "stderr":
+		writer = os.Stderr
+	case "stdout":
+		writer = os.Stdout
+	case "file":
+		if path == "" {
+			path = "go2rtc.log"
+		}
+		// if fail - only MemoryLog will be available
+		writer, _ = os.OpenFile(path, os.O_APPEND|os.O_CREATE|os.O_WRONLY, 0644)
+	}
+
+	timeFormat := modules["time"]
+
+	if writer != nil {
+		if format := modules["format"]; format != "json" {
+			console := &zerolog.ConsoleWriter{Out: writer}
+
+			switch format {
+			case "text":
+				console.NoColor = true
+			case "color":
+				console.NoColor = false // useless, but anyway
+			default:
+				// autodetection if output support color
+				// go-isatty - dependency for go-colorable - dependency for ConsoleWriter
+				console.NoColor = !isatty.IsTerminal(writer.(*os.File).Fd())
+			}
+
+			if timeFormat != "" {
+				console.TimeFormat = "15:04:05.000"
+			} else {
+				console.PartsOrder = []string{
+					zerolog.LevelFieldName,
+					zerolog.CallerFieldName,
+					zerolog.MessageFieldName,
+				}
+			}
+
+			writer = console
+		}
+
+		writer = zerolog.MultiLevelWriter(writer, MemoryLog)
+	} else {
+		writer = MemoryLog
+	}
+
+	lvl, _ := zerolog.ParseLevel(modules["level"])
+	Logger = zerolog.New(writer).Level(lvl)
+
+	if timeFormat != "" {
+		zerolog.TimeFieldFormat = timeFormat
+		Logger = Logger.With().Timestamp().Logger()
+	}
+}
+
+var Logger zerolog.Logger
+
+// modules log levels
+var modules = map[string]string{
+	"format": "", // useless, but anyway
+	"level":  "info",
+	"output": "stdout", // TODO: change to stderr someday
+	"time":   zerolog.TimeFormatUnixMs,
+}
+
+const (
+	chunkCount = 16
+	chunkSize  = 1 << 16
+)
 
 type circularBuffer struct {
 	chunks [][]byte
 	r, w   int
+	mu     sync.Mutex
 }
 
-func newBuffer(chunks int) *circularBuffer {
-	b := &circularBuffer{chunks: make([][]byte, 0, chunks)}
+func newBuffer() *circularBuffer {
+	b := &circularBuffer{chunks: make([][]byte, 0, chunkCount)}
 	// create first chunk
 	b.chunks = append(b.chunks, make([]byte, 0, chunkSize))
 	return b
@@ -65,16 +128,17 @@ func newBuffer(chunks int) *circularBuffer {
 func (b *circularBuffer) Write(p []byte) (n int, err error) {
 	n = len(p)
 
+	b.mu.Lock()
 	// check if chunk has size
 	if len(b.chunks[b.w])+n > chunkSize {
 		// increase write chunk index
-		if b.w++; b.w == cap(b.chunks) {
+		if b.w++; b.w == chunkCount {
 			b.w = 0
 		}
 		// check overflow
 		if b.r == b.w {
 			// increase read chunk index
-			if b.r++; b.r == cap(b.chunks) {
+			if b.r++; b.r == chunkCount {
 				b.r = 0
 			}
 		}
@@ -89,29 +153,34 @@ func (b *circularBuffer) Write(p []byte) (n int, err error) {
 	}
 
 	b.chunks[b.w] = append(b.chunks[b.w], p...)
+	b.mu.Unlock()
 	return
 }
 
 func (b *circularBuffer) WriteTo(w io.Writer) (n int64, err error) {
-	for i := b.r; ; {
-		var nn int
-		if nn, err = w.Write(b.chunks[i]); err != nil {
-			return
-		}
-		n += int64(nn)
+	buf := make([]byte, 0, chunkCount*chunkSize)
 
+	// use temp buffer inside mutex because w.Write can take some time
+	b.mu.Lock()
+	for i := b.r; ; {
+		buf = append(buf, b.chunks[i]...)
 		if i == b.w {
 			break
 		}
-		if i++; i == cap(b.chunks) {
+		if i++; i == chunkCount {
 			i = 0
 		}
 	}
-	return
+	b.mu.Unlock()
+
+	nn, err := w.Write(buf)
+	return int64(nn), err
 }
 
 func (b *circularBuffer) Reset() {
+	b.mu.Lock()
 	b.chunks[0] = b.chunks[0][:0]
 	b.r = 0
 	b.w = 0
+	b.mu.Unlock()
 }

internal/app/migrate.go

@@ -1,35 +0,0 @@
-package app
-
-import (
-	"encoding/json"
-	"os"
-
-	"github.com/rs/zerolog/log"
-)
-
-func migrateStore() {
-	const name = "go2rtc.json"
-
-	data, _ := os.ReadFile(name)
-	if data == nil {
-		return
-	}
-
-	var store struct {
-		Streams map[string]string `json:"streams"`
-	}
-
-	if err := json.Unmarshal(data, &store); err != nil {
-		log.Warn().Err(err).Caller().Send()
-		return
-	}
-
-	for id, url := range store.Streams {
-		if err := PatchConfig(id, url, "streams"); err != nil {
-			log.Warn().Err(err).Caller().Send()
-			return
-		}
-	}
-
-	_ = os.Remove(name)
-}

internal/bubble/bubble.go

@@ -7,13 +7,7 @@ import (
 )
 
 func Init() {
-	streams.HandleFunc("bubble", handle)
-}
-
-func handle(url string) (core.Producer, error) {
-	conn := bubble.NewClient(url)
-	if err := conn.Dial(); err != nil {
-		return nil, err
-	}
-	return conn, nil
+	streams.HandleFunc("bubble", func(source string) (core.Producer, error) {
+		return bubble.Dial(source)
+	})
 }

internal/debug/debug.go

@@ -2,16 +2,8 @@ package debug
 
 import (
 	"github.com/AlexxIT/go2rtc/internal/api"
-	"github.com/AlexxIT/go2rtc/internal/streams"
-	"github.com/AlexxIT/go2rtc/pkg/core"
 )
 
 func Init() {
 	api.HandleFunc("api/stack", stackHandler)
-
-	streams.HandleFunc("null", nullHandler)
-}
-
-func nullHandler(string) (core.Producer, error) {
-	return nil, nil
 }

internal/doorbird/doorbird.go

@@ -0,0 +1,36 @@
+package doorbird
+
+import (
+	"net/url"
+
+	"github.com/AlexxIT/go2rtc/internal/streams"
+	"github.com/AlexxIT/go2rtc/pkg/core"
+	"github.com/AlexxIT/go2rtc/pkg/doorbird"
+)
+
+func Init() {
+	streams.RedirectFunc("doorbird", func(rawURL string) (string, error) {
+		u, err := url.Parse(rawURL)
+		if err != nil {
+			return "", err
+		}
+
+		// https://www.doorbird.com/downloads/api_lan.pdf
+		switch u.Query().Get("media") {
+		case "video":
+			u.Path = "/bha-api/video.cgi"
+		case "audio":
+			u.Path = "/bha-api/audio-receive.cgi"
+		default:
+			return "", nil
+		}
+
+		u.Scheme = "http"
+
+		return u.String(), nil
+	})
+
+	streams.HandleFunc("doorbird", func(source string) (core.Producer, error) {
+		return doorbird.Dial(source)
+	})
+}

internal/dvrip/dvrip.go

@@ -10,26 +10,16 @@ import (
 
 	"github.com/AlexxIT/go2rtc/internal/api"
 	"github.com/AlexxIT/go2rtc/internal/streams"
-	"github.com/AlexxIT/go2rtc/pkg/core"
 	"github.com/AlexxIT/go2rtc/pkg/dvrip"
-	"github.com/rs/zerolog/log"
 )
 
 func Init() {
-	streams.HandleFunc("dvrip", handle)
+	streams.HandleFunc("dvrip", dvrip.Dial)
 
 	// DVRIP client autodiscovery
 	api.HandleFunc("api/dvrip", apiDvrip)
 }
 
-func handle(url string) (core.Producer, error) {
-	client, err := dvrip.Dial(url)
-	if err != nil {
-		return nil, err
-	}
-	return client, nil
-}
-
 const Port = 34569 // UDP port number for dvrip discovery
 
 func apiDvrip(w http.ResponseWriter, r *http.Request) {
@@ -92,10 +82,7 @@ func sendBroadcasts(conn *net.UDPConn) {
 
 	for i := 0; i < 3; i++ {
 		time.Sleep(100 * time.Millisecond)
-
-		if _, err = conn.WriteToUDP(data, addr); err != nil {
-			log.Err(err).Caller().Send()
-		}
+		_, _ = conn.WriteToUDP(data, addr)
 	}
 }
 

internal/eseecloud/eseecloud.go

@@ -0,0 +1,10 @@
+package eseecloud
+
+import (
+	"github.com/AlexxIT/go2rtc/internal/streams"
+	"github.com/AlexxIT/go2rtc/pkg/eseecloud"
+)
+
+func Init() {
+	streams.HandleFunc("eseecloud", eseecloud.Dial)
+}

internal/exec/README.md

@@ -0,0 +1,12 @@
+## Backchannel
+
+- You can check audio card names in the **Go2rtc > WebUI > Add**
+- You can specify multiple backchannel lines with different codecs
+
+```yaml
+sources:
+  two_way_audio_win:
+    - exec:ffmpeg -hide_banner -f dshow -i "audio=Microphone (High Definition Audio Device)" -c pcm_s16le -ar 16000 -ac 1 -f wav -
+    - exec:ffplay -nodisp -probesize 32 -f s16le -ar 16000 -#backchannel=1#audio=s16le/16000
+    - exec:ffplay -nodisp -probesize 32 -f alaw -ar 8000 -#backchannel=1#audio=alaw/8000
+```

internal/exec/exec.go

@@ -1,6 +1,7 @@
 package exec
 
 import (
+	"bufio"
 	"crypto/md5"
 	"encoding/hex"
 	"errors"
@@ -8,9 +9,9 @@ import (
 	"io"
 	"net/url"
 	"os"
-	"os/exec"
 	"strings"
 	"sync"
+	"syscall"
 	"time"
 
 	"github.com/AlexxIT/go2rtc/internal/app"
@@ -18,9 +19,9 @@ import (
 	"github.com/AlexxIT/go2rtc/internal/streams"
 	"github.com/AlexxIT/go2rtc/pkg/core"
 	"github.com/AlexxIT/go2rtc/pkg/magic"
+	"github.com/AlexxIT/go2rtc/pkg/pcm"
 	pkg "github.com/AlexxIT/go2rtc/pkg/rtsp"
 	"github.com/AlexxIT/go2rtc/pkg/shell"
-	"github.com/AlexxIT/go2rtc/pkg/stdin"
 	"github.com/rs/zerolog"
 )
 
@@ -48,9 +49,11 @@ func Init() {
 	log = app.GetLogger("exec")
 }
 
-func execHandle(rawURL string) (core.Producer, error) {
+func execHandle(rawURL string) (prod core.Producer, err error) {
+	rawURL, rawQuery, _ := strings.Cut(rawURL, "#")
+	query := streams.ParseQuery(rawQuery)
+
 	var path string
-	var query url.Values
 
 	// RTSP flow should have `{output}` inside URL
 	// pipe flow may have `#{params}` inside URL
@@ -62,60 +65,88 @@ func execHandle(rawURL string) (core.Producer, error) {
 		sum := md5.Sum([]byte(rawURL))
 		path = "/" + hex.EncodeToString(sum[:])
 		rawURL = rawURL[:i] + "rtsp://127.0.0.1:" + rtsp.Port + path + rawURL[i+8:]
-	} else if i = strings.IndexByte(rawURL, '#'); i > 0 {
-		query = streams.ParseQuery(rawURL[i+1:])
-		rawURL = rawURL[:i]
 	}
 
-	args := shell.QuoteSplit(rawURL[5:]) // remove `exec:`
-	cmd := exec.Command(args[0], args[1:]...)
-	if log.Debug().Enabled() {
-		cmd.Stderr = os.Stderr
+	cmd := shell.NewCommand(rawURL[5:]) // remove `exec:`
+	cmd.Stderr = &logWriter{
+		buf:   make([]byte, 512),
+		debug: log.Debug().Enabled(),
+	}
+
+	if s := query.Get("killsignal"); s != "" {
+		sig := syscall.Signal(core.Atoi(s))
+		cmd.Cancel = func() error {
+			log.Debug().Msgf("[exec] kill with signal=%d", sig)
+			return cmd.Process.Signal(sig)
+		}
+	}
+
+	if s := query.Get("killtimeout"); s != "" {
+		cmd.WaitDelay = time.Duration(core.Atoi(s)) * time.Second
+	}
+
+	if query.Get("backchannel") == "1" {
+		return pcm.NewBackchannel(cmd, query.Get("audio"))
 	}
 
 	if path == "" {
-		return handlePipe(rawURL, cmd, query)
+		prod, err = handlePipe(rawURL, cmd)
+	} else {
+		prod, err = handleRTSP(rawURL, cmd, path)
 	}
 
-	return handleRTSP(rawURL, cmd, path)
+	if err != nil {
+		_ = cmd.Close()
+	}
+
+	return
 }
 
-func handlePipe(_ string, cmd *exec.Cmd, query url.Values) (core.Producer, error) {
-	if query.Get("backchannel") == "1" {
-		return stdin.NewClient(cmd)
-	}
-
-	r, err := PipeCloser(cmd, query)
+func handlePipe(source string, cmd *shell.Command) (core.Producer, error) {
+	stdout, err := cmd.StdoutPipe()
 	if err != nil {
 		return nil, err
 	}
 
+	rd := struct {
+		io.Reader
+		io.Closer
+	}{
+		// add buffer for pipe reader to reduce syscall
+		bufio.NewReaderSize(stdout, core.BufferSize),
+		// stop cmd on close pipe call
+		cmd,
+	}
+
+	log.Debug().Strs("args", cmd.Args).Msg("[exec] run pipe")
+
+	ts := time.Now()
+
 	if err = cmd.Start(); err != nil {
 		return nil, err
 	}
 
-	prod, err := magic.Open(r)
+	prod, err := magic.Open(rd)
 	if err != nil {
-		_ = r.Close()
+		return nil, fmt.Errorf("exec/pipe: %w\n%s", err, cmd.Stderr)
 	}
 
-	return prod, err
+	if info, ok := prod.(core.Info); ok {
+		info.SetProtocol("pipe")
+		setRemoteInfo(info, source, cmd.Args)
+	}
+
+	log.Debug().Stringer("launch", time.Since(ts)).Msg("[exec] run pipe")
+
+	return prod, nil
 }
 
-func handleRTSP(url string, cmd *exec.Cmd, path string) (core.Producer, error) {
-	stderr := limitBuffer{buf: make([]byte, 512)}
-
-	if cmd.Stderr != nil {
-		cmd.Stderr = io.MultiWriter(cmd.Stderr, &stderr)
-	} else {
-		cmd.Stderr = &stderr
-	}
-
+func handleRTSP(source string, cmd *shell.Command, path string) (core.Producer, error) {
 	if log.Trace().Enabled() {
 		cmd.Stdout = os.Stdout
 	}
 
-	waiter := make(chan core.Producer)
+	waiter := make(chan *pkg.Conn, 1)
 
 	waitersMu.Lock()
 	waiters[path] = waiter
@@ -127,30 +158,31 @@ func handleRTSP(url string, cmd *exec.Cmd, path string) (core.Producer, error) {
 		waitersMu.Unlock()
 	}()
 
-	log.Debug().Str("url", url).Str("cmd", fmt.Sprintf("%s", strings.Join(cmd.Args, " "))).Msg("[exec] run")
+	log.Debug().Strs("args", cmd.Args).Msg("[exec] run rtsp")
 
 	ts := time.Now()
 
 	if err := cmd.Start(); err != nil {
-		log.Error().Err(err).Str("url", url).Msg("[exec]")
+		log.Error().Err(err).Str("source", source).Msg("[exec]")
 		return nil, err
 	}
 
-	done := make(chan error, 1)
-	go func() {
-		done <- cmd.Wait()
-	}()
+	timeout := time.NewTimer(30 * time.Second)
+	defer timeout.Stop()
 
 	select {
-	case <-time.After(time.Second * 60):
-		_ = cmd.Process.Kill()
-		log.Error().Str("url", url).Msg("[exec] timeout")
-		return nil, errors.New("timeout")
-	case <-done:
-		// limit message size
-		return nil, errors.New("exec: " + stderr.String())
+	case <-timeout.C:
+		// haven't received data from app in timeout
+		log.Error().Str("source", source).Msg("[exec] timeout")
+		return nil, errors.New("exec: timeout")
+	case <-cmd.Done():
+		// app fail before we receive any data
+		return nil, fmt.Errorf("exec/rtsp\n%s", cmd.Stderr)
 	case prod := <-waiter:
-		log.Debug().Stringer("launch", time.Since(ts)).Msg("[exec] run")
+		// app started successfully
+		log.Debug().Stringer("launch", time.Since(ts)).Msg("[exec] run rtsp")
+		setRemoteInfo(prod, source, cmd.Args)
+		prod.OnClose = cmd.Close
 		return prod, nil
 	}
 }
@@ -159,25 +191,63 @@ func handleRTSP(url string, cmd *exec.Cmd, path string) (core.Producer, error) {
 
 var (
 	log       zerolog.Logger
-	waiters   = map[string]chan core.Producer{}
+	waiters   = make(map[string]chan *pkg.Conn)
 	waitersMu sync.Mutex
 )
 
-type limitBuffer struct {
-	buf []byte
-	n   int
+type logWriter struct {
+	buf   []byte
+	debug bool
+	n     int
 }
 
-func (l *limitBuffer) String() string {
+func (l *logWriter) String() string {
 	if l.n == len(l.buf) {
 		return string(l.buf) + "..."
 	}
 	return string(l.buf[:l.n])
 }
 
-func (l *limitBuffer) Write(p []byte) (int, error) {
+func (l *logWriter) Write(p []byte) (n int, err error) {
 	if l.n < cap(l.buf) {
 		l.n += copy(l.buf[l.n:], p)
 	}
-	return len(p), nil
+	n = len(p)
+	if l.debug {
+		if p = trimSpace(p); p != nil {
+			log.Debug().Msgf("[exec] %s", p)
+		}
+	}
+	return
+}
+
+func trimSpace(b []byte) []byte {
+	start := 0
+	stop := len(b)
+	for ; start < stop; start++ {
+		if b[start] >= ' ' {
+			break // trim all ASCII before 0x20
+		}
+	}
+	for ; ; stop-- {
+		if stop == start {
+			return nil // skip empty output
+		}
+		if b[stop-1] > ' ' {
+			break // trim all ASCII before 0x21
+		}
+	}
+	return b[start:stop]
+}
+
+func setRemoteInfo(info core.Info, source string, args []string) {
+	info.SetSource(source)
+
+	if i := core.Index(args, "-i"); i > 0 && i < len(args)-1 {
+		rawURL := args[i+1]
+		if u, err := url.Parse(rawURL); err == nil && u.Host != "" {
+			info.SetRemoteAddr(u.Host)
+			info.SetURL(rawURL)
+		}
+	}
 }

internal/exec/pipe.go

@@ -1,56 +0,0 @@
-package exec
-
-import (
-	"bufio"
-	"errors"
-	"io"
-	"net/url"
-	"os/exec"
-	"syscall"
-	"time"
-
-	"github.com/AlexxIT/go2rtc/pkg/core"
-)
-
-// PipeCloser - return StdoutPipe that Kill cmd on Close call
-func PipeCloser(cmd *exec.Cmd, query url.Values) (io.ReadCloser, error) {
-	stdout, err := cmd.StdoutPipe()
-	if err != nil {
-		return nil, err
-	}
-
-	// add buffer for pipe reader to reduce syscall
-	return &pipeCloser{bufio.NewReaderSize(stdout, core.BufferSize), stdout, cmd, query}, nil
-}
-
-type pipeCloser struct {
-	io.Reader
-	io.Closer
-	cmd   *exec.Cmd
-	query url.Values
-}
-
-func (p *pipeCloser) Close() error {
-	return errors.Join(p.Closer.Close(), p.Kill(), p.Wait())
-}
-
-func (p *pipeCloser) Kill() error {
-	if s := p.query.Get("killsignal"); s != "" {
-		log.Trace().Msgf("[exec] kill with custom sig=%s", s)
-		sig := syscall.Signal(core.Atoi(s))
-		return p.cmd.Process.Signal(sig)
-	}
-	return p.cmd.Process.Kill()
-}
-
-func (p *pipeCloser) Wait() error {
-	if s := p.query.Get("killtimeout"); s != "" {
-		timeout := time.Duration(core.Atoi(s)) * time.Second
-		timer := time.AfterFunc(timeout, func() {
-			log.Trace().Msgf("[exec] kill after timeout=%s", s)
-			_ = p.cmd.Process.Kill()
-		})
-		defer timer.Stop() // stop timer if Wait ends before timeout
-	}
-	return p.cmd.Wait()
-}

internal/expr/expr.go

@@ -12,7 +12,7 @@ func Init() {
 	log := app.GetLogger("expr")
 
 	streams.RedirectFunc("expr", func(url string) (string, error) {
-		v, err := expr.Run(url[5:])
+		v, err := expr.Eval(url[5:], nil)
 		if err != nil {
 			return "", err
 		}

