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38 Commits

Author SHA1 Message Date
Alexey Khit 64247fc90f Adds support local USB cameras 2022-08-26 10:43:35 +03:00
Alexey Khit c019dc58b1 Update panic message for RTMP 2022-08-26 10:10:45 +03:00
Alexey Khit d3adaf05b1 Update build scripts 2022-08-26 10:09:52 +03:00
Alexey Khit e6cfd1818b Adds on the fly add producer feature 2022-08-26 10:07:35 +03:00
Alexey Khit fae4398d21 Adds FAQ to readme 2022-08-26 07:22:12 +03:00
Alexey Khit 5daf043937 Update readme 2022-08-25 14:21:34 +03:00
Alexey Khit 18d7b9075b Autoadd cameras from Hass config 2022-08-25 06:41:39 +03:00
Alexey Khit 7c4497f856 Fix incoming RTSP without OPTIONS 2022-08-25 06:38:08 +03:00
Alexey Khit befa4ca1e6 Remove wrong RTSP channel panic 2022-08-25 06:37:47 +03:00
Alexey Khit dd3b326f7a Update readme 2022-08-24 14:41:00 +03:00
Alexey Khit e36123bb19 Update build docker 2022-08-24 14:30:02 +03:00
Alexey Khit 9310343ad3 Update docker cwd to /config 2022-08-24 12:49:59 +03:00
Alexey Khit e2d4fa3393 Advanced debug on app start 2022-08-24 12:49:48 +03:00
Alexey Khit 5fea2932c1 Error on wrong config 2022-08-24 12:49:40 +03:00
Alexey Khit 1fd110b70d Update readme 2022-08-24 09:55:34 +03:00
Alexey Khit 8377cf2655 Change url param to src in Web API 2022-08-24 09:55:16 +03:00
Alexey Khit 8f01b08d42 Code refactoring 2022-08-24 09:54:28 +03:00
Alexey Khit 97ce4c3114 Adds Security section to readme 2022-08-23 09:34:06 +03:00
Alexey Khit 4813a64d9d Adds build script for mips 2022-08-23 05:43:15 +03:00
Alexey Khit 7923ec74a8 Adds network filter for webrtc 2022-08-23 05:43:01 +03:00
Alexey Khit 1f0a5fb880 Stop WebRTC conn on AddConsumer error 2022-08-22 22:46:08 +03:00
Alexey Khit c6a3ee65b8 Remove UPX from Windows builds because antiviruses 2022-08-22 22:32:23 +03:00
Alexey Khit 12b712426d Fix busy RTSP backchannel 2022-08-22 15:41:25 +03:00
Alexey Khit a9af245ef8 Fix async requests to Producer 2022-08-22 15:40:28 +03:00
Alexey Khit f251129a2f Fix RTSP Transport header parsing 2022-08-22 14:46:39 +03:00
Alexey Khit d28debabe9 Update fix for parsing RTSP SDP 2022-08-22 14:44:33 +03:00
Alexey Khit 07bf00f9f6 Update readme 2022-08-22 13:40:58 +03:00
Alexey Khit be6ec7dbb9 Fix RTSP requests for some cameras 2022-08-22 13:38:26 +03:00
Alexey Khit 4e575d1356 Adds build file for win64 2022-08-22 11:43:42 +03:00
Alexey Khit 4cbacfec0c Adds empty response on RTSP error 2022-08-22 11:43:26 +03:00
Alexey Khit 31e24c6e03 Adds stop with empty producer warning 2022-08-22 11:33:38 +03:00
Alexey Khit 401bf85a10 Update RTSP error output 2022-08-22 09:09:18 +03:00
Alexey Khit f36851f83a Fix response with empty producer 2022-08-22 09:06:40 +03:00
Alexey Khit 67522dbb19 Update readme 2022-08-22 08:44:27 +03:00
Alexey Khit 26b5745f0a Adds keep-alive to RTSP connection 2022-08-22 06:54:58 +03:00
Alexey Khit 46f6a5d8e1 Return unmodified errors from RTSP 2022-08-22 06:54:42 +03:00