internal/ffmpeg/README.md

@@ -45,6 +45,13 @@
 [video4linux2,v4l2 @ 0x7f7de7c58bc0] Compressed:       mjpeg :          Motion-JPEG : 640x480 160x120 176x144 320x176 320x240 352x288 432x240 544x288 640x360 752x416 800x448 800x600 864x480 960x544 960x720 1024x576 1184x656 1280x720 1280x960
 ```
 
+## TTS
+
+```yaml
+streams:
+  tts: ffmpeg:#input=-readrate 1 -readrate_initial_burst 0.001 -f lavfi -i "flite=text='1 2 3 4 5 6 7 8 9 0'"#audio=pcma
+```
+
 ## Useful links
 
 - https://superuser.com/questions/564402/explanation-of-x264-tune

internal/ffmpeg/api.go

@@ -0,0 +1,51 @@
+package ffmpeg
+
+import (
+	"net/http"
+	"strings"
+
+	"github.com/AlexxIT/go2rtc/internal/streams"
+)
+
+func apiFFmpeg(w http.ResponseWriter, r *http.Request) {
+	if r.Method != "POST" {
+		http.Error(w, "", http.StatusMethodNotAllowed)
+		return
+	}
+
+	query := r.URL.Query()
+	dst := query.Get("dst")
+	stream := streams.Get(dst)
+	if stream == nil {
+		http.Error(w, "", http.StatusNotFound)
+		return
+	}
+
+	var src string
+	if s := query.Get("file"); s != "" {
+		if streams.Validate(s) == nil {
+			src = "ffmpeg:" + s + "#audio=auto#input=file"
+		}
+	} else if s = query.Get("live"); s != "" {
+		if streams.Validate(s) == nil {
+			src = "ffmpeg:" + s + "#audio=auto"
+		}
+	} else if s = query.Get("text"); s != "" {
+		if strings.IndexAny(s, `'"&%$`) < 0 {
+			src = "ffmpeg:tts?text=" + s
+			if s = query.Get("voice"); s != "" {
+				src += "&voice=" + s
+			}
+			src += "#audio=auto"
+		}
+	}
+
+	if src == "" {
+		http.Error(w, "", http.StatusBadRequest)
+		return
+	}
+
+	if err := stream.Play(src); err != nil {
+		http.Error(w, err.Error(), http.StatusInternalServerError)
+	}
+}

internal/ffmpeg/device/devices.go

@@ -1,11 +1,9 @@
 package device
 
 import (
-	"errors"
 	"net/http"
 	"net/url"
 	"strconv"
-	"strings"
 	"sync"
 
 	"github.com/AlexxIT/go2rtc/internal/api"
@@ -17,24 +15,15 @@ func Init(bin string) {
 	api.HandleFunc("api/ffmpeg/devices", apiDevices)
 }
 
-func GetInput(src string) (string, error) {
-	i := strings.IndexByte(src, '?')
-	if i < 0 {
-		return "", errors.New("empty query: " + src)
-	}
-
-	query, err := url.ParseQuery(src[i+1:])
+func GetInput(src string) string {
+	query, err := url.ParseQuery(src)
 	if err != nil {
-		return "", err
+		return ""
 	}
 
 	runonce.Do(initDevices)
 
-	if input := queryToInput(query); input != "" {
-		return input, nil
-	}
-
-	return "", errors.New("wrong query: " + src)
+	return queryToInput(query)
 }
 
 var Bin string

internal/ffmpeg/ffmpeg.go

@@ -4,33 +4,55 @@ import (
 	"net/url"
 	"strings"
 
+	"github.com/AlexxIT/go2rtc/internal/api"
 	"github.com/AlexxIT/go2rtc/internal/app"
 	"github.com/AlexxIT/go2rtc/internal/ffmpeg/device"
 	"github.com/AlexxIT/go2rtc/internal/ffmpeg/hardware"
 	"github.com/AlexxIT/go2rtc/internal/ffmpeg/virtual"
 	"github.com/AlexxIT/go2rtc/internal/rtsp"
 	"github.com/AlexxIT/go2rtc/internal/streams"
+	"github.com/AlexxIT/go2rtc/pkg/core"
 	"github.com/AlexxIT/go2rtc/pkg/ffmpeg"
+	"github.com/rs/zerolog"
 )
 
 func Init() {
 	var cfg struct {
 		Mod map[string]string `yaml:"ffmpeg"`
+		Log struct {
+			Level string `yaml:"ffmpeg"`
+		} `yaml:"log"`
 	}
 
 	cfg.Mod = defaults // will be overriden from yaml
+	cfg.Log.Level = "error"
 
 	app.LoadConfig(&cfg)
 
-	if app.GetLogger("exec").GetLevel() >= 0 {
-		defaults["global"] += " -v error"
+	log = app.GetLogger("ffmpeg")
+
+	// zerolog levels: trace debug         info warn    error fatal panic disabled
+	// FFmpeg  levels: trace debug verbose info warning error fatal panic quiet
+	if cfg.Log.Level == "warn" {
+		cfg.Log.Level = "warning"
 	}
+	defaults["global"] += " -v " + cfg.Log.Level
 
 	streams.RedirectFunc("ffmpeg", func(url string) (string, error) {
+		if _, err := Version(); err != nil {
+			return "", err
+		}
 		args := parseArgs(url[7:])
+		if core.Contains(args.Codecs, "auto") {
+			return "", nil // force call streams.HandleFunc("ffmpeg")
+		}
 		return "exec:" + args.String(), nil
 	})
 
+	streams.HandleFunc("ffmpeg", NewProducer)
+
+	api.HandleFunc("api/ffmpeg", apiFFmpeg)
+
 	device.Init(defaults["bin"])
 	hardware.Init(defaults["bin"])
 }
@@ -49,6 +71,9 @@ var defaults = map[string]string{
 	// output
 	"output":       "-user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}",
 	"output/mjpeg": "-f mjpeg -",
+	"output/raw":   "-f yuv4mpegpipe -",
+	"output/aac":   "-f adts -",
+	"output/wav":   "-f wav -",
 
 	// `-preset superfast` - we can't use ultrafast because it doesn't support `-profile main -level 4.1`
 	// `-tune zerolatency` - for minimal latency
@@ -59,6 +84,12 @@ var defaults = map[string]string{
 	"mjpeg": "-c:v mjpeg",
 	//"mjpeg": "-c:v mjpeg -force_duplicated_matrix:v 1 -huffman:v 0 -pix_fmt:v yuvj420p",
 
+	"raw":         "-c:v rawvideo",
+	"raw/gray8":   "-c:v rawvideo -pix_fmt:v gray8",
+	"raw/yuv420p": "-c:v rawvideo -pix_fmt:v yuv420p",
+	"raw/yuv422p": "-c:v rawvideo -pix_fmt:v yuv422p",
+	"raw/yuv444p": "-c:v rawvideo -pix_fmt:v yuv444p",
+
 	// https://ffmpeg.org/ffmpeg-codecs.html#libopus-1
 	// https://github.com/pion/webrtc/issues/1514
 	// https://ffmpeg.org/ffmpeg-resampler.html
@@ -82,6 +113,7 @@ var defaults = map[string]string{
 	"pcm/48000":  "-c:a pcm_s16be -ar:a 48000 -ac:a 1",
 	"pcml":       "-c:a pcm_s16le -ar:a 8000 -ac:a 1",
 	"pcml/8000":  "-c:a pcm_s16le -ar:a 8000 -ac:a 1",
+	"pcml/16000": "-c:a pcm_s16le -ar:a 16000 -ac:a 1",
 	"pcml/44100": "-c:a pcm_s16le -ar:a 44100 -ac:a 1",
 
 	// hardware Intel and AMD on Linux
@@ -98,8 +130,9 @@ var defaults = map[string]string{
 	// hardware Rockchip
 	// important to use custom ffmpeg https://github.com/AlexxIT/go2rtc/issues/768
 	// hevc - doesn't have a profile setting
-	"h264/rkmpp": "-c:v h264_rkmpp_encoder -g 50 -bf 0 -profile:v high -level:v 4.1",
-	"h265/rkmpp": "-c:v hevc_rkmpp_encoder -g 50 -bf 0 -level:v 5.1",
+	"h264/rkmpp":  "-c:v h264_rkmpp -g 50 -bf 0 -profile:v high -level:v 4.1",
+	"h265/rkmpp":  "-c:v hevc_rkmpp -g 50 -bf 0 -profile:v main -level:v 5.1",
+	"mjpeg/rkmpp": "-c:v mjpeg_rkmpp",
 
 	// hardware NVidia on Linux and Windows
 	// preset=p2 - faster, tune=ll - low latency
@@ -116,6 +149,8 @@ var defaults = map[string]string{
 	"h265/videotoolbox": "-c:v hevc_videotoolbox -g 50 -bf 0 -profile:v main -level:v 5.1",
 }
 
+var log zerolog.Logger
+
 // configTemplate - return template from config (defaults) if exist or return raw template
 func configTemplate(template string) string {
 	if s := defaults[template]; s != "" {
@@ -140,11 +175,13 @@ func inputTemplate(name, s string, query url.Values) string {
 func parseArgs(s string) *ffmpeg.Args {
 	// init FFmpeg arguments
 	args := &ffmpeg.Args{
-		Bin:    defaults["bin"],
-		Global: defaults["global"],
-		Output: defaults["output"],
+		Bin:     defaults["bin"],
+		Global:  defaults["global"],
+		Output:  defaults["output"],
+		Version: verAV,
 	}
 
+	var source = s
 	var query url.Values
 	if i := strings.IndexByte(s, '#'); i >= 0 {
 		query = streams.ParseQuery(s[i+1:])
@@ -187,17 +224,19 @@ func parseArgs(s string) *ffmpeg.Args {
 		default:
 			s += "?video&audio"
 		}
-		args.Input = inputTemplate("rtsp", s, query)
-	} else if strings.HasPrefix(s, "device?") {
-		var err error
-		args.Input, err = device.GetInput(s)
-		if err != nil {
-			return nil
+		s += "&source=ffmpeg:" + url.QueryEscape(source)
+		for _, v := range query["query"] {
+			s += "&" + v
 		}
-	} else if strings.HasPrefix(s, "virtual?") {
-		var err error
-		if args.Input, err = virtual.GetInput(s[8:]); err != nil {
-			return nil
+		args.Input = inputTemplate("rtsp", s, query)
+	} else if i = strings.Index(s, "?"); i > 0 {
+		switch s[:i] {
+		case "device":
+			args.Input = device.GetInput(s[i+1:])
+		case "virtual":
+			args.Input = virtual.GetInput(s[i+1:])
+		case "tts":
+			args.Input = virtual.GetInputTTS(s[i+1:])
 		}
 	} else {
 		args.Input = inputTemplate("file", s, query)
@@ -315,11 +354,27 @@ func parseArgs(s string) *ffmpeg.Args {
 		args.AddCodec("-an")
 	}
 
-	// transcoding to only mjpeg
-	if (args.Video == 1 && args.Audio == 0 && query.Get("video") == "mjpeg") ||
-		// no transcoding from mjpeg input
-		(args.Video == 0 && args.Audio == 0 && strings.Contains(args.Input, " mjpeg ")) {
-		args.Output = defaults["output/mjpeg"]
+	// change otput from RTSP to some other pipe format
+	switch {
+	case args.Video == 0 && args.Audio == 0:
+		// no transcoding from mjpeg input (ffmpeg device with support output as raw MJPEG)
+		if strings.Contains(args.Input, " mjpeg ") {
+			args.Output = defaults["output/mjpeg"]
+		}
+	case args.Video == 1 && args.Audio == 0:
+		switch core.Before(query.Get("video"), "/") {
+		case "mjpeg":
+			args.Output = defaults["output/mjpeg"]
+		case "raw":
+			args.Output = defaults["output/raw"]
+		}
+	case args.Video == 0 && args.Audio == 1:
+		switch core.Before(query.Get("audio"), "/") {
+		case "aac":
+			args.Output = defaults["output/aac"]
+		case "pcma", "pcmu", "pcml":
+			args.Output = defaults["output/wav"]
+		}
 	}
 
 	return args

internal/ffmpeg/ffmpeg_test.go

@@ -3,6 +3,7 @@ package ffmpeg
 import (
 	"testing"
 
+	"github.com/AlexxIT/go2rtc/pkg/ffmpeg"
 	"github.com/stretchr/testify/require"
 )
 
@@ -30,7 +31,7 @@ func TestParseArgsFile(t *testing.T) {
 		{
 			name:   "[FILE] video will be transcoded to H265 and rotate 270º, audio will be skipped",
 			source: "/media/bbb.mp4#video=h265#rotate=-90",
-			expect: `ffmpeg -hide_banner -re -i /media/bbb.mp4 -c:v libx265 -g 50 -profile:v main -level:v 5.1 -preset:v superfast -tune:v zerolatency -an -vf "transpose=2" -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`,
+			expect: `ffmpeg -hide_banner -re -i /media/bbb.mp4 -c:v libx265 -g 50 -profile:v main -level:v 5.1 -preset:v superfast -tune:v zerolatency -pix_fmt:v yuv420p -an -vf "transpose=2" -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`,
 		},
 		{
 			name:   "[FILE] video will be output for MJPEG to pipe, audio will be skipped",
@@ -52,85 +53,143 @@ func TestParseArgsFile(t *testing.T) {
 }
 
 func TestParseArgsDevice(t *testing.T) {
-	// [DEVICE] video will be output for MJPEG to pipe, with size 1920x1080
-	args := parseArgs("device?video=0&video_size=1920x1080")
-	require.Equal(t, `ffmpeg -hide_banner -f dshow -video_size 1920x1080 -i "video=0" -c copy -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
-
-	// [DEVICE] video will be transcoded to H265 with framerate 20, audio will be skipped
-	//args = parseArgs("device?video=0&video_size=1280x720&framerate=20#video=h265#audio=pcma")
-	args = parseArgs("device?video=0&framerate=20#video=h265")
-	require.Equal(t, `ffmpeg -hide_banner -f dshow -framerate 20 -i "video=0" -c:v libx265 -g 50 -profile:v main -level:v 5.1 -preset:v superfast -tune:v zerolatency -an -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
-
-	args = parseArgs("device?video=FaceTime HD Camera&audio=Microphone (High Definition Audio Device)")
-	require.Equal(t, `ffmpeg -hide_banner -f dshow -i "video=FaceTime HD Camera:audio=Microphone (High Definition Audio Device)" -c copy -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
+	tests := []struct {
+		name   string
+		source string
+		expect string
+	}{
+		{
+			name:   "[DEVICE] video will be output for MJPEG to pipe, with size 1920x1080",
+			source: "device?video=0&video_size=1920x1080",
+			expect: `ffmpeg -hide_banner -f dshow -video_size 1920x1080 -i "video=0" -c copy -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`,
+		},
+		{
+			name:   "[DEVICE] video will be transcoded to H265 with framerate 20, audio will be skipped",
+			source: "device?video=0&framerate=20#video=h265",
+			expect: `ffmpeg -hide_banner -f dshow -framerate 20 -i "video=0" -c:v libx265 -g 50 -profile:v main -level:v 5.1 -preset:v superfast -tune:v zerolatency -pix_fmt:v yuv420p -an -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`,
+		},
+		{
+			name:   "[DEVICE] video/audio",
+			source: "device?video=FaceTime HD Camera&audio=Microphone (High Definition Audio Device)",
+			expect: `ffmpeg -hide_banner -f dshow -i "video=FaceTime HD Camera:audio=Microphone (High Definition Audio Device)" -c copy -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`,
+		},
+	}
+	for _, test := range tests {
+		t.Run(test.name, func(t *testing.T) {
+			args := parseArgs(test.source)
+			require.Equal(t, test.expect, args.String())
+		})
+	}
 }
 
 func TestParseArgsIpCam(t *testing.T) {
-	// [HTTP] video will be copied
-	args := parseArgs("http://example.com")
-	require.Equal(t, `ffmpeg -hide_banner -fflags nobuffer -flags low_delay -i http://example.com -c copy -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
-
-	// [HTTP-MJPEG] video will be transcoded to H264
-	args = parseArgs("http://example.com#video=h264")
-	require.Equal(t, `ffmpeg -hide_banner -fflags nobuffer -flags low_delay -i http://example.com -c:v libx264 -g 50 -profile:v high -level:v 4.1 -preset:v superfast -tune:v zerolatency -pix_fmt:v yuv420p -an -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
-
-	// [HLS] video will be copied, audio will be skipped
-	args = parseArgs("https://example.com#video=copy")
-	require.Equal(t, `ffmpeg -hide_banner -fflags nobuffer -flags low_delay -i https://example.com -c:v copy -an -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
-
-	// [RTSP] video will be copied without transcoding codecs
-	args = parseArgs("rtsp://example.com")
-	require.Equal(t, `ffmpeg -hide_banner -allowed_media_types video+audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp://example.com -c copy -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
-
-	// [RTSP] video with resize to 1280x720, should be transcoded, so select H265
-	args = parseArgs("rtsp://example.com#video=h265#width=1280#height=720")
-	require.Equal(t, `ffmpeg -hide_banner -allowed_media_types video -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp://example.com -c:v libx265 -g 50 -profile:v main -level:v 5.1 -preset:v superfast -tune:v zerolatency -an -vf "scale=1280:720" -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
-
-	// [RTSP] video will be copied, changing RTSP transport from TCP to UDP+TCP
-	args = parseArgs("rtsp://example.com#input=rtsp/udp")
-	require.Equal(t, `ffmpeg -hide_banner -allowed_media_types video+audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -i rtsp://example.com -c copy -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
-
-	// [RTMP] video will be copied, changing RTSP transport from TCP to UDP+TCP
-	args = parseArgs("rtmp://example.com#input=rtsp/udp")
-	require.Equal(t, `ffmpeg -hide_banner -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -i rtmp://example.com -c copy -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
+	tests := []struct {
+		name   string
+		source string
+		expect string
+	}{
+		{
+			name:   "[HTTP] video will be copied",
+			source: "http://example.com",
+			expect: `ffmpeg -hide_banner -fflags nobuffer -flags low_delay -i http://example.com -c copy -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`,
+		},
+		{
+			name:   "[HTTP-MJPEG] video will be transcoded to H264",
+			source: "http://example.com#video=h264",
+			expect: `ffmpeg -hide_banner -fflags nobuffer -flags low_delay -i http://example.com -c:v libx264 -g 50 -profile:v high -level:v 4.1 -preset:v superfast -tune:v zerolatency -pix_fmt:v yuv420p -an -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`,
+		},
+		{
+			name:   "[HLS] video will be copied, audio will be skipped",
+			source: "https://example.com#video=copy",
+			expect: `ffmpeg -hide_banner -fflags nobuffer -flags low_delay -i https://example.com -c:v copy -an -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`,
+		},
+		{
+			name:   "[RTSP] video will be copied without transcoding codecs",
+			source: "rtsp://example.com",
+			expect: `ffmpeg -hide_banner -allowed_media_types video+audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp://example.com -c copy -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`,
+		},
+		{
+			name:   "[RTSP] video with resize to 1280x720, should be transcoded, so select H265",
+			source: "rtsp://example.com#video=h265#width=1280#height=720",
+			expect: `ffmpeg -hide_banner -allowed_media_types video -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp://example.com -c:v libx265 -g 50 -profile:v main -level:v 5.1 -preset:v superfast -tune:v zerolatency -pix_fmt:v yuv420p -an -vf "scale=1280:720" -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`,
+		},
+		{
+			name:   "[RTSP] video will be copied, changing RTSP transport from TCP to UDP+TCP",
+			source: "rtsp://example.com#input=rtsp/udp",
+			expect: `ffmpeg -hide_banner -allowed_media_types video+audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -i rtsp://example.com -c copy -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`,
+		},
+		{
+			name:   "[RTMP] video will be copied, changing RTSP transport from TCP to UDP+TCP",
+			source: "rtmp://example.com#input=rtsp/udp",
+			expect: `ffmpeg -hide_banner -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -i rtmp://example.com -c copy -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`,
+		},
+	}
+	for _, test := range tests {
+		t.Run(test.name, func(t *testing.T) {
+			args := parseArgs(test.source)
+			require.Equal(t, test.expect, args.String())
+		})
+	}
 }
 
 func TestParseArgsAudio(t *testing.T) {
-	// [AUDIO] audio will be transcoded to AAC, video will be skipped
-	args := parseArgs("rtsp:///example.com#audio=aac")
-	require.Equal(t, `ffmpeg -hide_banner -allowed_media_types audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp:///example.com -c:a aac -vn -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
-
-	// [AUDIO] audio will be transcoded to AAC/16000, video will be skipped
-	args = parseArgs("rtsp:///example.com#audio=aac/16000")
-	require.Equal(t, `ffmpeg -hide_banner -allowed_media_types audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp:///example.com -c:a aac -ar:a 16000 -ac:a 1 -vn -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
-
-	// [AUDIO] audio will be transcoded to OPUS, video will be skipped
-	args = parseArgs("rtsp:///example.com#audio=opus")
-	require.Equal(t, `ffmpeg -hide_banner -allowed_media_types audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp:///example.com -c:a libopus -application:a lowdelay -min_comp 0 -vn -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
-
-	// [AUDIO] audio will be transcoded to PCMU, video will be skipped
-	args = parseArgs("rtsp:///example.com#audio=pcmu")
-	require.Equal(t, `ffmpeg -hide_banner -allowed_media_types audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp:///example.com -c:a pcm_mulaw -ar:a 8000 -ac:a 1 -vn -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
-
-	// [AUDIO] audio will be transcoded to PCMU/16000, video will be skipped
-	args = parseArgs("rtsp:///example.com#audio=pcmu/16000")
-	require.Equal(t, `ffmpeg -hide_banner -allowed_media_types audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp:///example.com -c:a pcm_mulaw -ar:a 16000 -ac:a 1 -vn -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
-
-	// [AUDIO] audio will be transcoded to PCMU/48000, video will be skipped
-	args = parseArgs("rtsp:///example.com#audio=pcmu/48000")
-	require.Equal(t, `ffmpeg -hide_banner -allowed_media_types audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp:///example.com -c:a pcm_mulaw -ar:a 48000 -ac:a 1 -vn -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
-
-	// [AUDIO] audio will be transcoded to PCMA, video will be skipped
-	args = parseArgs("rtsp:///example.com#audio=pcma")
-	require.Equal(t, `ffmpeg -hide_banner -allowed_media_types audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp:///example.com -c:a pcm_alaw -ar:a 8000 -ac:a 1 -vn -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
-
-	// [AUDIO] audio will be transcoded to PCMA/16000, video will be skipped
-	args = parseArgs("rtsp:///example.com#audio=pcma/16000")
-	require.Equal(t, `ffmpeg -hide_banner -allowed_media_types audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp:///example.com -c:a pcm_alaw -ar:a 16000 -ac:a 1 -vn -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
-
-	// [AUDIO] audio will be transcoded to PCMA/48000, video will be skipped
-	args = parseArgs("rtsp:///example.com#audio=pcma/48000")
-	require.Equal(t, `ffmpeg -hide_banner -allowed_media_types audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp:///example.com -c:a pcm_alaw -ar:a 48000 -ac:a 1 -vn -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
+	tests := []struct {
+		name   string
+		source string
+		expect string
+	}{
+		{
+			name:   "[AUDIO] audio will be transcoded to AAC, video will be skipped",
+			source: "rtsp://example.com#audio=aac",
+			expect: `ffmpeg -hide_banner -allowed_media_types audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp://example.com -c:a aac -vn -f adts -`,
+		},
+		{
+			name:   "[AUDIO] audio will be transcoded to AAC/16000, video will be skipped",
+			source: "rtsp://example.com#audio=aac/16000",
+			expect: `ffmpeg -hide_banner -allowed_media_types audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp://example.com -c:a aac -ar:a 16000 -ac:a 1 -vn -f adts -`,
+		},
+		{
+			name:   "[AUDIO] audio will be transcoded to OPUS, video will be skipped",
+			source: "rtsp://example.com#audio=opus",
+			expect: `ffmpeg -hide_banner -allowed_media_types audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp://example.com -c:a libopus -application:a lowdelay -min_comp 0 -vn -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`,
+		},
+		{
+			name:   "[AUDIO] audio will be transcoded to PCMU, video will be skipped",
+			source: "rtsp://example.com#audio=pcmu",
+			expect: `ffmpeg -hide_banner -allowed_media_types audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp://example.com -c:a pcm_mulaw -ar:a 8000 -ac:a 1 -vn -f wav -`,
+		},
+		{
+			name:   "[AUDIO] audio will be transcoded to PCMU/16000, video will be skipped",
+			source: "rtsp://example.com#audio=pcmu/16000",
+			expect: `ffmpeg -hide_banner -allowed_media_types audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp://example.com -c:a pcm_mulaw -ar:a 16000 -ac:a 1 -vn -f wav -`,
+		},
+		{
+			name:   "[AUDIO] audio will be transcoded to PCMU/48000, video will be skipped",
+			source: "rtsp://example.com#audio=pcmu/48000",
+			expect: `ffmpeg -hide_banner -allowed_media_types audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp://example.com -c:a pcm_mulaw -ar:a 48000 -ac:a 1 -vn -f wav -`,
+		},
+		{
+			name:   "[AUDIO] audio will be transcoded to PCMA, video will be skipped",
+			source: "rtsp://example.com#audio=pcma",
+			expect: `ffmpeg -hide_banner -allowed_media_types audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp://example.com -c:a pcm_alaw -ar:a 8000 -ac:a 1 -vn -f wav -`,
+		},
+		{
+			name:   "[AUDIO] audio will be transcoded to PCMA/16000, video will be skipped",
+			source: "rtsp://example.com#audio=pcma/16000",
+			expect: `ffmpeg -hide_banner -allowed_media_types audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp://example.com -c:a pcm_alaw -ar:a 16000 -ac:a 1 -vn -f wav -`,
+		},
+		{
+			name:   "[AUDIO] audio will be transcoded to PCMA/48000, video will be skipped",
+			source: "rtsp://example.com#audio=pcma/48000",
+			expect: `ffmpeg -hide_banner -allowed_media_types audio -fflags nobuffer -flags low_delay -timeout 5000000 -user_agent go2rtc/ffmpeg -rtsp_flags prefer_tcp -i rtsp://example.com -c:a pcm_alaw -ar:a 48000 -ac:a 1 -vn -f wav -`,
+		},
+	}
+	for _, test := range tests {
+		t.Run(test.name, func(t *testing.T) {
+			args := parseArgs(test.source)
+			require.Equal(t, test.expect, args.String())
+		})
+	}
 }
 
 func TestParseArgsHwVaapi(t *testing.T) {
@@ -192,15 +251,33 @@ func _TestParseArgsHwV4l2m2m(t *testing.T) {
 }
 
 func TestParseArgsHwRKMPP(t *testing.T) {
-	// [HTTP-MJPEG] video will be transcoded to H264
-	args := parseArgs("http://example.com#video=h264#hardware=rkmpp")
-	require.Equal(t, `ffmpeg -hide_banner -fflags nobuffer -flags low_delay -i http://example.com -c:v h264_rkmpp_encoder -g 50 -bf 0 -profile:v high -level:v 4.1 -an -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
-
-	args = parseArgs("http://example.com#video=h264#rotate=180#hardware=rkmpp")
-	require.Equal(t, `ffmpeg -hide_banner -fflags nobuffer -flags low_delay -i http://example.com -c:v h264_rkmpp_encoder -g 50 -bf 0 -profile:v high -level:v 4.1 -an -vf "transpose=1,transpose=1" -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
-
-	args = parseArgs("http://example.com#video=h264#height=320#hardware=rkmpp")
-	require.Equal(t, `ffmpeg -hide_banner -fflags nobuffer -flags low_delay -i http://example.com -c:v h264_rkmpp_encoder -g 50 -bf 0 -profile:v high -level:v 4.1 -height 320 -an -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`, args.String())
+	tests := []struct {
+		name   string
+		source string
+		expect string
+	}{
+		{
+			name:   "[FILE] transcoding to H264",
+			source: "bbb.mp4#video=h264#hardware=rkmpp",
+			expect: `ffmpeg -hide_banner -hwaccel rkmpp -hwaccel_output_format drm_prime -afbc rga -re -i bbb.mp4 -c:v h264_rkmpp -g 50 -bf 0 -profile:v high -level:v 4.1 -an -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`,
+		},
+		{
+			name:   "[FILE] transcoding with rotation",
+			source: "bbb.mp4#video=h264#rotate=180#hardware=rkmpp",
+			expect: `ffmpeg -hide_banner -hwaccel rkmpp -hwaccel_output_format drm_prime -afbc rga -re -i bbb.mp4 -c:v h264_rkmpp -g 50 -bf 0 -profile:v high -level:v 4.1 -an -vf "format=drm_prime|nv12,hwupload,vpp_rkrga=transpose=4" -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`,
+		},
+		{
+			name:   "[FILE] transcoding with scaling",
+			source: "bbb.mp4#video=h264#height=320#hardware=rkmpp",
+			expect: `ffmpeg -hide_banner -hwaccel rkmpp -hwaccel_output_format drm_prime -afbc rga -re -i bbb.mp4 -c:v h264_rkmpp -g 50 -bf 0 -profile:v high -level:v 4.1 -an -vf "format=drm_prime|nv12,hwupload,scale_rkrga=-1:320:force_original_aspect_ratio=0" -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`,
+		},
+	}
+	for _, test := range tests {
+		t.Run(test.name, func(t *testing.T) {
+			args := parseArgs(test.source)
+			require.Equal(t, test.expect, args.String())
+		})
+	}
 }
 
 func _TestParseArgsHwCuda(t *testing.T) {
@@ -292,3 +369,23 @@ func TestDrawText(t *testing.T) {
 		})
 	}
 }
+
+func TestVersion(t *testing.T) {
+	verAV = ffmpeg.Version61
+	tests := []struct {
+		name   string
+		source string
+		expect string
+	}{
+		{
+			source: "/media/bbb.mp4",
+			expect: `ffmpeg -hide_banner -readrate_initial_burst 0.001 -re -i /media/bbb.mp4 -c copy -user_agent ffmpeg/go2rtc -rtsp_transport tcp -f rtsp {output}`,
+		},
+	}
+	for _, test := range tests {
+		t.Run(test.name, func(t *testing.T) {
+			args := parseArgs(test.source)
+			require.Equal(t, test.expect, args.String())
+		})
+	}
+}

internal/ffmpeg/hardware/README.md

@@ -0,0 +1,106 @@
+# Hardware
+
+You **DON'T** need hardware acceleration if:
+
+- you not using [FFmpeg source](https://github.com/AlexxIT/go2rtc#source-ffmpeg)
+- you using only `#video=copy` for FFmpeg source
+- you using only `#audio=...` (any audio) transcoding for FFmpeg source
+
+You **NEED** hardware acceleration if you using `#video=h264`, `#video=h265`, `#video=mjpeg` (video) transcoding.
+
+## Important
+
+- Acceleration is disabled by default because it can be unstable (it can be changed in future)
+- go2rtc can automatically detect supported hardware acceleration if enabled
+- go2rtc will enable hardware decoding only if hardware encoding supported
+- go2rtc will use the same GPU for decoder and encoder
+- Intel and AMD will switch to software decoder if input codec is not supported with hardware decoder
+- NVidia will fail if input codec is not supported with hardware decoder
+- Raspberry always uses software decoder
+
+```yaml
+streams:
+  # auto select hardware encoder
+  camera1_hw: ffmpeg:rtsp://rtsp:12345678@192.168.1.123/av_stream/ch0#video=h264#hardware
+  
+  # manual select hardware encoder (vaapi, cuda, v4l2m2m, dxva2, videotoolbox)
+  camera1_vaapi: ffmpeg:rtsp://rtsp:12345678@192.168.1.123/av_stream/ch0#video=h264#hardware=vaapi
+```
+
+## Docker and Hass Addon
+
+There are two versions of the Docker container and Hass Add-on:
+
+- Latest (alpine) support hardware acceleration for Intel iGPU (CPU with Graphics) and Raspberry.
+- Hardware (debian 12) support Intel iGPU, AMD GPU, NVidia GPU.
+
+## Intel iGPU
+
+**Supported on:** Windows binary, Linux binary, Docker, Hass Addon.
+
+If you have Intel CPU Sandy Bridge (2011) with Graphics, you already have support hardware decoding/encoding for `AVC/H.264`.
+
+If you have Intel CPU Skylake (2015) with Graphics, you already have support hardware decoding/encoding for `AVC/H.264`, `HEVC/H.265` and `MJPEG`.
+
+Read more [here](https://en.wikipedia.org/wiki/Intel_Quick_Sync_Video#Hardware_decoding_and_encoding) and [here](https://en.wikipedia.org/wiki/Intel_Graphics_Technology#Capabilities_(GPU_video_acceleration)).
+
+Linux and Docker:
+
+- It may be important to have the latest version of the OS with the latest version of the Linux kernel. For example, on my **Debian 10 (kernel 4.19)** it did not work, but after update to **Debian 11 (kernel 5.10)** all was fine.
+- In case of troube check you have `/dev/dri/` folder on your host.
+
+Docker users should add `--privileged` option to container for access to Hardware.
+
+**PS.** Supported via [VAAPI](https://trac.ffmpeg.org/wiki/Hardware/VAAPI) engine on Linux and [DXVA2+QSV](https://trac.ffmpeg.org/wiki/Hardware/QuickSync) engine on Windows.
+
+## AMD GPU
+
+*I don't have the hardware for test support!!!*
+
+**Supported on:** Linux binary, Docker, Hass Addon.
+
+Docker users should install: `alexxit/go2rtc:master-hardware`. Docker users should add `--privileged` option to container for access to Hardware.
+
+Hass Addon users should install **go2rtc master hardware** version.
+
+**PS.** Supported via [VAAPI](https://trac.ffmpeg.org/wiki/Hardware/VAAPI) engine.
+
+## NVidia GPU
+
+**Supported on:** Windows binary, Linux binary, Docker.
+
+Docker users should install: `alexxit/go2rtc:master-hardware`.
+
+Read more [here](https://docs.frigate.video/configuration/hardware_acceleration) and [here](https://jellyfin.org/docs/general/administration/hardware-acceleration/#nvidia-hardware-acceleration-on-docker-linux).
+
+**PS.** Supported via [CUDA](https://trac.ffmpeg.org/wiki/HWAccelIntro#CUDANVENCNVDEC) engine.
+
+## Raspberry Pi 3
+
+**Supported on:** Linux binary, Docker, Hass Addon.
+
+I don't recommend using transcoding on the Raspberry Pi 3. It's extreamly slow, even with hardware acceleration. Also it may fail when transcoding 2K+ stream.
+
+## Raspberry Pi 4
+
+*I don't have the hardware for test support!!!*
+
+**Supported on:** Linux binary, Docker, Hass Addon.
+
+**PS.** Supported via [v4l2m2m](https://lalitm.com/hw-encoding-raspi/) engine.
+
+## macOS
+
+In my tests, transcoding is faster on the M1 CPU than on the M1 GPU. Transcoding time on M1 CPU better than any Intel iGPU and comparable to NVidia RTX 2070.
+
+**PS.** Supported via [videotoolbox](https://trac.ffmpeg.org/wiki/HWAccelIntro#VideoToolbox) engine.
+
+## Rockchip
+
+- Important to use custom FFmpeg with Rockchip support from [@nyanmisaka](https://github.com/nyanmisaka/ffmpeg-rockchip)
+  - Static binaries from [@MarcA711](https://github.com/MarcA711/Rockchip-FFmpeg-Builds/releases/)
+- Important to have Linux kernel 5.10 or 6.1
+
+**Tested**
+
+- [Orange Pi 3B](https://www.armbian.com/orangepi3b/) with Armbian 6.1, support transcoding H264, H265, MJPEG

internal/ffmpeg/hardware/hardware.go

@@ -7,8 +7,6 @@ import (
 
 	"github.com/AlexxIT/go2rtc/internal/api"
 	"github.com/AlexxIT/go2rtc/pkg/ffmpeg"
-
-	"github.com/rs/zerolog/log"
 )
 
 const (
@@ -130,19 +128,32 @@ func MakeHardware(args *ffmpeg.Args, engine string, defaults map[string]string)
 		case EngineRKMPP:
 			args.Codecs[i] = defaults[name+"/"+engine]
 
-			for j, filter := range args.Filters {
-				if strings.HasPrefix(filter, "scale=") {
-					args.Filters = append(args.Filters[:j], args.Filters[j+1:]...)
+			if !args.HasFilters("drawtext=") {
+				args.Input = "-hwaccel rkmpp -hwaccel_output_format drm_prime -afbc rga " + args.Input
 
-					width, height, _ := strings.Cut(filter[6:], ":")
-					if width != "-1" {
-						args.Codecs[i] += " -width " + width
+				for i, filter := range args.Filters {
+					if strings.HasPrefix(filter, "scale=") {
+						args.Filters[i] = "scale_rkrga=" + filter[6:] + ":force_original_aspect_ratio=0"
 					}
-					if height != "-1" {
-						args.Codecs[i] += " -height " + height
+					if strings.HasPrefix(filter, "transpose=") {
+						if filter == "transpose=1,transpose=1" { // 180 degrees half-turn
+							args.Filters[i] = "vpp_rkrga=transpose=4" // reversal
+						} else {
+							args.Filters[i] = "vpp_rkrga=transpose=" + filter[10:]
+						}
 					}
-					break
 				}
+
+				if len(args.Filters) > 0 {
+					// fix if input doesn't support hwaccel, do nothing when support
+					// insert as first filter before hardware scale and transpose
+					args.InsertFilter("format=drm_prime|nv12,hwupload")
+				}
+			} else {
+				// enable software pixel for drawtext, scale and transpose
+				args.Input = "-hwaccel rkmpp -hwaccel_output_format nv12 -afbc rga " + args.Input
+
+				args.AddFilter("hwupload")
 			}
 		}
 	}
@@ -152,7 +163,6 @@ var cache = map[string]string{}
 
 func run(bin string, args string) bool {
 	err := exec.Command(bin, strings.Split(args, " ")...).Run()
-	log.Printf("%v %v", args, err)
 	return err == nil
 }
 

internal/ffmpeg/hardware/hardware_unix.go

@@ -11,8 +11,9 @@ import (
 const (
 	ProbeV4L2M2MH264 = "-f lavfi -i testsrc2 -t 1 -c h264_v4l2m2m -f null -"
 	ProbeV4L2M2MH265 = "-f lavfi -i testsrc2 -t 1 -c hevc_v4l2m2m -f null -"
-	ProbeRKMPPH264   = "-f lavfi -i testsrc2 -t 1 -c h264_rkmpp_encoder -f null -"
-	ProbeRKMPPH265   = "-f lavfi -i testsrc2 -t 1 -c hevc_rkmpp_encoder -f null -"
+	ProbeRKMPPH264   = "-f lavfi -i testsrc2 -t 1 -c h264_rkmpp -f null -"
+	ProbeRKMPPH265   = "-f lavfi -i testsrc2 -t 1 -c hevc_rkmpp -f null -"
+	ProbeRKMPPJPEG   = "-f lavfi -i testsrc2 -t 1 -c mjpeg_rkmpp -f null -"
 	ProbeVAAPIH264   = "-init_hw_device vaapi -f lavfi -i testsrc2 -t 1 -vf format=nv12,hwupload -c h264_vaapi -f null -"
 	ProbeVAAPIH265   = "-init_hw_device vaapi -f lavfi -i testsrc2 -t 1 -vf format=nv12,hwupload -c hevc_vaapi -f null -"
 	ProbeVAAPIJPEG   = "-init_hw_device vaapi -f lavfi -i testsrc2 -t 1 -vf format=nv12,hwupload -c mjpeg_vaapi -f null -"
@@ -39,6 +40,10 @@ func ProbeAll(bin string) []*api.Source {
 				Name: runToString(bin, ProbeRKMPPH265),
 				URL:  "ffmpeg:...#video=h265#hardware=" + EngineRKMPP,
 			},
+			{
+				Name: runToString(bin, ProbeRKMPPJPEG),
+				URL:  "ffmpeg:...#video=mjpeg#hardware=" + EngineRKMPP,
+			},
 		}
 	}
 
@@ -83,6 +88,10 @@ func ProbeHardware(bin, name string) string {
 			if run(bin, ProbeRKMPPH265) {
 				return EngineRKMPP
 			}
+		case "mjpeg":
+			if run(bin, ProbeRKMPPJPEG) {
+				return EngineRKMPP
+			}
 		}
 
 		return EngineSoftware

internal/ffmpeg/producer.go

@@ -0,0 +1,121 @@
+package ffmpeg
+
+import (
+	"encoding/json"
+	"errors"
+	"net/url"
+	"strconv"
+	"strings"
+
+	"github.com/AlexxIT/go2rtc/internal/streams"
+	"github.com/AlexxIT/go2rtc/pkg/aac"
+	"github.com/AlexxIT/go2rtc/pkg/core"
+)
+
+type Producer struct {
+	core.Connection
+	url    string
+	query  url.Values
+	ffmpeg core.Producer
+}
+
+// NewProducer - FFmpeg producer with auto selection video/audio codec based on client capabilities
+func NewProducer(url string) (core.Producer, error) {
+	p := &Producer{}
+
+	i := strings.IndexByte(url, '#')
+	p.url, p.query = url[:i], streams.ParseQuery(url[i+1:])
+
+	// ffmpeg.NewProducer support only one audio
+	if len(p.query["video"]) != 0 || len(p.query["audio"]) != 1 {
+		return nil, errors.New("ffmpeg: unsupported params: " + url[i:])
+	}
+
+	p.ID = core.NewID()
+	p.FormatName = "ffmpeg"
+	p.Medias = []*core.Media{
+		{
+			// we can support only audio, because don't know FmtpLine for H264 and PayloadType for MJPEG
+			Kind:      core.KindAudio,
+			Direction: core.DirectionRecvonly,
+			// codecs in order from best to worst
+			Codecs: []*core.Codec{
+				// OPUS will always marked as OPUS/48000/2
+				{Name: core.CodecOpus, ClockRate: 48000, Channels: 2},
+				{Name: core.CodecPCML, ClockRate: 16000},
+				{Name: core.CodecPCM, ClockRate: 16000},
+				{Name: core.CodecPCMA, ClockRate: 16000},
+				{Name: core.CodecPCMU, ClockRate: 16000},
+				{Name: core.CodecPCM, ClockRate: 8000},
+				{Name: core.CodecPCMA, ClockRate: 8000},
+				{Name: core.CodecPCMU, ClockRate: 8000},
+				// AAC has unknown problems on Dahua two way
+				{Name: core.CodecAAC, ClockRate: 16000, FmtpLine: aac.FMTP + "1408"},
+			},
+		},
+	}
+	return p, nil
+}
+
+func (p *Producer) Start() error {
+	var err error
+	if p.ffmpeg, err = streams.GetProducer(p.newURL()); err != nil {
+		return err
+	}
+
+	for i, media := range p.ffmpeg.GetMedias() {
+		track, err := p.ffmpeg.GetTrack(media, media.Codecs[0])
+		if err != nil {
+			return err
+		}
+		p.Receivers[i].Replace(track)
+	}
+
+	return p.ffmpeg.Start()
+}
+
+func (p *Producer) Stop() error {
+	if p.ffmpeg == nil {
+		return nil
+	}
+	return p.ffmpeg.Stop()
+}
+
+func (p *Producer) MarshalJSON() ([]byte, error) {
+	if p.ffmpeg == nil {
+		return json.Marshal(p.Connection)
+	}
+	return json.Marshal(p.ffmpeg)
+}
+
+func (p *Producer) newURL() string {
+	s := p.url
+	// rewrite codecs in url from auto to known presets from defaults
+	for _, receiver := range p.Receivers {
+		codec := receiver.Codec
+		switch codec.Name {
+		case core.CodecOpus:
+			s += "#audio=opus"
+		case core.CodecAAC:
+			s += "#audio=aac/16000"
+		case core.CodecPCML:
+			s += "#audio=pcml/16000"
+		case core.CodecPCM:
+			s += "#audio=pcm/" + strconv.Itoa(int(codec.ClockRate))
+		case core.CodecPCMA:
+			s += "#audio=pcma/" + strconv.Itoa(int(codec.ClockRate))
+		case core.CodecPCMU:
+			s += "#audio=pcmu/" + strconv.Itoa(int(codec.ClockRate))
+		}
+	}
+	// add other params
+	for key, values := range p.query {
+		if key != "audio" {
+			for _, value := range values {
+				s += "#" + key + "=" + value
+			}
+		}
+	}
+
+	return s
+}

internal/ffmpeg/version.go

@@ -0,0 +1,46 @@
+package ffmpeg
+
+import (
+	"errors"
+	"os/exec"
+	"sync"
+
+	"github.com/AlexxIT/go2rtc/pkg/ffmpeg"
+)
+
+var verMu sync.Mutex
+var verErr error
+var verFF string
+var verAV string
+
+func Version() (string, error) {
+	verMu.Lock()
+	defer verMu.Unlock()
+
+	if verFF != "" {
+		return verFF, verErr
+	}
+
+	cmd := exec.Command(defaults["bin"], "-version")
+	b, err := cmd.Output()
+	if err != nil {
+		verFF = "-"
+		verErr = err
+		return verFF, verErr
+	}
+
+	verFF, verAV = ffmpeg.ParseVersion(b)
+
+	if verFF == "" {
+		verFF = "?"
+	}
+
+	// better to compare libavformat, because nightly/master builds
+	if verAV != "" && verAV < ffmpeg.Version50 {
+		verErr = errors.New("ffmpeg: unsupported version: " + verFF)
+	}
+
+	log.Debug().Str("version", verFF).Str("libavformat", verAV).Msgf("[ffmpeg] bin")
+
+	return verFF, verErr
+}

internal/ffmpeg/virtual/virtual.go

@@ -4,56 +4,76 @@ import (
 	"net/url"
 )
 
-func GetInput(src string) (string, error) {
+func GetInput(src string) string {
 	query, err := url.ParseQuery(src)
 	if err != nil {
-		return "", err
+		return ""
 	}
 
-	// set defaults (using Add instead of Set)
-	query.Add("video", "testsrc")
-	query.Add("size", "1920x1080")
-	query.Add("decimals", "2")
+	input := "-re"
 
-	// https://ffmpeg.org/ffmpeg-filters.html
-	video := query.Get("video")
-	input := "-re -f lavfi -i " + video
+	for _, video := range query["video"] {
+		// https://ffmpeg.org/ffmpeg-filters.html
+		sep := "=" // first separator
 
-	sep := "=" // first separator
-	for key, values := range query {
-		value := values[0]
-
-		// https://ffmpeg.org/ffmpeg-utils.html#video-size-syntax
-		switch key {
-		case "color", "rate", "duration", "sar":
-		case "size":
-			switch value {
-			case "720":
-				value = "1280x720" // crf=1 -> 12 Mbps
-			case "1080":
-				value = "1920x1080" // crf=1 -> 25 Mbps
-			case "2K":
-				value = "2560x1440" // crf=1 -> 43 Mbps
-			case "4K":
-				value = "3840x2160" // crf=1 -> 103 Mbps
-			case "8K":
-				value = "7680x4230" // https://reolink.com/blog/8k-resolution/
-			}
-		case "decimals":
-			if video != "testsrc" {
-				continue
-			}
-		default:
-			continue
+		if video == "" {
+			video = "testsrc=decimals=2" // default video
+			sep = ":"
 		}
 
-		input += sep + key + "=" + value
-		sep = ":" // next separator
+		input += " -f lavfi -i " + video
+
+		// set defaults (using Add instead of Set)
+		query.Add("size", "1920x1080")
+
+		for key, values := range query {
+			value := values[0]
+
+			// https://ffmpeg.org/ffmpeg-utils.html#video-size-syntax
+			switch key {
+			case "color", "rate", "duration", "sar", "decimals":
+			case "size":
+				switch value {
+				case "720":
+					value = "1280x720" // crf=1 -> 12 Mbps
+				case "1080":
+					value = "1920x1080" // crf=1 -> 25 Mbps
+				case "2K":
+					value = "2560x1440" // crf=1 -> 43 Mbps
+				case "4K":
+					value = "3840x2160" // crf=1 -> 103 Mbps
+				case "8K":
+					value = "7680x4230" // https://reolink.com/blog/8k-resolution/
+				}
+			default:
+				continue
+			}
+
+			input += sep + key + "=" + value
+			sep = ":" // next separator
+		}
+
+		if s := query.Get("format"); s != "" {
+			input += ",format=" + s
+		}
 	}
 
-	if s := query.Get("format"); s != "" {
-		input += ",format=" + s
-	}
-
-	return input, nil
+	return input
+}
+
+func GetInputTTS(src string) string {
+	query, err := url.ParseQuery(src)
+	if err != nil {
+		return ""
+	}
+
+	input := `-re -f lavfi -i "flite=text='` + query.Get("text") + `'`
+
+	// ffmpeg -f lavfi -i flite=list_voices=1
+	// awb, kal, kal16, rms, slt
+	if voice := query.Get("voice"); voice != "" {
+		input += ":voice" + voice
+	}
+
+	return input + `"`
 }

internal/ffmpeg/virtual/virtual_test.go

@@ -0,0 +1,20 @@
+package virtual
+
+import (
+	"testing"
+
+	"github.com/stretchr/testify/require"
+)
+
+func TestGetInput(t *testing.T) {
+	s := GetInput("video")
+	require.Equal(t, "-re -f lavfi -i testsrc=decimals=2:size=1920x1080", s)
+
+	s = GetInput("video=testsrc2&size=4K")
+	require.Equal(t, "-re -f lavfi -i testsrc2=size=3840x2160", s)
+}
+
+func TestGetInputTTS(t *testing.T) {
+	s := GetInputTTS("text=hello world&voice=slt")
+	require.Equal(t, `-re -f lavfi -i "flite=text='hello world':voiceslt"`, s)
+}

internal/flussonic/flussonic.go

@@ -0,0 +1,10 @@
+package flussonic
+
+import (
+	"github.com/AlexxIT/go2rtc/internal/streams"
+	"github.com/AlexxIT/go2rtc/pkg/flussonic"
+)
+
+func Init() {
+	streams.HandleFunc("flussonic", flussonic.Dial)
+}

internal/gopro/gopro.go

@@ -10,15 +10,13 @@ import (
 )
 
 func Init() {
-	streams.HandleFunc("gopro", handleGoPro)
+	streams.HandleFunc("gopro", func(source string) (core.Producer, error) {
+		return gopro.Dial(source)
+	})
 
 	api.HandleFunc("api/gopro", apiGoPro)
 }
 
-func handleGoPro(rawURL string) (core.Producer, error) {
-	return gopro.Dial(rawURL)
-}
-
 func apiGoPro(w http.ResponseWriter, r *http.Request) {
 	var items []*api.Source
 

internal/hass/api.go

@@ -63,7 +63,7 @@ func apiStream(w http.ResponseWriter, r *http.Request) {
 			return
 		}
 
-		s, err = webrtc.ExchangeSDP(stream, string(offer), "WebRTC/Hass sync", r.UserAgent())
+		s, err = webrtc.ExchangeSDP(stream, string(offer), "hass/webrtc", r.UserAgent())
 		if err != nil {
 			http.Error(w, err.Error(), http.StatusInternalServerError)
 			return

internal/hass/hass.go

@@ -45,19 +45,14 @@ func Init() {
 		return "", nil
 	})
 
-	streams.HandleFunc("hass", func(url string) (core.Producer, error) {
+	streams.HandleFunc("hass", func(source string) (core.Producer, error) {
 		// support hass://supervisor?entity_id=camera.driveway_doorbell
-		client, err := hass.NewClient(url)
-		if err != nil {
-			return nil, err
-		}
-
-		return client, nil
+		return hass.NewClient(source)
 	})
 
 	// load static entries from Hass config
 	if err := importConfig(conf.Mod.Config); err != nil {
-		log.Debug().Msgf("[hass] can't import config: %s", err)
+		log.Trace().Msgf("[hass] can't import config: %s", err)
 
 		api.HandleFunc("api/hass", func(w http.ResponseWriter, _ *http.Request) {
 			http.Error(w, "no hass config", http.StatusNotFound)

internal/hls/hls.go

@@ -12,7 +12,6 @@ import (
 	"github.com/AlexxIT/go2rtc/pkg/core"
 	"github.com/AlexxIT/go2rtc/pkg/mp4"
 	"github.com/AlexxIT/go2rtc/pkg/mpegts"
-	"github.com/AlexxIT/go2rtc/pkg/tcp"
 	"github.com/rs/zerolog"
 )
 
@@ -63,15 +62,13 @@ func handlerStream(w http.ResponseWriter, r *http.Request) {
 	medias := mp4.ParseQuery(r.URL.Query())
 	if medias != nil {
 		c := mp4.NewConsumer(medias)
-		c.Type = "HLS/fMP4 consumer"
-		c.RemoteAddr = tcp.RemoteAddr(r)
-		c.UserAgent = r.UserAgent()
+		c.FormatName = "hls/fmp4"
+		c.WithRequest(r)
 		cons = c
 	} else {
 		c := mpegts.NewConsumer()
-		c.Type = "HLS/TS consumer"
-		c.RemoteAddr = tcp.RemoteAddr(r)
-		c.UserAgent = r.UserAgent()
+		c.FormatName = "hls/mpegts"
+		c.WithRequest(r)
 		cons = c
 	}
 

internal/hls/ws.go

@@ -8,7 +8,6 @@ import (
 	"github.com/AlexxIT/go2rtc/internal/api/ws"
 	"github.com/AlexxIT/go2rtc/internal/streams"
 	"github.com/AlexxIT/go2rtc/pkg/mp4"
-	"github.com/AlexxIT/go2rtc/pkg/tcp"
 )
 
 func handlerWSHLS(tr *ws.Transport, msg *ws.Message) error {
@@ -20,9 +19,8 @@ func handlerWSHLS(tr *ws.Transport, msg *ws.Message) error {
 	codecs := msg.String()
 	medias := mp4.ParseCodecs(codecs, true)
 	cons := mp4.NewConsumer(medias)
-	cons.Type = "HLS/fMP4 consumer"
-	cons.RemoteAddr = tcp.RemoteAddr(tr.Request)
-	cons.UserAgent = tr.Request.UserAgent()
+	cons.FormatName = "hls/fmp4"
+	cons.WithRequest(tr.Request)
 
 	log.Trace().Msgf("[hls] new ws consumer codecs=%s", codecs)
 

internal/homekit/api.go

@@ -103,7 +103,7 @@ func apiPair(id, url string) error {
 
 	streams.New(id, conn.URL())
 
-	return app.PatchConfig(id, conn.URL(), "streams")
+	return app.PatchConfig([]string{"streams", id}, conn.URL())
 }
 
 func apiUnpair(id string) error {
@@ -112,7 +112,7 @@ func apiUnpair(id string) error {
 		return errors.New(api.StreamNotFound)
 	}
 
-	rawURL := findHomeKitURL(stream)
+	rawURL := findHomeKitURL(stream.Sources())
 	if rawURL == "" {
 		return errors.New("not homekit source")
 	}
@@ -123,15 +123,15 @@ func apiUnpair(id string) error {
 
 	streams.Delete(id)
 
-	return app.PatchConfig(id, nil, "streams")
+	return app.PatchConfig([]string{"streams", id}, nil)
 }
 
 func findHomeKitURLs() map[string]*url.URL {
 	urls := map[string]*url.URL{}
-	for id, stream := range streams.Streams() {
-		if rawURL := findHomeKitURL(stream); rawURL != "" {
+	for name, sources := range streams.GetAllSources() {
+		if rawURL := findHomeKitURL(sources); rawURL != "" {
 			if u, err := url.Parse(rawURL); err == nil {
-				urls[id] = u
+				urls[name] = u
 			}
 		}
 	}

internal/homekit/homekit.go

@@ -79,7 +79,7 @@ func Init() {
 			Handler:       homekit.ServerHandler(srv),
 		}
 
-		if url := findHomeKitURL(stream); url != "" {
+		if url := findHomeKitURL(stream.Sources()); url != "" {
 			// 1. Act as transparent proxy for HomeKit camera
 			dial := func() (net.Conn, error) {
 				client, err := homekit.Dial(url, srtp.Server)
@@ -118,8 +118,8 @@ func Init() {
 		servers[host] = srv
 	}
 
-	api.HandleFunc(hap.PathPairSetup, hapPairSetup)
-	api.HandleFunc(hap.PathPairVerify, hapPairVerify)
+	api.HandleFunc(hap.PathPairSetup, hapHandler)
+	api.HandleFunc(hap.PathPairVerify, hapHandler)
 
 	log.Trace().Msgf("[homekit] mdns: %s", entries)
 
@@ -133,21 +133,34 @@ func Init() {
 var log zerolog.Logger
 var servers map[string]*server
 
-func streamHandler(url string) (core.Producer, error) {
+func streamHandler(rawURL string) (core.Producer, error) {
 	if srtp.Server == nil {
 		return nil, errors.New("homekit: can't work without SRTP server")
 	}
 
-	return homekit.Dial(url, srtp.Server)
-}
-
-func hapPairSetup(w http.ResponseWriter, r *http.Request) {
-	srv, ok := servers[r.Host]
-	if !ok {
-		log.Error().Msg("[homekit] unknown host: " + r.Host)
-		return
+	rawURL, rawQuery, _ := strings.Cut(rawURL, "#")
+	client, err := homekit.Dial(rawURL, srtp.Server)
+	if client != nil && rawQuery != "" {
+		query := streams.ParseQuery(rawQuery)
+		client.Bitrate = parseBitrate(query.Get("bitrate"))
 	}
 
+	return client, err
+}
+
+func resolve(host string) *server {
+	if len(servers) == 1 {
+		for _, srv := range servers {
+			return srv
+		}
+	}
+	if srv, ok := servers[host]; ok {
+		return srv
+	}
+	return nil
+}
+
+func hapHandler(w http.ResponseWriter, r *http.Request) {
 	conn, rw, err := w.(http.Hijacker).Hijack()
 	if err != nil {
 		return
@@ -155,32 +168,29 @@ func hapPairSetup(w http.ResponseWriter, r *http.Request) {
 
 	defer conn.Close()
 
-	if err = srv.hap.PairSetup(r, rw, conn); err != nil {
-		log.Error().Err(err).Caller().Send()
-	}
-}
-
-func hapPairVerify(w http.ResponseWriter, r *http.Request) {
-	srv, ok := servers[r.Host]
-	if !ok {
+	// Can support multiple HomeKit cameras on single port ONLY for Apple devices.
+	// Doesn't support Home Assistant and any other open source projects
+	// because they don't send the host header in requests.
+	srv := resolve(r.Host)
+	if srv == nil {
 		log.Error().Msg("[homekit] unknown host: " + r.Host)
+		_ = hap.WriteBackoff(rw)
 		return
 	}
 
-	conn, rw, err := w.(http.Hijacker).Hijack()
-	if err != nil {
-		return
+	switch r.RequestURI {
+	case hap.PathPairSetup:
+		err = srv.hap.PairSetup(r, rw, conn)
+	case hap.PathPairVerify:
+		err = srv.hap.PairVerify(r, rw, conn)
 	}
 
-	defer conn.Close()
-
-	if err = srv.hap.PairVerify(r, rw, conn); err != nil && err != io.EOF {
+	if err != nil && err != io.EOF {
 		log.Error().Err(err).Caller().Send()
 	}
 }
 
-func findHomeKitURL(stream *streams.Stream) string {
-	sources := stream.Sources()
+func findHomeKitURL(sources []string) string {
 	if len(sources) == 0 {
 		return ""
 	}
@@ -199,3 +209,24 @@ func findHomeKitURL(stream *streams.Stream) string {
 
 	return ""
 }
+
+func parseBitrate(s string) int {
+	n := len(s)
+	if n == 0 {
+		return 0
+	}
+
+	var k int
+	switch n--; s[n] {
+	case 'K':
+		k = 1024
+		s = s[:n]
+	case 'M':
+		k = 1024 * 1024
+		s = s[:n]
+	default:
+		k = 1
+	}
+
+	return k * core.Atoi(s)
+}

internal/homekit/server.go

@@ -87,7 +87,7 @@ func (s *server) SetCharacteristic(conn net.Conn, aid uint8, iid uint64, value a
 	switch char.Type {
 	case camera.TypeSetupEndpoints:
 		var offer camera.SetupEndpoints
-		if err := tlv8.UnmarshalBase64(value.(string), &offer); err != nil {
+		if err := tlv8.UnmarshalBase64(value, &offer); err != nil {
 			return
 		}
 
@@ -96,7 +96,7 @@ func (s *server) SetCharacteristic(conn net.Conn, aid uint8, iid uint64, value a
 
 	case camera.TypeSelectedStreamConfiguration:
 		var conf camera.SelectedStreamConfig
-		if err := tlv8.UnmarshalBase64(value.(string), &conf); err != nil {
+		if err := tlv8.UnmarshalBase64(value, &conf); err != nil {
 			return
 		}
 
@@ -222,7 +222,7 @@ func (s *server) DelPair(conn net.Conn, id string) {
 }
 
 func (s *server) PatchConfig() {
-	if err := app.PatchConfig("pairings", s.pairings, "homekit", s.stream); err != nil {
+	if err := app.PatchConfig([]string{"homekit", s.stream, "pairings"}, s.pairings); err != nil {
 		log.Error().Err(err).Msgf(
 			"[homekit] can't save %s pairings=%v", s.stream, s.pairings,
 		)

internal/http/http.go

@@ -11,9 +11,10 @@ import (
 	"github.com/AlexxIT/go2rtc/internal/streams"
 	"github.com/AlexxIT/go2rtc/pkg/core"
 	"github.com/AlexxIT/go2rtc/pkg/hls"
+	"github.com/AlexxIT/go2rtc/pkg/image"
 	"github.com/AlexxIT/go2rtc/pkg/magic"
-	"github.com/AlexxIT/go2rtc/pkg/mjpeg"
-	"github.com/AlexxIT/go2rtc/pkg/multipart"
+	"github.com/AlexxIT/go2rtc/pkg/mpjpeg"
+	"github.com/AlexxIT/go2rtc/pkg/pcm"
 	"github.com/AlexxIT/go2rtc/pkg/tcp"
 )
 
@@ -45,6 +46,21 @@ func handleHTTP(rawURL string) (core.Producer, error) {
 		}
 	}
 
+	prod, err := do(req)
+	if err != nil {
+		return nil, err
+	}
+
+	if info, ok := prod.(core.Info); ok {
+		info.SetProtocol("http")
+		info.SetRemoteAddr(req.URL.Host) // TODO: rewrite to net.Conn
+		info.SetURL(rawURL)
+	}
+
+	return prod, nil
+}
+
+func do(req *http.Request) (core.Producer, error) {
 	res, err := tcp.Do(req)
 	if err != nil {
 		return nil, err
@@ -66,14 +82,15 @@ func handleHTTP(rawURL string) (core.Producer, error) {
 	}
 
 	switch {
-	case ct == "image/jpeg":
-		return mjpeg.NewClient(res), nil
-
-	case ct == "multipart/x-mixed-replace":
-		return multipart.Open(res.Body)
-
 	case ct == "application/vnd.apple.mpegurl" || ext == "m3u8":
 		return hls.OpenURL(req.URL, res.Body)
+	case ct == "image/jpeg":
+		return image.Open(res)
+	case ct == "multipart/x-mixed-replace":
+		return mpjpeg.Open(res.Body)
+	//https://www.iana.org/assignments/media-types/audio/basic
+	case ct == "audio/basic":
+		return pcm.Open(res.Body)
 	}
 
 	return magic.Open(res.Body)

internal/isapi/init.go

@@ -7,16 +7,7 @@ import (
 )
 
 func Init() {
-	streams.HandleFunc("isapi", handle)
-}
-
-func handle(url string) (core.Producer, error) {
-	conn, err := isapi.NewClient(url)
-	if err != nil {
-		return nil, err
-	}
-	if err = conn.Dial(); err != nil {
-		return nil, err
-	}
-	return conn, nil
+	streams.HandleFunc("isapi", func(source string) (core.Producer, error) {
+		return isapi.Dial(source)
+	})
 }

internal/ivideon/ivideon.go

@@ -2,18 +2,9 @@ package ivideon
 
 import (
 	"github.com/AlexxIT/go2rtc/internal/streams"
-	"github.com/AlexxIT/go2rtc/pkg/core"
 	"github.com/AlexxIT/go2rtc/pkg/ivideon"
-	"strings"
 )
 
 func Init() {
-	streams.HandleFunc("ivideon", func(url string) (core.Producer, error) {
-		id := strings.Replace(url[8:], "/", ":", 1)
-		prod := ivideon.NewClient(id)
-		if err := prod.Dial(); err != nil {
-			return nil, err
-		}
-		return prod, nil
-	})
+	streams.HandleFunc("ivideon", ivideon.Dial)
 }

internal/mjpeg/init.go

@@ -10,35 +10,40 @@ import (
 
 	"github.com/AlexxIT/go2rtc/internal/api"
 	"github.com/AlexxIT/go2rtc/internal/api/ws"
+	"github.com/AlexxIT/go2rtc/internal/app"
 	"github.com/AlexxIT/go2rtc/internal/ffmpeg"
 	"github.com/AlexxIT/go2rtc/internal/streams"
 	"github.com/AlexxIT/go2rtc/pkg/ascii"
 	"github.com/AlexxIT/go2rtc/pkg/core"
 	"github.com/AlexxIT/go2rtc/pkg/magic"
 	"github.com/AlexxIT/go2rtc/pkg/mjpeg"
-	"github.com/AlexxIT/go2rtc/pkg/tcp"
-	"github.com/rs/zerolog/log"
+	"github.com/AlexxIT/go2rtc/pkg/mpjpeg"
+	"github.com/AlexxIT/go2rtc/pkg/y4m"
+	"github.com/rs/zerolog"
 )
 
 func Init() {
 	api.HandleFunc("api/frame.jpeg", handlerKeyframe)
 	api.HandleFunc("api/stream.mjpeg", handlerStream)
 	api.HandleFunc("api/stream.ascii", handlerStream)
+	api.HandleFunc("api/stream.y4m", apiStreamY4M)
 
 	ws.HandleFunc("mjpeg", handlerWS)
+
+	log = app.GetLogger("mjpeg")
 }
 
+var log zerolog.Logger
+
 func handlerKeyframe(w http.ResponseWriter, r *http.Request) {
-	src := r.URL.Query().Get("src")
-	stream := streams.Get(src)
+	stream := streams.GetOrPatch(r.URL.Query())
 	if stream == nil {
 		http.Error(w, api.StreamNotFound, http.StatusNotFound)
 		return
 	}
 
 	cons := magic.NewKeyframe()
-	cons.RemoteAddr = tcp.RemoteAddr(r)
-	cons.UserAgent = r.UserAgent()
+	cons.WithRequest(r)
 
 	if err := stream.AddConsumer(cons); err != nil {
 		log.Error().Err(err).Caller().Send()
@@ -93,8 +98,7 @@ func outputMjpeg(w http.ResponseWriter, r *http.Request) {
 	}
 
 	cons := mjpeg.NewConsumer()
-	cons.RemoteAddr = tcp.RemoteAddr(r)
-	cons.UserAgent = r.UserAgent()
+	cons.WithRequest(r)
 
 	if err := stream.AddConsumer(cons); err != nil {
 		log.Error().Err(err).Msg("[api.mjpeg] add consumer")
@@ -110,7 +114,7 @@ func outputMjpeg(w http.ResponseWriter, r *http.Request) {
 		wr := mjpeg.NewWriter(w)
 		_, _ = cons.WriteTo(wr)
 	} else {
-		cons.Type = "ASCII passive consumer "
+		cons.FormatName = "ascii"
 
 		query := r.URL.Query()
 		wr := ascii.NewWriter(w, query.Get("color"), query.Get("back"), query.Get("text"))
@@ -128,17 +132,16 @@ func inputMjpeg(w http.ResponseWriter, r *http.Request) {
 		return
 	}
 
-	res := &http.Response{Body: r.Body, Header: r.Header, Request: r}
-	res.Header.Set("Content-Type", "multipart/mixed;boundary=")
+	prod, _ := mpjpeg.Open(r.Body)
+	prod.WithRequest(r)
 
-	client := mjpeg.NewClient(res)
-	stream.AddProducer(client)
+	stream.AddProducer(prod)
 
-	if err := client.Start(); err != nil && err != io.EOF {
+	if err := prod.Start(); err != nil && err != io.EOF {
 		log.Warn().Err(err).Caller().Send()
 	}
 
-	stream.RemoveProducer(client)
+	stream.RemoveProducer(prod)
 }
 
 func handlerWS(tr *ws.Transport, _ *ws.Message) error {
@@ -148,8 +151,7 @@ func handlerWS(tr *ws.Transport, _ *ws.Message) error {
 	}
 
 	cons := mjpeg.NewConsumer()
-	cons.RemoteAddr = tcp.RemoteAddr(tr.Request)
-	cons.UserAgent = tr.Request.UserAgent()
+	cons.WithRequest(tr.Request)
 
 	if err := stream.AddConsumer(cons); err != nil {
 		log.Debug().Err(err).Msg("[mjpeg] add consumer")
@@ -166,3 +168,24 @@ func handlerWS(tr *ws.Transport, _ *ws.Message) error {
 
 	return nil
 }
+
+func apiStreamY4M(w http.ResponseWriter, r *http.Request) {
+	src := r.URL.Query().Get("src")
+	stream := streams.Get(src)
+	if stream == nil {
+		http.Error(w, api.StreamNotFound, http.StatusNotFound)
+		return
+	}
+
+	cons := y4m.NewConsumer()
+	cons.WithRequest(r)
+
+	if err := stream.AddConsumer(cons); err != nil {
+		log.Error().Err(err).Caller().Send()
+		return
+	}
+
+	_, _ = cons.WriteTo(w)
+
+	stream.RemoveConsumer(cons)
+}

internal/mp4/mp4.go

@@ -1,6 +1,7 @@
 package mp4
 
 import (
+	"context"
 	"net/http"
 	"strconv"
 	"strings"
@@ -12,7 +13,6 @@ import (
 	"github.com/AlexxIT/go2rtc/internal/streams"
 	"github.com/AlexxIT/go2rtc/pkg/core"
 	"github.com/AlexxIT/go2rtc/pkg/mp4"
-	"github.com/AlexxIT/go2rtc/pkg/tcp"
 	"github.com/rs/zerolog"
 )
 
@@ -99,9 +99,9 @@ func handlerMP4(w http.ResponseWriter, r *http.Request) {
 
 	medias := mp4.ParseQuery(r.URL.Query())
 	cons := mp4.NewConsumer(medias)
-	cons.Type = "MP4/HTTP active consumer"
-	cons.RemoteAddr = tcp.RemoteAddr(r)
-	cons.UserAgent = r.UserAgent()
+	cons.FormatName = "mp4"
+	cons.Protocol = "http"
+	cons.WithRequest(r)
 
 	if err := stream.AddConsumer(cons); err != nil {
 		log.Error().Err(err).Caller().Send()
@@ -127,20 +127,20 @@ func handlerMP4(w http.ResponseWriter, r *http.Request) {
 		header.Set("Content-Disposition", `attachment; filename="`+filename+`"`)
 	}
 
-	var duration *time.Timer
-	if s := query.Get("duration"); s != "" {
-		if i, _ := strconv.Atoi(s); i > 0 {
-			duration = time.AfterFunc(time.Second*time.Duration(i), func() {
-				_ = cons.Stop()
-			})
-		}
+	ctx := r.Context() // handle when the client drops the connection
+
+	if i := core.Atoi(query.Get("duration")); i > 0 {
+		timeout := time.Second * time.Duration(i)
+		var cancel context.CancelFunc
+		ctx, cancel = context.WithTimeout(ctx, timeout)
+		defer cancel()
 	}
 
+	go func() {
+		<-ctx.Done()
+		_ = cons.Stop()
+		stream.RemoveConsumer(cons)
+	}()
+
 	_, _ = cons.WriteTo(w)
-
-	stream.RemoveConsumer(cons)
-
-	if duration != nil {
-		duration.Stop()
-	}
 }

internal/mp4/ws.go

@@ -8,7 +8,6 @@ import (
 	"github.com/AlexxIT/go2rtc/internal/streams"
 	"github.com/AlexxIT/go2rtc/pkg/core"
 	"github.com/AlexxIT/go2rtc/pkg/mp4"
-	"github.com/AlexxIT/go2rtc/pkg/tcp"
 )
 
 func handlerWSMSE(tr *ws.Transport, msg *ws.Message) error {
@@ -24,9 +23,8 @@ func handlerWSMSE(tr *ws.Transport, msg *ws.Message) error {
 	}
 
 	cons := mp4.NewConsumer(medias)
-	cons.Type = "MSE/WebSocket active consumer"
-	cons.RemoteAddr = tcp.RemoteAddr(tr.Request)
-	cons.UserAgent = tr.Request.UserAgent()
+	cons.FormatName = "mse/fmp4"
+	cons.WithRequest(tr.Request)
 
 	if err := stream.AddConsumer(cons); err != nil {
 		log.Debug().Err(err).Msg("[mp4] add consumer")
@@ -57,9 +55,7 @@ func handlerWSMP4(tr *ws.Transport, msg *ws.Message) error {
 	}
 
 	cons := mp4.NewKeyframe(medias)
-	cons.Type = "MP4/WebSocket active consumer"
-	cons.RemoteAddr = tcp.RemoteAddr(tr.Request)
-	cons.UserAgent = tr.Request.UserAgent()
+	cons.WithRequest(tr.Request)
 
 	if err := stream.AddConsumer(cons); err != nil {
 		log.Error().Err(err).Caller().Send()

internal/mpegts/aac.go

@@ -6,8 +6,6 @@ import (
 	"github.com/AlexxIT/go2rtc/internal/api"
 	"github.com/AlexxIT/go2rtc/internal/streams"
 	"github.com/AlexxIT/go2rtc/pkg/aac"
-	"github.com/AlexxIT/go2rtc/pkg/tcp"
-	"github.com/rs/zerolog/log"
 )
 
 func apiStreamAAC(w http.ResponseWriter, r *http.Request) {
@@ -19,11 +17,9 @@ func apiStreamAAC(w http.ResponseWriter, r *http.Request) {
 	}
 
 	cons := aac.NewConsumer()
-	cons.RemoteAddr = tcp.RemoteAddr(r)
-	cons.UserAgent = r.UserAgent()
+	cons.WithRequest(r)
 
 	if err := stream.AddConsumer(cons); err != nil {
-		log.Error().Err(err).Caller().Send()
 		http.Error(w, err.Error(), http.StatusInternalServerError)
 		return
 	}

internal/mpegts/mpegts.go

@@ -6,8 +6,6 @@ import (
 	"github.com/AlexxIT/go2rtc/internal/api"
 	"github.com/AlexxIT/go2rtc/internal/streams"
 	"github.com/AlexxIT/go2rtc/pkg/mpegts"
-	"github.com/AlexxIT/go2rtc/pkg/tcp"
-	"github.com/rs/zerolog/log"
 )
 
 func Init() {
@@ -32,11 +30,9 @@ func outputMpegTS(w http.ResponseWriter, r *http.Request) {
 	}
 
 	cons := mpegts.NewConsumer()
-	cons.RemoteAddr = tcp.RemoteAddr(r)
-	cons.UserAgent = r.UserAgent()
+	cons.WithRequest(r)
 
 	if err := stream.AddConsumer(cons); err != nil {
-		log.Error().Err(err).Caller().Send()
 		http.Error(w, err.Error(), http.StatusInternalServerError)
 		return
 	}

internal/nest/init.go

@@ -2,6 +2,7 @@ package nest
 
 import (
 	"net/http"
+	"strings"
 
 	"github.com/AlexxIT/go2rtc/internal/api"
 	"github.com/AlexxIT/go2rtc/internal/streams"
@@ -10,19 +11,13 @@ import (
 )
 
 func Init() {
-	streams.HandleFunc("nest", streamNest)
+	streams.HandleFunc("nest", func(source string) (core.Producer, error) {
+		return nest.Dial(source)
+	})
 
 	api.HandleFunc("api/nest", apiNest)
 }
 
-func streamNest(url string) (core.Producer, error) {
-	client, err := nest.NewClient(url)
-	if err != nil {
-		return nil, err
-	}
-	return client, nil
-}
-
 func apiNest(w http.ResponseWriter, r *http.Request) {
 	query := r.URL.Query()
 	cliendID := query.Get("client_id")
@@ -44,11 +39,12 @@ func apiNest(w http.ResponseWriter, r *http.Request) {
 
 	var items []*api.Source
 
-	for name, deviceID := range devices {
-		query.Set("device_id", deviceID)
+	for _, device := range devices {
+		query.Set("device_id", device.DeviceID)
+		query.Set("protocols", strings.Join(device.Protocols, ","))
 
 		items = append(items, &api.Source{
-			Name: name, URL: "nest:?" + query.Encode(),
+			Name: device.Name, URL: "nest:?" + query.Encode(),
 		})
 	}
 

internal/onvif/README.md

@@ -0,0 +1,25 @@
+# ONVIF
+
+A regular camera has a single video source (`GetVideoSources`) and two profiles (`GetProfiles`).
+
+Go2rtc has one video source and one profile per stream.
+
+## Tested clients
+
+Go2rtc works as ONVIF server:
+
+- Happytime onvif client (windows)
+- Home Assistant ONVIF integration (linux)
+- Onvier (android)
+- ONVIF Device Manager (windows)
+
+PS. Support only TCP transport for RTSP protocol. UDP and HTTP transports - unsupported yet.
+
+## Tested cameras
+
+Go2rtc works as ONVIF client:
+
+- Dahua IPC-K42
+- OpenIPC
+- Reolink RLC-520A
+- TP-Link Tapo TC60

internal/onvif/onvif.go

@@ -55,49 +55,73 @@ func onvifDeviceService(w http.ResponseWriter, r *http.Request) {
 		return
 	}
 
-	action := onvif.GetRequestAction(b)
-	if action == "" {
+	operation := onvif.GetRequestAction(b)
+	if operation == "" {
 		http.Error(w, "malformed request body", http.StatusBadRequest)
 		return
 	}
 
-	log.Trace().Msgf("[onvif] %s", action)
+	log.Trace().Msgf("[onvif] server request %s %s:\n%s", r.Method, r.RequestURI, b)
 
-	var res string
+	switch operation {
+	case onvif.DeviceGetNetworkInterfaces, // important for Hass
+		onvif.DeviceGetSystemDateAndTime, // important for Hass
+		onvif.DeviceGetDiscoveryMode,
+		onvif.DeviceGetDNS,
+		onvif.DeviceGetHostname,
+		onvif.DeviceGetNetworkDefaultGateway,
+		onvif.DeviceGetNetworkProtocols,
+		onvif.DeviceGetNTP,
+		onvif.DeviceGetScopes,
+		onvif.MediaGetVideoEncoderConfigurations,
+		onvif.MediaGetAudioEncoderConfigurations,
+		onvif.MediaGetAudioSources,
+		onvif.MediaGetAudioSourceConfigurations:
+		b = onvif.StaticResponse(operation)
 
-	switch action {
-	case onvif.ActionGetCapabilities:
+	case onvif.DeviceGetCapabilities:
 		// important for Hass: Media section
-		res = onvif.GetCapabilitiesResponse(r.Host)
+		b = onvif.GetCapabilitiesResponse(r.Host)
 
-	case onvif.ActionGetSystemDateAndTime:
-		// important for Hass
-		res = onvif.GetSystemDateAndTimeResponse()
+	case onvif.DeviceGetServices:
+		b = onvif.GetServicesResponse(r.Host)
 
-	case onvif.ActionGetNetworkInterfaces:
-		// important for Hass: none
-		res = onvif.GetNetworkInterfacesResponse()
-
-	case onvif.ActionGetDeviceInformation:
+	case onvif.DeviceGetDeviceInformation:
 		// important for Hass: SerialNumber (unique server ID)
-		res = onvif.GetDeviceInformationResponse("", "go2rtc", app.Version, r.Host)
+		b = onvif.GetDeviceInformationResponse("", "go2rtc", app.Version, r.Host)
 
-	case onvif.ActionGetServiceCapabilities:
+	case onvif.ServiceGetServiceCapabilities:
 		// important for Hass
-		res = onvif.GetServiceCapabilitiesResponse()
+		// TODO: check path links to media
+		b = onvif.GetMediaServiceCapabilitiesResponse()
 
-	case onvif.ActionSystemReboot:
-		res = onvif.SystemRebootResponse()
+	case onvif.DeviceSystemReboot:
+		b = onvif.StaticResponse(operation)
 
 		time.AfterFunc(time.Second, func() {
 			os.Exit(0)
 		})
 
-	case onvif.ActionGetProfiles:
-		// important for Hass: H264 codec, width, height
-		res = onvif.GetProfilesResponse(streams.GetAll())
+	case onvif.MediaGetVideoSources:
+		b = onvif.GetVideoSourcesResponse(streams.GetAllNames())
 
-	case onvif.ActionGetStreamUri:
+	case onvif.MediaGetProfiles:
+		// important for Hass: H264 codec, width, height
+		b = onvif.GetProfilesResponse(streams.GetAllNames())
+
+	case onvif.MediaGetProfile:
+		token := onvif.FindTagValue(b, "ProfileToken")
+		b = onvif.GetProfileResponse(token)
+
+	case onvif.MediaGetVideoSourceConfigurations:
+		// important for Happytime Onvif Client
+		b = onvif.GetVideoSourceConfigurationsResponse(streams.GetAllNames())
+
+	case onvif.MediaGetVideoSourceConfiguration:
+		token := onvif.FindTagValue(b, "ConfigurationToken")
+		b = onvif.GetVideoSourceConfigurationResponse(token)
+
+	case onvif.MediaGetStreamUri:
 		host, _, err := net.SplitHostPort(r.Host)
 		if err != nil {
 			http.Error(w, err.Error(), http.StatusInternalServerError)
@@ -105,16 +129,23 @@ func onvifDeviceService(w http.ResponseWriter, r *http.Request) {
 		}
 
 		uri := "rtsp://" + host + ":" + rtsp.Port + "/" + onvif.FindTagValue(b, "ProfileToken")
-		res = onvif.GetStreamUriResponse(uri)
+		b = onvif.GetStreamUriResponse(uri)
+
+	case onvif.MediaGetSnapshotUri:
+		uri := "http://" + r.Host + "/api/frame.jpeg?src=" + onvif.FindTagValue(b, "ProfileToken")
+		b = onvif.GetSnapshotUriResponse(uri)
 
 	default:
-		http.Error(w, "unsupported action", http.StatusBadRequest)
+		http.Error(w, "unsupported operation", http.StatusBadRequest)
+		log.Warn().Msgf("[onvif] unsupported operation: %s", operation)
 		log.Debug().Msgf("[onvif] unsupported request:\n%s", b)
 		return
 	}
 
+	log.Trace().Msgf("[onvif] server response:\n%s", b)
+
 	w.Header().Set("Content-Type", "application/soap+xml; charset=utf-8")
-	if _, err = w.Write([]byte(res)); err != nil {
+	if _, err = w.Write(b); err != nil {
 		log.Error().Err(err).Caller().Send()
 	}
 }
@@ -160,7 +191,7 @@ func apiOnvif(w http.ResponseWriter, r *http.Request) {
 		}
 
 		if l := log.Trace(); l.Enabled() {
-			b, _ := client.GetProfiles()
+			b, _ := client.MediaRequest(onvif.MediaGetProfiles)
 			l.Msgf("[onvif] src=%s profiles:\n%s", src, b)
 		}
 

internal/ring/ring.go

@@ -0,0 +1,102 @@
+package ring
+
+import (
+	"encoding/json"
+	"net/http"
+	"net/url"
+
+	"github.com/AlexxIT/go2rtc/internal/api"
+	"github.com/AlexxIT/go2rtc/internal/streams"
+	"github.com/AlexxIT/go2rtc/pkg/core"
+	"github.com/AlexxIT/go2rtc/pkg/ring"
+)
+
+func Init() {
+	streams.HandleFunc("ring", func(source string) (core.Producer, error) {
+		return ring.Dial(source)
+	})
+
+	api.HandleFunc("api/ring", apiRing)
+}
+
+func apiRing(w http.ResponseWriter, r *http.Request) {
+	query := r.URL.Query()
+	var ringAPI *ring.RingRestClient
+	var err error
+
+	// Check auth method
+	if email := query.Get("email"); email != "" {
+		// Email/Password Flow
+		password := query.Get("password")
+		code := query.Get("code")
+
+		ringAPI, err = ring.NewRingRestClient(ring.EmailAuth{
+			Email:    email,
+			Password: password,
+		}, nil)
+
+		if err != nil {
+			http.Error(w, err.Error(), http.StatusInternalServerError)
+			return
+		}
+
+		// Try authentication (this will trigger 2FA if needed)
+		if _, err = ringAPI.GetAuth(code); err != nil {
+			if ringAPI.Using2FA {
+				// Return 2FA prompt
+				json.NewEncoder(w).Encode(map[string]interface{}{
+					"needs_2fa": true,
+					"prompt":    ringAPI.PromptFor2FA,
+				})
+				return
+			}
+			http.Error(w, err.Error(), http.StatusInternalServerError)
+			return
+		}
+	} else {
+		// Refresh Token Flow
+		refreshToken := query.Get("refresh_token")
+		if refreshToken == "" {
+			http.Error(w, "either email/password or refresh_token is required", http.StatusBadRequest)
+			return
+		}
+
+		ringAPI, err = ring.NewRingRestClient(ring.RefreshTokenAuth{
+			RefreshToken: refreshToken,
+		}, nil)
+		if err != nil {
+			http.Error(w, err.Error(), http.StatusInternalServerError)
+			return
+		}
+	}
+
+	// Fetch devices
+	devices, err := ringAPI.FetchRingDevices()
+	if err != nil {
+		http.Error(w, err.Error(), http.StatusInternalServerError)
+		return
+	}
+
+	// Create clean query with only required parameters
+	cleanQuery := url.Values{}
+	cleanQuery.Set("refresh_token", ringAPI.RefreshToken)
+
+	var items []*api.Source
+	for _, camera := range devices.AllCameras {
+		cleanQuery.Set("device_id", camera.DeviceID)
+
+		// Stream source
+		items = append(items, &api.Source{
+			Name: camera.Description,
+			URL:  "ring:?" + cleanQuery.Encode(),
+		})
+
+		// Snapshot source
+		items = append(items, &api.Source{
+			Name: camera.Description + " Snapshot",
+			URL:  "ring:?" + cleanQuery.Encode() + "&snapshot",
+		})
+	}
+
+	api.ResponseSources(w, items)
+}

internal/roborock/roborock.go

@@ -11,22 +11,13 @@ import (
 )
 
 func Init() {
-	streams.HandleFunc("roborock", handle)
+	streams.HandleFunc("roborock", func(source string) (core.Producer, error) {
+		return roborock.Dial(source)
+	})
 
 	api.HandleFunc("api/roborock", apiHandle)
 }
 
-func handle(url string) (core.Producer, error) {
-	conn := roborock.NewClient(url)
-	if err := conn.Dial(); err != nil {
-		return nil, err
-	}
-	if err := conn.Connect(); err != nil {
-		return nil, err
-	}
-	return conn, nil
-}
-
 var Auth struct {
 	UserData *roborock.UserInfo `json:"user_data"`
 	BaseURL  string             `json:"base_url"`

internal/rtmp/README.md

@@ -0,0 +1,60 @@
+## Tested client
+
+| From   | To                              | Comment |
+|--------|---------------------------------|---------|
+| go2rtc | Reolink RLC-520A fw. v3.1.0.801 | OK      |
+
+**go2rtc.yaml**
+
+```yaml
+streams:
+  rtmp-reolink1: rtmp://192.168.10.92/bcs/channel0_main.bcs?channel=0&stream=0&user=admin&password=password
+  rtmp-reolink2: rtmp://192.168.10.92/bcs/channel0_sub.bcs?channel=0&stream=1&user=admin&password=password
+  rtmp-reolink3: rtmp://192.168.10.92/bcs/channel0_ext.bcs?channel=0&stream=1&user=admin&password=password
+```
+
+## Tested server
+
+| From                   | To     | Comment             |
+|------------------------|--------|---------------------|
+| OBS 31.0.2             | go2rtc | OK                  |
+| OpenIPC 2.5.03.02-lite | go2rtc | OK                  |
+| FFmpeg 6.1             | go2rtc | OK                  |
+| GoPro Black 12         | go2rtc | OK, 1080p, 5000kbps |
+
+**go2rtc.yaml**
+
+```yaml
+rtmp:
+  listen: :1935
+streams:
+  tmp:
+```
+
+**OBS**
+ 
+Settings > Stream:
+
+- Service: Custom
+- Server: rtmp://192.168.10.101/tmp
+- Stream Key: <empty>
+- Use auth: <disabled>
+
+**OpenIPC**
+
+WebUI > Majestic > Settings > Outgoing
+
+- Enable
+- Address: rtmp://192.168.10.101/tmp
+- Save
+- Restart
+
+**FFmpeg**
+
+```shell
+ffmpeg -re -i bbb.mp4 -c copy -f flv rtmp://192.168.10.101/tmp
+```
+
+**GoPro**
+
+GoPro Quik > Camera > Translation > Other

internal/rtmp/rtmp.go

@@ -12,7 +12,6 @@ import (
 	"github.com/AlexxIT/go2rtc/pkg/core"
 	"github.com/AlexxIT/go2rtc/pkg/flv"
 	"github.com/AlexxIT/go2rtc/pkg/rtmp"
-	"github.com/AlexxIT/go2rtc/pkg/tcp"
 	"github.com/rs/zerolog"
 )
 
@@ -128,17 +127,13 @@ func tcpHandle(netConn net.Conn) error {
 var log zerolog.Logger
 
 func streamsHandle(url string) (core.Producer, error) {
-	client, err := rtmp.DialPlay(url)
-	if err != nil {
-		return nil, err
-	}
-	return client, nil
+	return rtmp.DialPlay(url)
 }
 
 func streamsConsumerHandle(url string) (core.Consumer, func(), error) {
 	cons := flv.NewConsumer()
 	run := func() {
-		wr, err := rtmp.DialPublish(url)
+		wr, err := rtmp.DialPublish(url, cons)
 		if err != nil {
 			return
 		}
@@ -165,9 +160,7 @@ func outputFLV(w http.ResponseWriter, r *http.Request) {
 	}
 
 	cons := flv.NewConsumer()
-	cons.Type = "HTTP-FLV consumer"
-	cons.RemoteAddr = tcp.RemoteAddr(r)
-	cons.UserAgent = r.UserAgent()
+	cons.WithRequest(r)
 
 	if err := stream.AddConsumer(cons); err != nil {
 		log.Error().Err(err).Caller().Send()

internal/rtsp/rtsp.go

@@ -1,6 +1,7 @@
 package rtsp
 
 import (
+	"errors"
 	"io"
 	"net"
 	"net/url"
@@ -147,6 +148,7 @@ func tcpHandler(conn *rtsp.Conn) {
 	var closer func()
 
 	trace := log.Trace().Enabled()
+	level := zerolog.WarnLevel
 
 	conn.Listen(func(msg any) {
 		if trace {
@@ -184,12 +186,38 @@ func tcpHandler(conn *rtsp.Conn) {
 				}
 			}
 
+			if query.Get("backchannel") == "1" {
+				conn.Medias = append(conn.Medias, &core.Media{
+					Kind:      core.KindAudio,
+					Direction: core.DirectionRecvonly,
+					Codecs: []*core.Codec{
+						{Name: core.CodecOpus, ClockRate: 48000, Channels: 2},
+						{Name: core.CodecPCM, ClockRate: 16000},
+						{Name: core.CodecPCMA, ClockRate: 16000},
+						{Name: core.CodecPCMU, ClockRate: 16000},
+						{Name: core.CodecPCM, ClockRate: 8000},
+						{Name: core.CodecPCMA, ClockRate: 8000},
+						{Name: core.CodecPCMU, ClockRate: 8000},
+					},
+				})
+			}
+
 			if s := query.Get("pkt_size"); s != "" {
 				conn.PacketSize = uint16(core.Atoi(s))
 			}
 
+			// param name like ffmpeg style https://ffmpeg.org/ffmpeg-protocols.html
+			if s := query.Get("log_level"); s != "" {
+				if lvl, err := zerolog.ParseLevel(s); err == nil {
+					level = lvl
+				}
+			}
+
+			// will help to protect looping requests to same source
+			conn.Connection.Source = query.Get("source")
+
 			if err := stream.AddConsumer(conn); err != nil {
-				log.Warn().Err(err).Str("stream", name).Msg("[rtsp]")
+				log.WithLevel(level).Err(err).Str("stream", name).Msg("[rtsp]")
 				return
 			}
 
@@ -210,6 +238,11 @@ func tcpHandler(conn *rtsp.Conn) {
 				return
 			}
 
+			query := conn.URL.Query()
+			if s := query.Get("timeout"); s != "" {
+				conn.Timeout = core.Atoi(s)
+			}
+
 			log.Debug().Str("stream", name).Msg("[rtsp] new producer")
 
 			stream.AddProducer(conn)
@@ -221,8 +254,10 @@ func tcpHandler(conn *rtsp.Conn) {
 	})
 
 	if err := conn.Accept(); err != nil {
-		if err != io.EOF {
-			log.Warn().Err(err).Caller().Send()
+		if errors.Is(err, rtsp.FailedAuth) {
+			log.Warn().Str("remote_addr", conn.Connection.RemoteAddr).Msg("[rtsp] failed authentication")
+		} else if err != io.EOF {
+			log.WithLevel(level).Err(err).Caller().Send()
 		}
 		if closer != nil {
 			closer()

internal/streams/README.md

@@ -0,0 +1,55 @@
+## Examples
+
+```yaml
+streams:
+  # known RTSP sources
+  rtsp-dahua1:   rtsp://admin:password@192.168.10.90/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif
+  rtsp-dahua2:   rtsp://admin:password@192.168.10.90/cam/realmonitor?channel=1&subtype=1
+  rtsp-tplink1:  rtsp://admin:password@192.168.10.91/stream1
+  rtsp-tplink2:  rtsp://admin:password@192.168.10.91/stream2
+  rtsp-reolink1: rtsp://admin:password@192.168.10.92/h264Preview_01_main
+  rtsp-reolink2: rtsp://admin:password@192.168.10.92/h264Preview_01_sub
+  rtsp-sonoff1:  rtsp://admin:password@192.168.10.93/av_stream/ch0
+  rtsp-sonoff2:  rtsp://admin:password@192.168.10.93/av_stream/ch1
+
+  # known RTMP sources
+  rtmp-reolink1: rtmp://192.168.10.92/bcs/channel0_main.bcs?channel=0&stream=0&user=admin&password=password
+  rtmp-reolink2: rtmp://192.168.10.92/bcs/channel0_sub.bcs?channel=0&stream=1&user=admin&password=password
+  rtmp-reolink3: rtmp://192.168.10.92/bcs/channel0_ext.bcs?channel=0&stream=1&user=admin&password=password
+
+  # known HTTP sources
+  http-reolink1: http://192.168.10.92/flv?port=1935&app=bcs&stream=channel0_main.bcs&user=admin&password=password
+  http-reolink2: http://192.168.10.92/flv?port=1935&app=bcs&stream=channel0_sub.bcs&user=admin&password=password
+  http-reolink3: http://192.168.10.92/flv?port=1935&app=bcs&stream=channel0_ext.bcs&user=admin&password=password
+
+  # known ONVIF sources
+  onvif-dahua1:   onvif://admin:password@192.168.10.90?subtype=MediaProfile00000
+  onvif-dahua2:   onvif://admin:password@192.168.10.90?subtype=MediaProfile00001
+  onvif-dahua3:   onvif://admin:password@192.168.10.90?subtype=MediaProfile00000&snapshot
+  onvif-tplink1:  onvif://admin:password@192.168.10.91:2020?subtype=profile_1
+  onvif-tplink2:  onvif://admin:password@192.168.10.91:2020?subtype=profile_2
+  onvif-reolink1: onvif://admin:password@192.168.10.92:8000?subtype=000
+  onvif-reolink2: onvif://admin:password@192.168.10.92:8000?subtype=001
+  onvif-reolink3: onvif://admin:password@192.168.10.92:8000?subtype=000&snapshot
+  onvif-openipc1: onvif://admin:password@192.168.10.95:80?subtype=PROFILE_000
+  onvif-openipc2: onvif://admin:password@192.168.10.95:80?subtype=PROFILE_001
+
+  # some EXEC examples
+  exec-h264-pipe:   exec:ffmpeg -re -i bbb.mp4 -c copy -f h264 -
+  exec-flv-pipe:    exec:ffmpeg -re -i bbb.mp4 -c copy -f flv -
+  exec-mpegts-pipe: exec:ffmpeg -re -i bbb.mp4 -c copy -f mpegts -
+  exec-adts-pipe:   exec:ffmpeg -re -i bbb.mp4 -c copy -f adts -
+  exec-mjpeg-pipe:  exec:ffmpeg -re -i bbb.mp4 -c mjpeg -f mjpeg -
+  exec-hevc-pipe:   exec:ffmpeg -re -i bbb.mp4 -c libx265 -preset superfast -tune zerolatency -f hevc -
+  exec-wav-pipe:    exec:ffmpeg -re -i bbb.mp4 -c pcm_alaw -ar 8000 -ac 1 -f wav -
+  exec-y4m-pipe:    exec:ffmpeg -re -i bbb.mp4 -c rawvideo -f yuv4mpegpipe -
+  exec-pcma-pipe:   exec:ffmpeg -re -i numb.mp3 -c:a pcm_alaw -ar:a 8000 -ac:a 1 -f wav -
+  exec-pcmu-pipe:   exec:ffmpeg -re -i numb.mp3 -c:a pcm_mulaw -ar:a 8000 -ac:a 1 -f wav -
+  exec-s16le-pipe:  exec:ffmpeg -re -i numb.mp3 -c:a pcm_s16le -ar:a 16000 -ac:a 1 -f wav -
+
+  # some FFmpeg examples
+  ffmpeg-video-h264: ffmpeg:virtual?video#video=h264
+  ffmpeg-video-4K:   ffmpeg:virtual?video&size=4K#video=h264
+  ffmpeg-video-10s:  ffmpeg:virtual?video&duration=10#video=h264
+  ffmpeg-video-src2: ffmpeg:virtual?video=testsrc2&size=2K#video=h264
+```

internal/streams/add_consumer.go

@@ -22,6 +22,12 @@ func (s *Stream) AddConsumer(cons core.Consumer) (err error) {
 
 	producers:
 		for prodN, prod := range s.producers {
+			// check for loop request, ex. `camera1: ffmpeg:camera1`
+			if info, ok := cons.(core.Info); ok && prod.url == info.GetSource() {
+				log.Trace().Msgf("[streams] skip cons=%d prod=%d", consN, prodN)
+				continue
+			}
+
 			if prodErrors[prodN] != nil {
 				log.Trace().Msgf("[streams] skip cons=%d prod=%d", consN, prodN)
 				continue
@@ -129,7 +135,7 @@ func formatError(consMedias, prodMedias []*core.Media, prodErrors []error) error
 		for _, media := range prodMedias {
 			if media.Direction == core.DirectionRecvonly {
 				for _, codec := range media.Codecs {
-					prod = appendString(prod, codec.PrintName())
+					prod = appendString(prod, media.Kind+":"+codec.PrintName())
 				}
 			}
 		}
@@ -137,7 +143,7 @@ func formatError(consMedias, prodMedias []*core.Media, prodErrors []error) error
 		for _, media := range consMedias {
 			if media.Direction == core.DirectionSendonly {
 				for _, codec := range media.Codecs {
-					cons = appendString(cons, codec.PrintName())
+					cons = appendString(cons, media.Kind+":"+codec.PrintName())
 				}
 			}
 		}

internal/streams/api.go

@@ -0,0 +1,124 @@
+package streams
+
+import (
+	"net/http"
+
+	"github.com/AlexxIT/go2rtc/internal/api"
+	"github.com/AlexxIT/go2rtc/internal/app"
+	"github.com/AlexxIT/go2rtc/pkg/probe"
+)
+
+func apiStreams(w http.ResponseWriter, r *http.Request) {
+	query := r.URL.Query()
+	src := query.Get("src")
+
+	// without source - return all streams list
+	if src == "" && r.Method != "POST" {
+		api.ResponseJSON(w, streams)
+		return
+	}
+
+	// Not sure about all this API. Should be rewrited...
+	switch r.Method {
+	case "GET":
+		stream := Get(src)
+		if stream == nil {
+			http.Error(w, "", http.StatusNotFound)
+			return
+		}
+
+		cons := probe.NewProbe(query)
+		if len(cons.Medias) != 0 {
+			cons.WithRequest(r)
+			if err := stream.AddConsumer(cons); err != nil {
+				http.Error(w, err.Error(), http.StatusInternalServerError)
+				return
+			}
+
+			api.ResponsePrettyJSON(w, stream)
+
+			stream.RemoveConsumer(cons)
+		} else {
+			api.ResponsePrettyJSON(w, streams[src])
+		}
+
+	case "PUT":
+		name := query.Get("name")
+		if name == "" {
+			name = src
+		}
+
+		if New(name, query["src"]...) == nil {
+			http.Error(w, "", http.StatusBadRequest)
+			return
+		}
+
+		if err := app.PatchConfig([]string{"streams", name}, query["src"]); err != nil {
+			http.Error(w, err.Error(), http.StatusBadRequest)
+		}
+
+	case "PATCH":
+		name := query.Get("name")
+		if name == "" {
+			http.Error(w, "", http.StatusBadRequest)
+			return
+		}
+
+		// support {input} templates: https://github.com/AlexxIT/go2rtc#module-hass
+		if Patch(name, src) == nil {
+			http.Error(w, "", http.StatusBadRequest)
+		}
+
+	case "POST":
+		// with dst - redirect source to dst
+		if dst := query.Get("dst"); dst != "" {
+			if stream := Get(dst); stream != nil {
+				if err := Validate(src); err != nil {
+					http.Error(w, err.Error(), http.StatusBadRequest)
+				} else if err = stream.Play(src); err != nil {
+					http.Error(w, err.Error(), http.StatusInternalServerError)
+				} else {
+					api.ResponseJSON(w, stream)
+				}
+			} else if stream = Get(src); stream != nil {
+				if err := Validate(dst); err != nil {
+					http.Error(w, err.Error(), http.StatusBadRequest)
+				} else if err = stream.Publish(dst); err != nil {
+					http.Error(w, err.Error(), http.StatusInternalServerError)
+				}
+			} else {
+				http.Error(w, "", http.StatusNotFound)
+			}
+		} else {
+			http.Error(w, "", http.StatusBadRequest)
+		}
+
+	case "DELETE":
+		delete(streams, src)
+
+		if err := app.PatchConfig([]string{"streams", src}, nil); err != nil {
+			http.Error(w, err.Error(), http.StatusBadRequest)
+		}
+	}
+}
+
+func apiStreamsDOT(w http.ResponseWriter, r *http.Request) {
+	query := r.URL.Query()
+
+	dot := make([]byte, 0, 1024)
+	dot = append(dot, "digraph {\n"...)
+	if query.Has("src") {
+		for _, name := range query["src"] {
+			if stream := streams[name]; stream != nil {
+				dot = AppendDOT(dot, stream)
+			}
+		}
+	} else {
+		for _, stream := range streams {
+			dot = AppendDOT(dot, stream)
+		}
+	}
+	dot = append(dot, '}')
+
+	api.Response(w, dot, "text/vnd.graphviz")
+}

internal/streams/dot.go

@@ -0,0 +1,176 @@
+package streams
+
+import (
+	"encoding/json"
+	"fmt"
+	"strings"
+)
+
+func AppendDOT(dot []byte, stream *Stream) []byte {
+	for _, prod := range stream.producers {
+		if prod.conn == nil {
+			continue
+		}
+		c, err := marshalConn(prod.conn)
+		if err != nil {
+			continue
+		}
+		dot = c.appendDOT(dot, "producer")
+	}
+	for _, cons := range stream.consumers {
+		c, err := marshalConn(cons)
+		if err != nil {
+			continue
+		}
+		dot = c.appendDOT(dot, "consumer")
+	}
+	return dot
+}
+
+func marshalConn(v any) (*conn, error) {
+	b, err := json.Marshal(v)
+	if err != nil {
+		return nil, err
+	}
+	var c conn
+	if err = json.Unmarshal(b, &c); err != nil {
+		return nil, err
+	}
+	return &c, nil
+}
+
+const bytesK = "KMGTP"
+
+func humanBytes(i int) string {
+	if i < 1000 {
+		return fmt.Sprintf("%d B", i)
+	}
+
+	f := float64(i) / 1000
+	var n uint8
+	for f >= 1000 && n < 5 {
+		f /= 1000
+		n++
+	}
+	return fmt.Sprintf("%.2f %cB", f, bytesK[n])
+}
+
+type node struct {
+	ID     uint32         `json:"id"`
+	Codec  map[string]any `json:"codec"`
+	Parent uint32         `json:"parent"`
+	Childs []uint32       `json:"childs"`
+	Bytes  int            `json:"bytes"`
+	//Packets uint32         `json:"packets"`
+	//Drops   uint32         `json:"drops"`
+}
+
+var codecKeys = []string{"codec_name", "sample_rate", "channels", "profile", "level"}
+
+func (n *node) name() string {
+	if name, ok := n.Codec["codec_name"].(string); ok {
+		return name
+	}
+	return "unknown"
+}
+
+func (n *node) codec() []byte {
+	b := make([]byte, 0, 128)
+	for _, k := range codecKeys {
+		if v := n.Codec[k]; v != nil {
+			b = fmt.Appendf(b, "%s=%v\n", k, v)
+		}
+	}
+	if l := len(b); l > 0 {
+		return b[:l-1]
+	}
+	return b
+}
+
+func (n *node) appendDOT(dot []byte, group string) []byte {
+	dot = fmt.Appendf(dot, "%d [group=%s, label=%q, title=%q];\n", n.ID, group, n.name(), n.codec())
+	//for _, sink := range n.Childs {
+	//	dot = fmt.Appendf(dot, "%d -> %d;\n", n.ID, sink)
+	//}
+	return dot
+}
+
+type conn struct {
+	ID         uint32 `json:"id"`
+	FormatName string `json:"format_name"`
+	Protocol   string `json:"protocol"`
+	RemoteAddr string `json:"remote_addr"`
+	Source     string `json:"source"`
+	URL        string `json:"url"`
+	UserAgent  string `json:"user_agent"`
+	Receivers  []node `json:"receivers"`
+	Senders    []node `json:"senders"`
+	BytesRecv  int    `json:"bytes_recv"`
+	BytesSend  int    `json:"bytes_send"`
+}
+
+func (c *conn) appendDOT(dot []byte, group string) []byte {
+	host := c.host()
+	dot = fmt.Appendf(dot, "%s [group=host];\n", host)
+	dot = fmt.Appendf(dot, "%d [group=%s, label=%q, title=%q];\n", c.ID, group, c.FormatName, c.label())
+	if group == "producer" {
+		dot = fmt.Appendf(dot, "%s -> %d [label=%q];\n", host, c.ID, humanBytes(c.BytesRecv))
+	} else {
+		dot = fmt.Appendf(dot, "%d -> %s [label=%q];\n", c.ID, host, humanBytes(c.BytesSend))
+	}
+
+	for _, recv := range c.Receivers {
+		dot = fmt.Appendf(dot, "%d -> %d [label=%q];\n", c.ID, recv.ID, humanBytes(recv.Bytes))
+		dot = recv.appendDOT(dot, "node")
+	}
+	for _, send := range c.Senders {
+		dot = fmt.Appendf(dot, "%d -> %d [label=%q];\n", send.Parent, c.ID, humanBytes(send.Bytes))
+		//dot = fmt.Appendf(dot, "%d -> %d [label=%q];\n", send.ID, c.ID, humanBytes(send.Bytes))
+		//dot = send.appendDOT(dot, "node")
+	}
+	return dot
+}
+
+func (c *conn) host() (s string) {
+	if c.Protocol == "pipe" {
+		return "127.0.0.1"
+	}
+
+	if s = c.RemoteAddr; s == "" {
+		return "unknown"
+	}
+
+	if i := strings.Index(s, "forwarded"); i > 0 {
+		s = s[i+10:]
+	}
+
+	if s[0] == '[' {
+		if i := strings.Index(s, "]"); i > 0 {
+			return s[1:i]
+		}
+	}
+
+	if i := strings.IndexAny(s, " ,:"); i > 0 {
+		return s[:i]
+	}
+	return
+}
+
+func (c *conn) label() string {
+	var sb strings.Builder
+	sb.WriteString("format_name=" + c.FormatName)
+	if c.Protocol != "" {
+		sb.WriteString("\nprotocol=" + c.Protocol)
+	}
+	if c.Source != "" {
+		sb.WriteString("\nsource=" + c.Source)
+	}
+	if c.URL != "" {
+		sb.WriteString("\nurl=" + c.URL)
+	}
+	if c.UserAgent != "" {
+		sb.WriteString("\nuser_agent=" + c.UserAgent)
+	}
+	// escape quotes https://github.com/AlexxIT/go2rtc/issues/1603
+	return strings.ReplaceAll(sb.String(), `"`, `'`)
+}

internal/streams/handlers.go

@@ -7,7 +7,7 @@ import (
 	"github.com/AlexxIT/go2rtc/pkg/core"
 )
 
-type Handler func(url string) (core.Producer, error)
+type Handler func(source string) (core.Producer, error)
 
 var handlers = map[string]Handler{}
 

internal/streams/play.go

@@ -2,10 +2,12 @@ package streams
 
 import (
 	"errors"
+	"time"
+
 	"github.com/AlexxIT/go2rtc/pkg/core"
 )
 
-func (s *Stream) Play(source string) error {
+func (s *Stream) Play(urlOrProd any) error {
 	s.mu.Lock()
 	for _, producer := range s.producers {
 		if producer.state == stateInternal && producer.conn != nil {
@@ -14,12 +16,18 @@ func (s *Stream) Play(source string) error {
 	}
 	s.mu.Unlock()
 
-	if source == "" {
-		return nil
-	}
-
+	var source string
 	var src core.Producer
 
+	switch urlOrProd.(type) {
+	case string:
+		if source = urlOrProd.(string); source == "" {
+			return nil
+		}
+	case core.Producer:
+		src = urlOrProd.(core.Producer)
+	}
+
 	for _, producer := range s.producers {
 		if producer.conn == nil {
 			continue
@@ -80,18 +88,20 @@ func (s *Stream) Play(source string) error {
 		s.AddInternalProducer(src)
 		s.AddInternalConsumer(cons)
 
-		go func() {
-			_ = src.Start()
-			_ = dst.Stop()
-			s.RemoveProducer(src)
-		}()
-
 		go func() {
 			_ = dst.Start()
 			_ = src.Stop()
 			s.RemoveInternalConsumer(cons)
 		}()
 
+		go func() {
+			_ = src.Start()
+			// little timeout before stop dst, so the buffer can be transferred
+			time.Sleep(time.Second)
+			_ = dst.Stop()
+			s.RemoveProducer(src)
+		}()
+
 		return nil
 	}
 
@@ -99,7 +109,7 @@ func (s *Stream) Play(source string) error {
 }
 
 func (s *Stream) AddInternalProducer(conn core.Producer) {
-	producer := &Producer{conn: conn, state: stateInternal}
+	producer := &Producer{conn: conn, state: stateInternal, url: "internal"}
 	s.mu.Lock()
 	s.producers = append(s.producers, producer)
 	s.mu.Unlock()
@@ -136,10 +146,12 @@ func matchMedia(prod core.Producer, cons core.Consumer) bool {
 
 			track, err := prod.GetTrack(prodMedia, prodCodec)
 			if err != nil {
+				log.Warn().Err(err).Msg("[streams] can't get track")
 				continue
 			}
 
 			if err = cons.AddTrack(consMedia, consCodec, track); err != nil {
+				log.Warn().Err(err).Msg("[streams] can't add track")
 				continue
 			}
 

internal/streams/producer.go

@@ -132,11 +132,10 @@ func (p *Producer) AddTrack(media *core.Media, codec *core.Codec, track *core.Re
 }
 
 func (p *Producer) MarshalJSON() ([]byte, error) {
-	if p.conn != nil {
-		return json.Marshal(p.conn)
+	if conn := p.conn; conn != nil {
+		return json.Marshal(conn)
 	}
-
-	info := core.Info{URL: p.url}
+	info := map[string]string{"url": p.url}
 	return json.Marshal(info)
 }
 
@@ -207,7 +206,7 @@ func (p *Producer) reconnect(workerID, retry int) {
 	for _, media := range conn.GetMedias() {
 		switch media.Direction {
 		case core.DirectionRecvonly:
-			for _, receiver := range p.receivers {
+			for i, receiver := range p.receivers {
 				codec := media.MatchCodec(receiver.Codec)
 				if codec == nil {
 					continue
@@ -219,6 +218,7 @@ func (p *Producer) reconnect(workerID, retry int) {
 				}
 
 				receiver.Replace(track)
+				p.receivers[i] = track
 				break
 			}
 
@@ -234,6 +234,9 @@ func (p *Producer) reconnect(workerID, retry int) {
 		}
 	}
 
+	// stop previous connection after moving tracks (fix ghost exec/ffmpeg)
+	_ = p.conn.Stop()
+	// swap connections
 	p.conn = conn
 
 	go p.worker(conn, workerID)

internal/streams/stream.go

@@ -21,6 +21,12 @@ func NewStream(source any) *Stream {
 		return &Stream{
 			producers: []*Producer{NewProducer(source)},
 		}
+	case []string:
+		s := new(Stream)
+		for _, str := range source {
+			s.producers = append(s.producers, NewProducer(str))
+		}
+		return s
 	case []any:
 		s := new(Stream)
 		for _, src := range source {
@@ -41,11 +47,12 @@ func NewStream(source any) *Stream {
 	}
 }
 
-func (s *Stream) Sources() (sources []string) {
+func (s *Stream) Sources() []string {
+	sources := make([]string, 0, len(s.producers))
 	for _, prod := range s.producers {
 		sources = append(sources, prod.url)
 	}
-	return
+	return sources
 }
 
 func (s *Stream) SetSource(source string) {
@@ -70,7 +77,7 @@ func (s *Stream) RemoveConsumer(cons core.Consumer) {
 }
 
 func (s *Stream) AddProducer(prod core.Producer) {
-	producer := &Producer{conn: prod, state: stateExternal}
+	producer := &Producer{conn: prod, state: stateExternal, url: "external"}
 	s.mu.Lock()
 	s.producers = append(s.producers, producer)
 	s.mu.Unlock()
@@ -112,19 +119,12 @@ producers:
 }
 
 func (s *Stream) MarshalJSON() ([]byte, error) {
-	if !s.mu.TryLock() {
-		log.Warn().Msgf("[streams] json locked")
-		return json.Marshal(nil)
-	}
-
-	var info struct {
+	var info = struct {
 		Producers []*Producer     `json:"producers"`
 		Consumers []core.Consumer `json:"consumers"`
+	}{
+		Producers: s.producers,
+		Consumers: s.consumers,
 	}
-	info.Producers = s.producers
-	info.Consumers = s.consumers
-
-	s.mu.Unlock()
-
 	return json.Marshal(info)
 }

internal/streams/streams.go

@@ -1,7 +1,7 @@
 package streams
 
 import (
-	"net/http"
+	"errors"
 	"net/url"
 	"regexp"
 	"sync"
@@ -9,8 +9,6 @@ import (
 
 	"github.com/AlexxIT/go2rtc/internal/api"
 	"github.com/AlexxIT/go2rtc/internal/app"
-	"github.com/AlexxIT/go2rtc/pkg/probe"
-	"github.com/AlexxIT/go2rtc/pkg/tcp"
 	"github.com/rs/zerolog"
 )
 
@@ -28,7 +26,8 @@ func Init() {
 		streams[name] = NewStream(item)
 	}
 
-	api.HandleFunc("api/streams", streamsHandler)
+	api.HandleFunc("api/streams", apiStreams)
+	api.HandleFunc("api/streams.dot", apiStreamsDOT)
 
 	if cfg.Publish == nil {
 		return
@@ -43,20 +42,29 @@ func Init() {
 	})
 }
 
-func Get(name string) *Stream {
-	return streams[name]
-}
-
 var sanitize = regexp.MustCompile(`\s`)
 
-func New(name string, source string) *Stream {
-	// not allow creating dynamic streams with spaces in the source
+// Validate - not allow creating dynamic streams with spaces in the source
+func Validate(source string) error {
 	if sanitize.MatchString(source) {
-		return nil
+		return errors.New("streams: invalid dynamic source")
+	}
+	return nil
+}
+
+func New(name string, sources ...string) *Stream {
+	for _, source := range sources {
+		if Validate(source) != nil {
+			return nil
+		}
 	}
 
-	stream := NewStream(source)
+	stream := NewStream(sources)
+
+	streamsMu.Lock()
 	streams[name] = stream
+	streamsMu.Unlock()
+
 	return stream
 }
 
@@ -89,6 +97,10 @@ func Patch(name string, source string) *Stream {
 		return nil
 	}
 
+	if Validate(source) != nil {
+		return nil
+	}
+
 	// check an existing stream with this name
 	if stream, ok := streams[name]; ok {
 		stream.SetSource(source)
@@ -96,7 +108,9 @@ func Patch(name string, source string) *Stream {
 	}
 
 	// create new stream with this name
-	return New(name, source)
+	stream := NewStream(source)
+	streams[name] = stream
+	return stream
 }
 
 func GetOrPatch(query url.Values) *Stream {
@@ -107,7 +121,7 @@ func GetOrPatch(query url.Values) *Stream {
 	}
 
 	// check if src is stream name
-	if stream, ok := streams[source]; ok {
+	if stream := Get(source); stream != nil {
 		return stream
 	}
 
@@ -122,112 +136,41 @@ func GetOrPatch(query url.Values) *Stream {
 	return Patch(source, source)
 }
 
-func GetAll() (names []string) {
+var log zerolog.Logger
+
+// streams map
+
+var streams = map[string]*Stream{}
+var streamsMu sync.Mutex
+
+func Get(name string) *Stream {
+	streamsMu.Lock()
+	defer streamsMu.Unlock()
+	return streams[name]
+}
+
+func Delete(name string) {
+	streamsMu.Lock()
+	defer streamsMu.Unlock()
+	delete(streams, name)
+}
+
+func GetAllNames() []string {
+	streamsMu.Lock()
+	names := make([]string, 0, len(streams))
 	for name := range streams {
 		names = append(names, name)
 	}
-	return
+	streamsMu.Unlock()
+	return names
 }
 
-func Streams() map[string]*Stream {
-	return streams
-}
-
-func Delete(id string) {
-	delete(streams, id)
-}
-
-func streamsHandler(w http.ResponseWriter, r *http.Request) {
-	query := r.URL.Query()
-	src := query.Get("src")
-
-	// without source - return all streams list
-	if src == "" && r.Method != "POST" {
-		api.ResponseJSON(w, streams)
-		return
-	}
-
-	// Not sure about all this API. Should be rewrited...
-	switch r.Method {
-	case "GET":
-		stream := Get(src)
-		if stream == nil {
-			http.Error(w, "", http.StatusNotFound)
-			return
-		}
-
-		cons := probe.NewProbe(query)
-		if len(cons.Medias) != 0 {
-			cons.RemoteAddr = tcp.RemoteAddr(r)
-			cons.UserAgent = r.UserAgent()
-			if err := stream.AddConsumer(cons); err != nil {
-				http.Error(w, err.Error(), http.StatusInternalServerError)
-				return
-			}
-
-			api.ResponsePrettyJSON(w, stream)
-
-			stream.RemoveConsumer(cons)
-		} else {
-			api.ResponsePrettyJSON(w, streams[src])
-		}
-
-	case "PUT":
-		name := query.Get("name")
-		if name == "" {
-			name = src
-		}
-
-		if New(name, src) == nil {
-			http.Error(w, "", http.StatusBadRequest)
-			return
-		}
-
-		if err := app.PatchConfig(name, src, "streams"); err != nil {
-			http.Error(w, err.Error(), http.StatusBadRequest)
-		}
-
-	case "PATCH":
-		name := query.Get("name")
-		if name == "" {
-			http.Error(w, "", http.StatusBadRequest)
-			return
-		}
-
-		// support {input} templates: https://github.com/AlexxIT/go2rtc#module-hass
-		if Patch(name, src) == nil {
-			http.Error(w, "", http.StatusBadRequest)
-		}
-
-	case "POST":
-		// with dst - redirect source to dst
-		if dst := query.Get("dst"); dst != "" {
-			if stream := Get(dst); stream != nil {
-				if err := stream.Play(src); err != nil {
-					http.Error(w, err.Error(), http.StatusInternalServerError)
-				} else {
-					api.ResponseJSON(w, stream)
-				}
-			} else if stream = Get(src); stream != nil {
-				if err := stream.Publish(dst); err != nil {
-					http.Error(w, err.Error(), http.StatusInternalServerError)
-				}
-			} else {
-				http.Error(w, "", http.StatusNotFound)
-			}
-		} else {
-			http.Error(w, "", http.StatusBadRequest)
-		}
-
-	case "DELETE":
-		delete(streams, src)
-
-		if err := app.PatchConfig(src, nil, "streams"); err != nil {
-			http.Error(w, err.Error(), http.StatusBadRequest)
-		}
+func GetAllSources() map[string][]string {
+	streamsMu.Lock()
+	sources := make(map[string][]string, len(streams))
+	for name, stream := range streams {
+		sources[name] = stream.Sources()
 	}
+	streamsMu.Unlock()
+	return sources
 }
-
-var log zerolog.Logger
-var streams = map[string]*Stream{}
-var streamsMu sync.Mutex

internal/tapo/tapo.go

@@ -8,11 +8,15 @@ import (
 )
 
 func Init() {
-	streams.HandleFunc("kasa", func(url string) (core.Producer, error) {
-		return kasa.Dial(url)
+	streams.HandleFunc("kasa", func(source string) (core.Producer, error) {
+		return kasa.Dial(source)
 	})
 
-	streams.HandleFunc("tapo", func(url string) (core.Producer, error) {
-		return tapo.Dial(url)
+	streams.HandleFunc("tapo", func(source string) (core.Producer, error) {
+		return tapo.Dial(source)
+	})
+
+	streams.HandleFunc("vigi", func(source string) (core.Producer, error) {
+		return tapo.Dial(source)
 	})
 }

internal/v4l2/README.md

@@ -0,0 +1,39 @@
+# V4L2
+
+What you should to know about [V4L2](https://en.wikipedia.org/wiki/Video4Linux):
+
+- V4L2 (Video for Linux API version 2) works only in Linux
+- supports USB cameras and other similar devices
+- one device can only be connected to one software simultaneously
+- cameras support a fixed list of formats, resolutions and frame rates
+- basic cameras supports only RAW (non-compressed) pixel formats
+- regular cameras supports MJPEG format (series of JPEG frames)
+- advances cameras support H264 format (MSE/MP4, WebRTC compatible)
+- using MJPEG and H264 formats (if the camera supports them) won't cost you the CPU usage
+- transcoding RAW format to MJPEG or H264 - will cost you a significant CPU usage
+- H265 (HEVC) format is also supported (if the camera supports it)
+
+Tests show that the basic Keenetic router with MIPS processor can broadcast three MJPEG cameras in the following resolutions: 1600х1200 + 640х480 + 640х480. The USB bus bandwidth is no more enough for larger resolutions. CPU consumption is no more than 5%.
+
+Supported formats for your camera can be found here: **Go2rtc > WebUI > Add > V4L2**.
+
+## RAW format
+
+Example:
+
+```yaml
+streams:
+  camera1: v4l2:device?video=/dev/video0&input_format=yuyv422&video_size=1280x720&framerate=10
+```
+
+Go2rtc supports built-in transcoding of RAW to MJPEG format. This does not need to be additionally configured.
+
+```
+ffplay http://localhost:1984/api/stream.mjpeg?src=camera1
+```
+
+**Important.** You don't have to transcode the RAW format to transmit it over the network. You can stream it in `y4m` format, which is perfectly supported by ffmpeg. It won't cost you a CPU usage. But will require high network bandwidth.
+
+```
+ffplay http://localhost:1984/api/stream.y4m?src=camera1
+```

internal/v4l2/v4l2.go

@@ -0,0 +1,7 @@
+//go:build !(linux && (386 || arm || mipsle || amd64 || arm64))
+
+package v4l2
+
+func Init() {
+	// not supported
+}

internal/v4l2/v4l2_linux.go

@@ -0,0 +1,91 @@
+//go:build linux && (386 || arm || mipsle || amd64 || arm64)
+
+package v4l2
+
+import (
+	"encoding/binary"
+	"fmt"
+	"net/http"
+	"os"
+	"strings"
+
+	"github.com/AlexxIT/go2rtc/internal/api"
+	"github.com/AlexxIT/go2rtc/internal/streams"
+	"github.com/AlexxIT/go2rtc/pkg/core"
+	"github.com/AlexxIT/go2rtc/pkg/v4l2"
+	"github.com/AlexxIT/go2rtc/pkg/v4l2/device"
+)
+
+func Init() {
+	streams.HandleFunc("v4l2", func(source string) (core.Producer, error) {
+		return v4l2.Open(source)
+	})
+
+	api.HandleFunc("api/v4l2", apiV4L2)
+}
+
+func apiV4L2(w http.ResponseWriter, r *http.Request) {
+	files, err := os.ReadDir("/dev")
+	if err != nil {
+		return
+	}
+
+	var sources []*api.Source
+
+	for _, file := range files {
+		if !strings.HasPrefix(file.Name(), core.KindVideo) {
+			continue
+		}
+
+		path := "/dev/" + file.Name()
+
+		dev, err := device.Open(path)
+		if err != nil {
+			continue
+		}
+
+		formats, _ := dev.ListFormats()
+		for _, fourCC := range formats {
+			name, ffmpeg := findFormat(fourCC)
+			source := &api.Source{Name: name}
+
+			sizes, _ := dev.ListSizes(fourCC)
+			for _, wh := range sizes {
+				if source.Info != "" {
+					source.Info += " "
+				}
+
+				source.Info += fmt.Sprintf("%dx%d", wh[0], wh[1])
+
+				frameRates, _ := dev.ListFrameRates(fourCC, wh[0], wh[1])
+				for _, fr := range frameRates {
+					source.Info += fmt.Sprintf("@%d", fr)
+
+					if source.URL == "" && ffmpeg != "" {
+						source.URL = fmt.Sprintf(
+							"v4l2:device?video=%s&input_format=%s&video_size=%dx%d&framerate=%d",
+							path, ffmpeg, wh[0], wh[1], fr,
+						)
+					}
+				}
+			}
+
+			if source.Info != "" {
+				sources = append(sources, source)
+			}
+		}
+
+		_ = dev.Close()
+	}
+
+	api.ResponseSources(w, sources)
+}
+
+func findFormat(fourCC uint32) (name, ffmpeg string) {
+	for _, format := range device.Formats {
+		if format.FourCC == fourCC {
+			return format.Name, format.FFmpeg
+		}
+	}
+	return string(binary.LittleEndian.AppendUint32(nil, fourCC)), ""
+}

internal/webrtc/README.md

@@ -11,13 +11,15 @@ If an external connection via STUN is used:
 
 - Uses [UDP hole punching](https://en.wikipedia.org/wiki/UDP_hole_punching) technology to bypass NAT even if you not open your server to the World
 - For about 20% of users, the techology will not work because of the [Symmetric NAT](https://tomchen.github.io/symmetric-nat-test/)
-- UDP is not suitable for transmitting 2K and 4K high bitrate video over open networks because of the high loss rate
+- UDP is not suitable for transmitting 2K and 4K high bitrate video over open networks because of the high loss rate:
+  - https://habr.com/ru/companies/flashphoner/articles/480006/
+  - https://www.youtube.com/watch?v=FXVg2ckuKfs
 
 ## Default config
 
 ```yaml
 webrtc:
-  listen: ":8555/tcp"
+  listen: ":8555"
   ice_servers:
     - urls: [ "stun:stun.l.google.com:19302" ]
 ```
@@ -29,7 +31,7 @@ webrtc:
 ```yaml
 webrtc:
   # fix local TCP or UDP or both ports for WebRTC media
-  listen: ":8555/tcp"        # address of your local server
+  listen: ":8555"            # address of your local server
 
   # add additional host candidates manually
   # order is important, the first will have a higher priority
@@ -53,17 +55,20 @@ webrtc:
     # including candidates from the `listen` option
     # use `candidates: []` to remove all auto discovery candidates
     candidates: [ 192.168.1.123 ]
+    
+    # enable localhost candidates
+    loopback: true
 
     # list of network types to be used for connection
     # including candidates from the `listen` option
     networks: [ udp4, udp6, tcp4, tcp6 ]
 
     # list of interfaces to be used for connection
-    # not related to the `listen` option
+    # including interfaces from unspecified `listen` option (empty host)
     interfaces: [ eno1 ]
 
     # list of host IP-addresses to be used for connection
-    # not related to the `listen` option
+    # including IPs from unspecified `listen` option (empty host)
     ips: [ 192.168.1.123 ]
 
     # range for random UDP ports [min, max] to be used for connection
@@ -71,14 +76,16 @@ webrtc:
     udp_ports: [ 50000, 50100 ]
 ```
 
-By default go2rtc uses **fixed TCP** port and multiple **random UDP** ports for each WebRTC connection - `listen: ":8555/tcp"`.
+By default go2rtc uses **fixed TCP** port and **fixed UDP** ports for each **direct** WebRTC connection - `listen: ":8555"`.
 
-You can set **fixed TCP** and **fixed UDP** port for all connections - `listen: ":8555"`. This may has lower performance, but it's your choice. 
+You can set **fixed TCP** and **random UDP** port for all connections - `listen: ":8555/tcp"`.
 
 Don't know why, but you can disable TCP port and leave only random UDP ports - `listen: ""`.
 
 ## Config filters
 
+**Importan!** By default go2rtc exclude all Docker-like candidates (`172.16.0.0/12`). This can not be disabled.
+
 Filters allow you to exclude unnecessary candidates. Extra candidates don't make your connection worse or better. But the wrong filter settings can break everything. Skip this setting if you don't understand it.
 
 For example, go2rtc is installed on the host system. And there are unnecessary interfaces. You can keep only the relevant via `interfaces` or `ips` options. You can also exclude IPv6 candidates if your server supports them but your home network does not.
@@ -97,8 +104,6 @@ For example, go2rtc inside closed docker container (ex. [Frigate](https://frigat
 webrtc:
   listen: ":8555"                   # use fixed TCP and UDP ports
   candidates: [ 192.168.1.2:8555 ]  # add manual host candidate (use docker port forwarding)
-  filters:
-    candidates: []                  # skip all internal docker candidates
 ```
 
 ## Userful links

internal/webrtc/candidates.go

@@ -2,12 +2,13 @@ package webrtc
 
 import (
 	"net"
-	"slices"
 	"strings"
 
 	"github.com/AlexxIT/go2rtc/internal/api/ws"
+	"github.com/AlexxIT/go2rtc/pkg/core"
 	"github.com/AlexxIT/go2rtc/pkg/webrtc"
-	pion "github.com/pion/webrtc/v3"
+	"github.com/AlexxIT/go2rtc/pkg/xnet"
+	pion "github.com/pion/webrtc/v4"
 )
 
 type Address struct {
@@ -73,16 +74,21 @@ func FilterCandidate(candidate *pion.ICECandidate) bool {
 		return false
 	}
 
+	// remove any Docker-like IP from candidates
+	if ip := net.ParseIP(candidate.Address); ip != nil && xnet.Docker.Contains(ip) {
+		return false
+	}
+
 	// host candidate should be in the hosts list
 	if candidate.Typ == pion.ICECandidateTypeHost && filters.Candidates != nil {
-		if !slices.Contains(filters.Candidates, candidate.Address) {
+		if !core.Contains(filters.Candidates, candidate.Address) {
 			return false
 		}
 	}
 
 	if filters.Networks != nil {
 		networkType := NetworkType(candidate.Protocol.String(), candidate.Address)
-		if !slices.Contains(filters.Networks, networkType) {
+		if !core.Contains(filters.Networks, networkType) {
 			return false
 		}
 	}

internal/webrtc/client.go

@@ -15,7 +15,7 @@ import (
 	"github.com/AlexxIT/go2rtc/pkg/core"
 	"github.com/AlexxIT/go2rtc/pkg/webrtc"
 	"github.com/gorilla/websocket"
-	pion "github.com/pion/webrtc/v3"
+	pion "github.com/pion/webrtc/v4"
 )
 
 // streamsHandler supports:
@@ -41,9 +41,11 @@ func streamsHandler(rawURL string) (core.Producer, error) {
 				// https://aws.amazon.com/kinesis/video-streams/
 				// https://docs.aws.amazon.com/kinesisvideostreams-webrtc-dg/latest/devguide/what-is-kvswebrtc.html
 				// https://github.com/orgs/awslabs/repositories?q=kinesis+webrtc
-				return kinesisClient(rawURL, query, "WebRTC/Kinesis")
+				return kinesisClient(rawURL, query, "webrtc/kinesis", nil)
 			} else if format == "openipc" {
 				return openIPCClient(rawURL, query)
+			} else if format == "switchbot" {
+				return switchbotClient(rawURL, query)
 			} else {
 				return go2rtcClient(rawURL)
 			}
@@ -54,6 +56,8 @@ func streamsHandler(rawURL string) (core.Producer, error) {
 			} else if format == "wyze" {
 				// https://github.com/mrlt8/docker-wyze-bridge
 				return wyzeClient(rawURL)
+			} else if format == "creality" {
+				return crealityClient(rawURL)
 			} else {
 				return whepClient(rawURL)
 			}
@@ -77,17 +81,23 @@ func go2rtcClient(url string) (core.Producer, error) {
 	// 2. Create PeerConnection
 	pc, err := PeerConnection(true)
 	if err != nil {
-		log.Error().Err(err).Caller().Send()
 		return nil, err
 	}
 
+	defer func() {
+		if err != nil {
+			_ = pc.Close()
+		}
+	}()
+
 	// waiter will wait PC error or WS error or nil (connection OK)
 	var connState core.Waiter
 	var connMu sync.Mutex
 
 	prod := webrtc.NewConn(pc)
-	prod.Desc = "WebRTC/WebSocket async"
 	prod.Mode = core.ModeActiveProducer
+	prod.Protocol = "ws"
+	prod.URL = url
 	prod.Listen(func(msg any) {
 		switch msg := msg.(type) {
 		case *pion.ICECandidate:
@@ -132,7 +142,8 @@ func go2rtcClient(url string) (core.Producer, error) {
 	}
 
 	if msg.Type != "webrtc/answer" {
-		return nil, errors.New("wrong answer: " + msg.Type)
+		err = errors.New("wrong answer: " + msg.String())
+		return nil, err
 	}
 
 	answer := msg.String()
@@ -180,8 +191,9 @@ func whepClient(url string) (core.Producer, error) {
 	}
 
 	prod := webrtc.NewConn(pc)
-	prod.Desc = "WebRTC/WHEP sync"
 	prod.Mode = core.ModeActiveProducer
+	prod.Protocol = "http"
+	prod.URL = url
 
 	medias := []*core.Media{
 		{Kind: core.KindVideo, Direction: core.DirectionRecvonly},

internal/webrtc/client_creality.go

@@ -0,0 +1,152 @@
+package webrtc
+
+import (
+	"encoding/base64"
+	"encoding/json"
+	"io"
+	"net/http"
+	"strings"
+	"time"
+
+	"github.com/AlexxIT/go2rtc/pkg/core"
+	"github.com/AlexxIT/go2rtc/pkg/webrtc"
+	"github.com/pion/sdp/v3"
+)
+
+// https://github.com/AlexxIT/go2rtc/issues/1600
+func crealityClient(url string) (core.Producer, error) {
+	pc, err := PeerConnection(true)
+	if err != nil {
+		return nil, err
+	}
+
+	prod := webrtc.NewConn(pc)
+	prod.FormatName = "webrtc/creality"
+	prod.Mode = core.ModeActiveProducer
+	prod.Protocol = "http"
+	prod.URL = url
+
+	medias := []*core.Media{
+		{Kind: core.KindVideo, Direction: core.DirectionRecvonly},
+	}
+
+	// TODO: return webrtc.SessionDescription
+	offer, err := prod.CreateCompleteOffer(medias)
+	if err != nil {
+		return nil, err
+	}
+
+	log.Trace().Msgf("[webrtc] offer:\n%s", offer)
+
+	body, err := offerToB64(offer)
+	if err != nil {
+		return nil, err
+	}
+
+	req, err := http.NewRequest("POST", url, body)
+	if err != nil {
+		return nil, err
+	}
+	req.Header.Set("Content-Type", "plain/text")
+
+	// TODO: change http.DefaultClient settings
+	client := http.Client{Timeout: time.Second * 5000}
+	defer client.CloseIdleConnections()
+
+	res, err := client.Do(req)
+	if err != nil {
+		return nil, err
+	}
+
+	answer, err := answerFromB64(res.Body)
+	if err != nil {
+		return nil, err
+	}
+
+	log.Trace().Msgf("[webrtc] answer:\n%s", answer)
+
+	if answer, err = fixCrealitySDP(answer); err != nil {
+		return nil, err
+	}
+
+	if err = prod.SetAnswer(answer); err != nil {
+		return nil, err
+	}
+
+	return prod, nil
+}
+
+func offerToB64(sdp string) (io.Reader, error) {
+	// JS object
+	v := map[string]string{
+		"type": "offer",
+		"sdp":  sdp,
+	}
+
+	// bytes
+	b, err := json.Marshal(v)
+	if err != nil {
+		return nil, err
+	}
+
+	// base64, why? who knows...
+	s := base64.StdEncoding.EncodeToString(b)
+
+	return strings.NewReader(s), nil
+}
+
+func answerFromB64(r io.Reader) (string, error) {
+	// base64
+	b, err := io.ReadAll(r)
+	if err != nil {
+		return "", err
+	}
+
+	// bytes
+	if b, err = base64.StdEncoding.DecodeString(string(b)); err != nil {
+		return "", err
+	}
+
+	// JS object
+	var v map[string]string
+	if err = json.Unmarshal(b, &v); err != nil {
+		return "", err
+	}
+
+	// string "v=0..."
+	return v["sdp"], nil
+}
+
+func fixCrealitySDP(value string) (string, error) {
+	var sd sdp.SessionDescription
+	if err := sd.UnmarshalString(value); err != nil {
+		return "", err
+	}
+
+	md := sd.MediaDescriptions[0]
+
+	// important to skip first codec, because second codec will be used
+	skip := md.MediaName.Formats[0]
+	md.MediaName.Formats = md.MediaName.Formats[1:]
+
+	attrs := make([]sdp.Attribute, 0, len(md.Attributes))
+	for _, attr := range md.Attributes {
+		switch attr.Key {
+		case "fmtp", "rtpmap":
+			// important to skip fmtp with x-google, because this is second fmtp for same codec
+			// and pion library will fail parsing this SDP
+			if strings.HasPrefix(attr.Value, skip) || strings.Contains(attr.Value, "x-google") {
+				continue
+			}
+		}
+		attrs = append(attrs, attr)
+	}
+
+	md.Attributes = attrs
+
+	b, err := sd.Marshal()
+	if err != nil {
+		return "", err
+	}
+	return string(b), nil
+}

internal/webrtc/kinesis.go

@@ -12,7 +12,7 @@ import (
 	"github.com/AlexxIT/go2rtc/pkg/core"
 	"github.com/AlexxIT/go2rtc/pkg/webrtc"
 	"github.com/gorilla/websocket"
-	pion "github.com/pion/webrtc/v3"
+	pion "github.com/pion/webrtc/v4"
 )
 
 type kinesisRequest struct {
@@ -34,7 +34,10 @@ func (k kinesisResponse) String() string {
 	return fmt.Sprintf("type=%s, payload=%s", k.Type, k.Payload)
 }
 
-func kinesisClient(rawURL string, query url.Values, desc string) (core.Producer, error) {
+func kinesisClient(
+	rawURL string, query url.Values, format string,
+	sdpOffer func(prod *webrtc.Conn, query url.Values) (any, error),
+) (core.Producer, error) {
 	// 1. Connect to signalign server
 	conn, _, err := websocket.DefaultDialer.Dial(rawURL, nil)
 	if err != nil {
@@ -79,8 +82,10 @@ func kinesisClient(rawURL string, query url.Values, desc string) (core.Producer,
 	}
 
 	prod := webrtc.NewConn(pc)
-	prod.Desc = desc
+	prod.FormatName = format
 	prod.Mode = core.ModeActiveProducer
+	prod.Protocol = "ws"
+	prod.URL = rawURL
 	prod.Listen(func(msg any) {
 		switch msg := msg.(type) {
 		case *pion.ICECandidate:
@@ -106,23 +111,33 @@ func kinesisClient(rawURL string, query url.Values, desc string) (core.Producer,
 		}
 	})
 
-	medias := []*core.Media{
-		{Kind: core.KindVideo, Direction: core.DirectionRecvonly},
-		{Kind: core.KindAudio, Direction: core.DirectionRecvonly},
+	var payload any
+
+	if sdpOffer == nil {
+		medias := []*core.Media{
+			{Kind: core.KindVideo, Direction: core.DirectionRecvonly},
+			{Kind: core.KindAudio, Direction: core.DirectionRecvonly},
+		}
+
+		// 4. Create offer
+		var offer string
+		if offer, err = prod.CreateOffer(medias); err != nil {
+			return nil, err
+		}
+
+		// 5. Send offer
+		payload = pion.SessionDescription{
+			Type: pion.SDPTypeOffer,
+			SDP:  offer,
+		}
+	} else {
+		if payload, err = sdpOffer(prod, query); err != nil {
+			return nil, err
+		}
 	}
 
-	// 4. Create offer
-	offer, err := prod.CreateOffer(medias)
-	if err != nil {
-		return nil, err
-	}
-
-	// 5. Send offer
 	req.Action = "SDP_OFFER"
-	req.Payload, _ = json.Marshal(pion.SessionDescription{
-		Type: pion.SDPTypeOffer,
-		SDP:  offer,
-	})
+	req.Payload, _ = json.Marshal(payload)
 	if err = conn.WriteJSON(req); err != nil {
 		return nil, err
 	}
@@ -216,5 +231,5 @@ func wyzeClient(rawURL string) (core.Producer, error) {
 		"ice_servers": []string{string(kvs.Servers)},
 	}
 
-	return kinesisClient(kvs.URL, query, "WebRTC/Wyze")
+	return kinesisClient(kvs.URL, query, "webrtc/wyze", nil)
 }

internal/webrtc/milestone.go

@@ -12,7 +12,7 @@ import (
 	"github.com/AlexxIT/go2rtc/pkg/core"
 	"github.com/AlexxIT/go2rtc/pkg/tcp"
 	"github.com/AlexxIT/go2rtc/pkg/webrtc"
-	pion "github.com/pion/webrtc/v3"
+	pion "github.com/pion/webrtc/v4"
 )
 
 // This package handles the Milestone WebRTC session lifecycle, including authentication,
@@ -193,8 +193,10 @@ func milestoneClient(rawURL string, query url.Values) (core.Producer, error) {
 	}
 
 	prod := webrtc.NewConn(pc)
-	prod.Desc = "WebRTC/Milestone"
+	prod.FormatName = "webrtc/milestone"
 	prod.Mode = core.ModeActiveProducer
+	prod.Protocol = "http"
+	prod.URL = rawURL
 
 	offer, err := mc.GetOffer()
 	if err != nil {

internal/webrtc/openipc.go

@@ -9,7 +9,7 @@ import (
 	"github.com/AlexxIT/go2rtc/pkg/core"
 	"github.com/AlexxIT/go2rtc/pkg/webrtc"
 	"github.com/gorilla/websocket"
-	pion "github.com/pion/webrtc/v3"
+	pion "github.com/pion/webrtc/v4"
 )
 
 func openIPCClient(rawURL string, query url.Values) (core.Producer, error) {
@@ -53,8 +53,10 @@ func openIPCClient(rawURL string, query url.Values) (core.Producer, error) {
 	var connState core.Waiter
 
 	prod := webrtc.NewConn(pc)
-	prod.Desc = "WebRTC/OpenIPC"
+	prod.FormatName = "webrtc/openipc"
 	prod.Mode = core.ModeActiveProducer
+	prod.Protocol = "ws"
+	prod.URL = rawURL
 	prod.Listen(func(msg any) {
 		switch msg := msg.(type) {
 		case *pion.ICECandidate:

internal/webrtc/server.go

@@ -1,6 +1,7 @@
 package webrtc
 
 import (
+	"encoding/base64"
 	"encoding/json"
 	"io"
 	"net/http"
@@ -12,7 +13,7 @@ import (
 	"github.com/AlexxIT/go2rtc/internal/streams"
 	"github.com/AlexxIT/go2rtc/pkg/core"
 	"github.com/AlexxIT/go2rtc/pkg/webrtc"
-	pion "github.com/pion/webrtc/v3"
+	pion "github.com/pion/webrtc/v4"
 )
 
 const MimeSDP = "application/sdp"
@@ -62,9 +63,10 @@ func syncHandler(w http.ResponseWriter, r *http.Request) {
 // 2. application/sdp - receive/response SDP via WebRTC-HTTP Egress Protocol (WHEP)
 // 3. other - receive/response raw SDP
 func outputWebRTC(w http.ResponseWriter, r *http.Request) {
-	url := r.URL.Query().Get("src")
-	stream := streams.Get(url)
+	u := r.URL.Query().Get("src")
+	stream := streams.Get(u)
 	if stream == nil {
+		http.Error(w, api.StreamNotFound, http.StatusNotFound)
 		return
 	}
 
@@ -86,6 +88,21 @@ func outputWebRTC(w http.ResponseWriter, r *http.Request) {
 		}
 		offer = desc.SDP
 
+	case "application/x-www-form-urlencoded":
+		if err := r.ParseForm(); err != nil {
+			log.Error().Err(err).Caller().Send()
+			http.Error(w, err.Error(), http.StatusBadRequest)
+			return
+		}
+		offerB64 := r.Form.Get("data")
+		b, err := base64.StdEncoding.DecodeString(offerB64)
+		if err != nil {
+			log.Error().Err(err).Caller().Send()
+			http.Error(w, err.Error(), http.StatusBadRequest)
+			return
+		}
+		offer = string(b)
+
 	default:
 		body, err := io.ReadAll(r.Body)
 		if err != nil {
@@ -100,11 +117,11 @@ func outputWebRTC(w http.ResponseWriter, r *http.Request) {
 
 	switch mediaType {
 	case "application/json":
-		desc = "WebRTC/JSON sync"
+		desc = "webrtc/json"
 	case MimeSDP:
-		desc = "WebRTC/WHEP sync"
+		desc = "webrtc/whep"
 	default:
-		desc = "WebRTC/HTTP sync"
+		desc = "webrtc/post"
 	}
 
 	answer, err := ExchangeSDP(stream, offer, desc, r.UserAgent())
@@ -123,6 +140,11 @@ func outputWebRTC(w http.ResponseWriter, r *http.Request) {
 		}
 		err = json.NewEncoder(w).Encode(v)
 
+	case "application/x-www-form-urlencoded":
+		w.Header().Set("Content-Type", mediaType)
+		answerB64 := base64.StdEncoding.EncodeToString([]byte(answer))
+		_, err = w.Write([]byte(answerB64))
+
 	case MimeSDP:
 		w.Header().Set("Content-Type", mediaType)
 		w.WriteHeader(http.StatusCreated)
@@ -168,8 +190,8 @@ func inputWebRTC(w http.ResponseWriter, r *http.Request) {
 
 	// create new webrtc instance
 	prod := webrtc.NewConn(pc)
-	prod.Desc = "WebRTC/WHIP sync"
 	prod.Mode = core.ModePassiveProducer
+	prod.Protocol = "http"
 	prod.UserAgent = r.UserAgent()
 
 	if err = prod.SetOffer(string(offer)); err != nil {

internal/webrtc/switchbot.go

@@ -0,0 +1,40 @@
+package webrtc
+
+import (
+	"net/url"
+
+	"github.com/AlexxIT/go2rtc/pkg/core"
+	"github.com/AlexxIT/go2rtc/pkg/webrtc"
+)
+
+func switchbotClient(rawURL string, query url.Values) (core.Producer, error) {
+	return kinesisClient(rawURL, query, "webrtc/switchbot", func(prod *webrtc.Conn, query url.Values) (any, error) {
+		medias := []*core.Media{
+			{Kind: core.KindVideo, Direction: core.DirectionRecvonly},
+		}
+
+		offer, err := prod.CreateOffer(medias)
+		if err != nil {
+			return nil, err
+		}
+
+		v := struct {
+			Type       string `json:"type"`
+			SDP        string `json:"sdp"`
+			Resolution int    `json:"resolution"`
+			PlayType   int    `json:"play_type"`
+		}{
+			Type: "offer",
+			SDP:  offer,
+		}
+
+		switch query.Get("resolution") {
+		case "hd":
+			v.Resolution = 0
+		case "sd":
+			v.Resolution = 1
+		}
+
+		return v, nil
+	})
+}

internal/webrtc/webrtc.go

@@ -10,7 +10,7 @@ import (
 	"github.com/AlexxIT/go2rtc/internal/streams"
 	"github.com/AlexxIT/go2rtc/pkg/core"
 	"github.com/AlexxIT/go2rtc/pkg/webrtc"
-	pion "github.com/pion/webrtc/v3"
+	pion "github.com/pion/webrtc/v4"
 	"github.com/rs/zerolog"
 )
 
@@ -24,7 +24,7 @@ func Init() {
 		} `yaml:"webrtc"`
 	}
 
-	cfg.Mod.Listen = ":8555/tcp"
+	cfg.Mod.Listen = ":8555"
 	cfg.Mod.IceServers = []pion.ICEServer{
 		{URLs: []string{"stun:stun.l.google.com:19302"}},
 	}
@@ -40,15 +40,17 @@ func Init() {
 		AddCandidate(network, candidate)
 	}
 
+	var err error
+
 	// create pionAPI with custom codecs list and custom network settings
-	serverAPI, err := webrtc.NewServerAPI(network, address, &filters)
+	serverAPI, err = webrtc.NewServerAPI(network, address, &filters)
 	if err != nil {
 		log.Error().Err(err).Caller().Send()
 		return
 	}
 
 	// use same API for WebRTC server and client if no address
-	clientAPI := serverAPI
+	clientAPI = serverAPI
 
 	if address != "" {
 		log.Info().Str("addr", cfg.Mod.Listen).Msg("[webrtc] listen")
@@ -81,11 +83,13 @@ func Init() {
 	streams.HandleFunc("webrtc", streamsHandler)
 }
 
+var serverAPI, clientAPI *pion.API
+
 var log zerolog.Logger
 
 var PeerConnection func(active bool) (*pion.PeerConnection, error)
 
-func asyncHandler(tr *ws.Transport, msg *ws.Message) error {
+func asyncHandler(tr *ws.Transport, msg *ws.Message) (err error) {
 	var stream *streams.Stream
 	var mode core.Mode
 
@@ -104,8 +108,30 @@ func asyncHandler(tr *ws.Transport, msg *ws.Message) error {
 		return errors.New(api.StreamNotFound)
 	}
 
+	var offer struct {
+		Type       string           `json:"type"`
+		SDP        string           `json:"sdp"`
+		ICEServers []pion.ICEServer `json:"ice_servers"`
+	}
+
+	// V2 - json/object exchange, V1 - raw SDP exchange
+	apiV2 := msg.Type == "webrtc"
+
+	if apiV2 {
+		if err = msg.Unmarshal(&offer); err != nil {
+			return err
+		}
+	} else {
+		offer.SDP = msg.String()
+	}
+
 	// create new PeerConnection instance
-	pc, err := PeerConnection(false)
+	var pc *pion.PeerConnection
+	if offer.ICEServers == nil {
+		pc, err = PeerConnection(false)
+	} else {
+		pc, err = serverAPI.NewPeerConnection(pion.Configuration{ICEServers: offer.ICEServers})
+	}
 	if err != nil {
 		log.Error().Err(err).Caller().Send()
 		return err
@@ -117,8 +143,8 @@ func asyncHandler(tr *ws.Transport, msg *ws.Message) error {
 	defer sendAnswer.Done(nil)
 
 	conn := webrtc.NewConn(pc)
-	conn.Desc = "WebRTC/WebSocket async"
 	conn.Mode = mode
+	conn.Protocol = "ws"
 	conn.UserAgent = tr.Request.UserAgent()
 	conn.Listen(func(msg any) {
 		switch msg := msg.(type) {
@@ -145,20 +171,10 @@ func asyncHandler(tr *ws.Transport, msg *ws.Message) error {
 		}
 	})
 
-	// V2 - json/object exchange, V1 - raw SDP exchange
-	apiV2 := msg.Type == "webrtc"
+	log.Trace().Msgf("[webrtc] offer:\n%s", offer.SDP)
 
 	// 1. SetOffer, so we can get remote client codecs
-	var offer string
-	if apiV2 {
-		offer = msg.GetString("sdp")
-	} else {
-		offer = msg.String()
-	}
-
-	log.Trace().Msgf("[webrtc] offer:\n%s", offer)
-
-	if err = conn.SetOffer(offer); err != nil {
+	if err = conn.SetOffer(offer.SDP); err != nil {
 		log.Warn().Err(err).Caller().Send()
 		return err
 	}
@@ -207,8 +223,9 @@ func ExchangeSDP(stream *streams.Stream, offer, desc, userAgent string) (answer
 
 	// create new webrtc instance
 	conn := webrtc.NewConn(pc)
-	conn.Desc = desc
+	conn.FormatName = desc
 	conn.UserAgent = userAgent
+	conn.Protocol = "http"
 	conn.Listen(func(msg any) {
 		switch msg := msg.(type) {
 		case pion.PeerConnectionState:

internal/webrtc/webrtc_test.go

@@ -0,0 +1,73 @@
+package webrtc
+
+import (
+	"encoding/json"
+	"strings"
+	"testing"
+
+	"github.com/AlexxIT/go2rtc/internal/api/ws"
+	pion "github.com/pion/webrtc/v4"
+	"github.com/stretchr/testify/require"
+)
+
+func TestWebRTCAPIv1(t *testing.T) {
+	raw := `{"type":"webrtc/offer","value":"v=0\n..."}`
+	msg := new(ws.Message)
+	err := json.Unmarshal([]byte(raw), msg)
+	require.Nil(t, err)
+
+	require.Equal(t, "v=0\n...", msg.String())
+}
+
+func TestWebRTCAPIv2(t *testing.T) {
+	raw := `{"type":"webrtc","value":{"type":"offer","sdp":"v=0\n...","ice_servers":[{"urls":["stun:stun.l.google.com:19302"]}]}}`
+	msg := new(ws.Message)
+	err := json.Unmarshal([]byte(raw), msg)
+	require.Nil(t, err)
+
+	var offer struct {
+		Type       string           `json:"type"`
+		SDP        string           `json:"sdp"`
+		ICEServers []pion.ICEServer `json:"ice_servers"`
+	}
+	err = msg.Unmarshal(&offer)
+	require.Nil(t, err)
+
+	require.Equal(t, "offer", offer.Type)
+	require.Equal(t, "v=0\n...", offer.SDP)
+	require.Equal(t, "stun:stun.l.google.com:19302", offer.ICEServers[0].URLs[0])
+}
+
+func TestCrealitySDP(t *testing.T) {
+	sdp := `v=0
+o=- 1495799811084970 1495799811084970 IN IP4 0.0.0.0
+s=-
+t=0 0
+a=msid-semantic:WMS *
+a=group:BUNDLE 0
+m=video 9 UDP/TLS/RTP/SAVPF 96 98
+a=rtcp-fb:98 nack
+a=rtcp-fb:98 nack pli
+a=fmtp:96 profile-level-id=42e01f;level-asymmetry-allowed=1
+a=fmtp:98 profile-level-id=42e01f;packetization-mode=1;level-asymmetry-allowed=1
+a=fmtp:98 x-google-max-bitrate=6000;x-google-min-bitrate=2000;x-google-start-bitrate=4000
+a=rtpmap:96 H264/90000
+a=rtpmap:98 H264/90000
+a=ssrc:1 cname:pear
+c=IN IP4 0.0.0.0
+a=sendonly
+a=mid:0
+a=rtcp-mux
+a=ice-ufrag:7AVa
+a=ice-pwd:T+F/5y05Paw+mtG5Jrd8N3
+a=ice-options:trickle
+a=fingerprint:sha-256 A5:AB:C0:4E:29:5B:BD:3B:7D:88:24:6C:56:6B:2A:79:A3:76:99:35:57:75:AD:C8:5A:A6:34:20:88:1B:68:EF
+a=setup:passive
+a=candidate:1 1 UDP 2015363327 172.22.233.10 48929 typ host
+a=candidate:2 1 TCP 1015021823 172.22.233.10 0 typ host tcptype active
+a=candidate:3 1 TCP 1010827519 172.22.233.10 60677 typ host tcptype passive
+`
+	sdp, err := fixCrealitySDP(sdp)
+	require.Nil(t, err)
+	require.False(t, strings.Contains(sdp, "x-google-max-bitrate"))
+}

internal/webtorrent/init.go

@@ -47,7 +47,7 @@ func Init() {
 			if stream == nil {
 				return "", errors.New(api.StreamNotFound)
 			}
-			return webrtc.ExchangeSDP(stream, offer, "WebRTC/WebTorrent sync", "")
+			return webrtc.ExchangeSDP(stream, offer, "webtorrent", "")
 		},
 	}
 

internal/webtorrent/tracker.go

@@ -2,9 +2,10 @@ package webtorrent
 
 import (
 	"fmt"
+	"net/http"
+
 	"github.com/AlexxIT/go2rtc/pkg/webtorrent"
 	"github.com/gorilla/websocket"
-	"net/http"
 )
 
 var upgrader *websocket.Upgrader

internal/wyoming/README.md

@@ -0,0 +1,281 @@
+# Wyoming
+
+This module provide [Wyoming Protocol](https://www.home-assistant.io/integrations/wyoming/) support to create local voice assistants using [Home Assistant](https://www.home-assistant.io/).
+
+- go2rtc can act as [Wyoming Satellite](https://github.com/rhasspy/wyoming-satellite)
+- go2rtc can act as [Wyoming External Microphone](https://github.com/rhasspy/wyoming-mic-external)
+- go2rtc can act as [Wyoming External Sound](https://github.com/rhasspy/wyoming-snd-external)
+- any supported audio source with PCM codec can be used as audio input
+- any supported two-way audio source with PCM codec can be used as audio output
+- any desktop/server microphone/speaker can be used as two-way audio source
+  - supported any OS via FFmpeg or any similar software
+  - supported Linux via alsa source
+- you can change the behavior using the built-in scripting engine
+
+## Typical Voice Pipeline
+
+1. Audio stream (MIC)
+   - any audio source with PCM codec support (include PCMA/PCMU)
+2. Voice Activity Detector (VAD)
+3. Wake Word (WAKE)
+   - [OpenWakeWord](https://www.home-assistant.io/voice_control/create_wake_word/)
+4. Speech-to-Text (STT)
+   - [Whisper](https://github.com/home-assistant/addons/blob/master/whisper/README.md) 
+   - [Vosk](https://github.com/rhasspy/hassio-addons/blob/master/vosk/README.md)
+5. Conversation agent (INTENT)
+   - [Home Assistant](https://www.home-assistant.io/integrations/conversation/)
+6. Text-to-speech (TTS)
+   - [Google Translate](https://www.home-assistant.io/integrations/google_translate/)
+   - [Piper](https://github.com/home-assistant/addons/blob/master/piper/README.md)
+7. Audio stream (SND)
+   - any source with two-way audio (backchannel) and PCM codec support (include PCMA/PCMU)
+
+You can use a large number of different projects for WAKE, STT, INTENT and TTS thanks to the Home Assistant.
+
+And you can use a large number of different technologies for MIC and SND thanks to Go2rtc.
+
+## Configuration
+
+You can optionally specify WAKE service. So go2rtc will start transmitting audio to Home Assistant only after WAKE word. If the WAKE service cannot be connected to or not specified - go2rtc will pass all audio to Home Assistant. In this case WAKE service must be configured in your Voice Assistant pipeline.
+
+You can optionally specify VAD threshold. So go2rtc will start transmitting audio to WAKE service only after some audio noise.
+
+Your stream must support audio transmission in PCM codec (include PCMA/PCMU).
+
+```yaml
+wyoming:
+  stream_name_from_streams_section:
+    listen: :10700 
+    name: "My Satellite"                # optional name
+    wake_uri: tcp://192.168.1.23:10400  # optional WAKE service
+    vad_threshold: 1                    # optional VAD threshold (from 0.1 to 3.5)
+```
+
+Home Assistant -> Settings -> Integrations -> Add -> Wyoming Protocol -> Host + Port from `go2rtc.yaml`
+
+Select one or multiple wake words:
+```yaml
+wake_uri: tcp://192.168.1.23:10400?name=alexa_v0.1&name=hey_jarvis_v0.1&name=hey_mycroft_v0.1&name=hey_rhasspy_v0.1&name=ok_nabu_v0.1
+```
+
+## Events
+
+You can add wyoming event handling using the [expr](https://github.com/AlexxIT/go2rtc/blob/master/internal/expr/README.md) language. For example, to pronounce TTS on some media player from HA.
+
+Turn on the logs to see what kind of events happens.
+
+This is what the default scripts look like:
+
+```yaml
+wyoming:
+  script_example:
+    event:
+      run-satellite: Detect()
+      pause-satellite: Stop()
+      voice-stopped: Pause()
+      audio-stop: PlayAudio() && WriteEvent("played") && Detect()
+      error: Detect()
+      internal-run: WriteEvent("run-pipeline", '{"start_stage":"wake","end_stage":"tts"}') && Stream()
+      internal-detection: WriteEvent("run-pipeline", '{"start_stage":"asr","end_stage":"tts"}') && Stream()
+```
+
+Supported functions and variables:
+
+- `Detect()` - start the VAD and WAKE word detection process
+- `Stream()` - start transmission of audio data to the client (Home Assistant)
+- `Stop()` - stop and disconnect stream without disconnecting client (Home Assistant)
+- `Pause()` - temporary pause of audio transfer, without disconnecting the stream
+- `PlayAudio()` - playing the last audio that was sent from client (Home Assistant)
+- `WriteEvent(type, data)` - send event to client (Home Assistant)
+- `Sleep(duration)` - temporary script pause (ex. `Sleep('1.5s')`)
+- `PlayFile(path)` - play audio from `wav` file
+- `Type` - type (name) of event
+- `Data` - event data in JSON format (ex. `{"text":"how are you"}`)
+- also available other functions from [expr](https://github.com/AlexxIT/go2rtc/blob/master/internal/expr/README.md) module (ex. `fetch`)
+
+If you write a script for an event - the default action is no longer executed. You need to repeat the necessary steps yourself.
+
+In addition to the standard events, there are two additional events:
+
+- `internal-run` - called after `Detect()` when VAD detected, but WAKE service unavailable
+- `internal-detection` - called after `Detect()` when WAKE word detected
+
+**Example 1.** You want to play a sound file when a wake word detected (only `wav` supported):
+
+- `PlayFile` and `PlayAudio` functions are executed synchronously, the following steps will be executed only after they are completed
+
+```yaml
+wyoming:
+  script_example:
+    event:
+      internal-detection: PlayFile('/media/beep.wav') && WriteEvent("run-pipeline", '{"start_stage":"asr","end_stage":"tts"}') && Stream()
+```
+
+**Example 2.** You want to play TTS on a Home Assistant media player:
+
+Each event has a `Type` and `Data` in JSON format. You can use their values in scripts.
+
+- in the `synthesize` step, we get the value of the `text` and call the HA REST API
+- in the `audio-stop` step we get the duration of the TTS in seconds, wait for this time and start the pipeline again
+
+```yaml
+wyoming:
+  script_example:
+    event:
+      synthesize: |
+        let text = fromJSON(Data).text;
+        let token = 'eyJhbGci...';
+        fetch('http://localhost:8123/api/services/tts/speak', {
+          method: 'POST',
+          headers: {'Authorization': 'Bearer '+token,'Content-Type': 'application/json'},
+          body: toJSON({
+            entity_id: 'tts.google_translate_com',
+            media_player_entity_id: 'media_player.google_nest',
+            message: text,
+            language: 'en',
+          }),
+        }).ok
+      audio-stop: |
+        let timestamp = fromJSON(Data).timestamp;
+        let delay = string(timestamp)+'s';
+        Sleep(delay) && WriteEvent("played") && Detect()
+```
+
+## Config examples
+
+Satellite on Windows server using FFmpeg and FFplay.
+
+```yaml
+streams:
+  satellite_win:
+    - exec:ffmpeg -hide_banner -f dshow -i "audio=Microphone (High Definition Audio Device)" -c pcm_s16le -ar 16000 -ac 1 -f wav -
+    - exec:ffplay -hide_banner -nodisp -probesize 32 -f s16le -ar 22050 -#backchannel=1#audio=s16le/22050
+
+wyoming:
+  satellite_win:
+    listen: :10700
+    name: "Windows Satellite"
+    wake_uri: tcp://192.168.1.23:10400
+    vad_threshold: 1
+```
+
+Satellite on Dahua camera with two-way audio support.
+
+```yaml
+streams:
+  dahua_camera:
+    - rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=1&unicast=true&proto=Onvif
+
+wyoming:
+  dahua_camera:
+    listen: :10700
+    name: "Dahua Satellite"
+    wake_uri: tcp://192.168.1.23:10400
+    vad_threshold: 1
+```
+
+Satellite on external wyoming Microphone and Sound.
+
+```yaml
+streams:
+  wyoming_external:
+     - wyoming://192.168.1.23:10600                # wyoming-mic-external
+     - wyoming://192.168.1.23:10601?backchannel=1  # wyoming-snd-external
+
+wyoming:
+   wyoming_external:
+    listen: :10700
+    name: "Wyoming Satellite"
+    wake_uri: tcp://192.168.1.23:10400
+    vad_threshold: 1
+```
+
+## Wyoming External Microphone and Sound
+
+Advanced users, who want to enjoy the [Wyoming Satellite](https://github.com/rhasspy/wyoming-satellite) project, can use go2rtc as a [Wyoming External Microphone](https://github.com/rhasspy/wyoming-mic-external) or [Wyoming External Sound](https://github.com/rhasspy/wyoming-snd-external).
+
+**go2rtc.yaml**
+
+```yaml
+streams:
+  wyoming_mic_external:
+    - exec:ffmpeg -hide_banner -f dshow -i "audio=Microphone (High Definition Audio Device)" -c pcm_s16le -ar 16000 -ac 1 -f wav -
+  wyoming_snd_external:
+    - exec:ffplay -hide_banner -nodisp -probesize 32 -f s16le -ar 22050 -#backchannel=1#audio=s16le/22050
+
+wyoming:
+  wyoming_mic_external:
+    listen: :10600
+    mode: mic
+  wyoming_snd_external:
+    listen: :10601
+    mode: snd
+```
+
+**docker-compose.yml**
+
+```yaml
+version: "3.8"
+services:
+  satellite:
+    build: wyoming-satellite  # https://github.com/rhasspy/wyoming-satellite
+    ports:
+      - "10700:10700"
+    command:
+      - "--name"
+      - "my satellite"
+      - "--mic-uri"
+      - "tcp://192.168.1.23:10600"
+      - "--snd-uri"
+      - "tcp://192.168.1.23:10601"
+      - "--debug"
+```
+
+## Wyoming External Source
+
+**go2rtc.yaml**
+
+```yaml
+streams:
+  wyoming_external:
+    - wyoming://192.168.1.23:10600
+    - wyoming://192.168.1.23:10601?backchannel=1
+```
+
+**docker-compose.yml**
+
+```yaml
+version: "3.8"
+services:
+   microphone:
+      build: wyoming-mic-external  # https://github.com/rhasspy/wyoming-mic-external
+      ports:
+         - "10600:10600"
+      devices:
+         - /dev/snd:/dev/snd
+      group_add:
+         - audio
+      command:
+         - "--device"
+         - "sysdefault"
+         - "--debug"
+   playback:
+      build: wyoming-snd-external  # https://github.com/rhasspy/wyoming-snd-external
+      ports:
+         - "10601:10601"
+      devices:
+         - /dev/snd:/dev/snd
+      group_add:
+         - audio
+      command:
+         - "--device"
+         - "sysdefault"
+         - "--debug"
+```
+
+## Debug
+
+```yaml
+log:
+  wyoming: trace
+```

internal/wyoming/wyoming.go

@@ -0,0 +1,106 @@
+package wyoming
+
+import (
+	"net"
+
+	"github.com/AlexxIT/go2rtc/internal/app"
+	"github.com/AlexxIT/go2rtc/internal/streams"
+	"github.com/AlexxIT/go2rtc/pkg/core"
+	"github.com/AlexxIT/go2rtc/pkg/wyoming"
+	"github.com/rs/zerolog"
+)
+
+func Init() {
+	streams.HandleFunc("wyoming", wyoming.Dial)
+
+	// server
+	var cfg struct {
+		Mod map[string]struct {
+			Listen       string            `yaml:"listen"`
+			Name         string            `yaml:"name"`
+			Mode         string            `yaml:"mode"`
+			Event        map[string]string `yaml:"event"`
+			WakeURI      string            `yaml:"wake_uri"`
+			VADThreshold float32           `yaml:"vad_threshold"`
+		} `yaml:"wyoming"`
+	}
+	app.LoadConfig(&cfg)
+
+	log = app.GetLogger("wyoming")
+
+	for name, cfg := range cfg.Mod {
+		stream := streams.Get(name)
+		if stream == nil {
+			log.Warn().Msgf("[wyoming] missing stream: %s", name)
+			continue
+		}
+
+		if cfg.Name == "" {
+			cfg.Name = name
+		}
+
+		srv := &wyoming.Server{
+			Name:         cfg.Name,
+			Event:        cfg.Event,
+			VADThreshold: int16(1000 * cfg.VADThreshold), // 1.0 => 1000
+			WakeURI:      cfg.WakeURI,
+			MicHandler: func(cons core.Consumer) error {
+				if err := stream.AddConsumer(cons); err != nil {
+					return err
+				}
+				// not best solution
+				if i, ok := cons.(interface{ OnClose(func()) }); ok {
+					i.OnClose(func() {
+						stream.RemoveConsumer(cons)
+					})
+				}
+				return nil
+			},
+			SndHandler: func(prod core.Producer) error {
+				return stream.Play(prod)
+			},
+			Trace: func(format string, v ...any) {
+				log.Trace().Msgf("[wyoming] "+format, v...)
+			},
+			Error: func(format string, v ...any) {
+				log.Error().Msgf("[wyoming] "+format, v...)
+			},
+		}
+		go serve(srv, cfg.Mode, cfg.Listen)
+	}
+}
+
+var log zerolog.Logger
+
+func serve(srv *wyoming.Server, mode, address string) {
+	ln, err := net.Listen("tcp", address)
+	if err != nil {
+		log.Warn().Err(err).Msgf("[wyoming] listen")
+	}
+
+	for {
+		conn, err := ln.Accept()
+		if err != nil {
+			return
+		}
+
+		go handle(srv, mode, conn)
+	}
+}
+
+func handle(srv *wyoming.Server, mode string, conn net.Conn) {
+	addr := conn.RemoteAddr()
+
+	log.Trace().Msgf("[wyoming] %s connected", addr)
+
+	switch mode {
+	case "mic":
+		srv.HandleMic(conn)
+	case "snd":
+		srv.HandleSnd(conn)
+	default:
+		srv.Handle(conn)
+	}
+
+	log.Trace().Msgf("[wyoming] %s disconnected", addr)
+}

internal/yandex/README.md

@@ -0,0 +1,22 @@
+# Yandex
+
+Source for receiving stream from new [Yandex IP camera](https://alice.yandex.ru/smart-home/security/ipcamera).
+
+## Get Yandex token
+
+1. Install HomeAssistant integration [YandexStation](https://github.com/AlexxIT/YandexStation).
+2. Copy token from HomeAssistant config folder: `/config/.storage/core.config_entries`, key: `"x_token"`.
+
+## Get device ID
+
+1. Open this link in any browser: https://iot.quasar.yandex.ru/m/v3/user/devices
+2. Copy ID of your camera, key: `"id"`.
+
+## Config examples
+
+```yaml
+streams:
+  yandex_stream: yandex:?x_token=XXXX&device_id=XXXX
+  yandex_snapshot: yandex:?x_token=XXXX&device_id=XXXX&snapshot
+  yandex_snapshot_custom_size: yandex:?x_token=XXXX&device_id=XXXX&snapshot=h=540
+```