Alexey Khit 48f58d0669 Fix wrong stream name request 2022-08-22 06:54:08 +03:00
Alexey Khit fd0b8f3c39 Fix RTMP with audio 2022-08-22 05:46:22 +03:00
41 changed files with 819 additions and 199 deletions
+97 -43
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@@ -1,20 +1,21 @@
# go2rtc
**go2rtc** - ultimate camera streaming application with support RTSP, WebRTC, FFmpeg, RTMP, etc.
Ultimate camera streaming application with support RTSP, WebRTC, FFmpeg, RTMP, etc.
- zero-dependency and zero-config small [app for all OS](#installation) (Windows, macOS, Linux, ARM)
- zero-dependency and zero-config [small app](#go2rtc-binary) for all OS (Windows, macOS, Linux, ARM)
- zero-delay for all supported protocols (lowest possible streaming latency)
- zero-load on CPU for supported codecs
- low CPU load for supported codecs
- on the fly transcoding for unsupported codecs via [FFmpeg](#source-ffmpeg)
- multi-source 2-way [codecs negotiation](#codecs-negotiation)
- streaming from private networks via [Ngrok](#module-webrtc)
- 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)
**Inspired by:**
- [webrtc](https://github.com/pion/webrtc) go library and whole [@pion](https://github.com/pion) team
- series of streaming projects from [@deepch](https://github.com/deepch)
- [webrtc](https://github.com/pion/webrtc) go library and whole [@pion](https://github.com/pion) team
- [rtsp-simple-server](https://github.com/aler9/rtsp-simple-server) idea from [@aler9](https://github.com/aler9)
- [GStreamer](https://gstreamer.freedesktop.org/) multimedia framework pipeline idea
- [GStreamer](https://gstreamer.freedesktop.org/) framework pipeline idea
- [MediaSoup](https://mediasoup.org/) framework routing idea
## Codecs negotiation
@@ -30,25 +31,26 @@ For example, you want to watch RTSP-stream from [Dahua IPC-K42](https://www.dahu
- 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
- 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**
**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.
**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.
Now you have stream with two sources - **RTSP and FFmpeg**:
```yaml
streams:
dahua:
- rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif
- ffmpeg:rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif#audio=opus
- 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.
![](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.
## Fast start
1. Download [binary](#go2rtc-binary) or use [Docker](#go2rtc-docker) or [Home Assistant Add-on](#go2rtc-home-assistant-add-on)
2. Open web interface [http://localhost:1984/](http://localhost:1984/)
2. Open web interface: `http://localhost:1984/`
**Optionally:**
@@ -76,7 +78,7 @@ Download binary for your OS from [latest release](https://github.com/AlexxIT/go2
- `go2rtc_mac_amd64` - Mac with Intel
- `go2rtc_mac_arm64` - Mac with M1
Don't forget to fix the rights `chmod +x go2rtc_linux_xxx` on Linux and Mac.
Don't forget to fix the rights `chmod +x go2rtc_xxx_xxx` on Linux and Mac.
### go2rtc: Home Assistant Add-on
@@ -95,6 +97,16 @@ Don't forget to fix the rights `chmod +x go2rtc_linux_xxx` on Linux and Mac.
Container [alexxit/go2rtc](https://hub.docker.com/r/alexxit/go2rtc) with support `amd64`, `386`, `arm64`, `arm`. This container same as [Home Assistant Add-on](#go2rtc-home-assistant-add-on), but can be used separately from the Home Assistant. Container has preinstalled [FFmpeg](#source-ffmpeg) and [Ngrok](#module-ngrok) applications.
```yaml
services:
go2rtc:
image: alexxit/go2rtc
network_mode: host
restart: always
volumes:
- "~/go2rtc.yaml:/config/go2rtc.yaml"
```
## Configuration
Create file `go2rtc.yaml` next to the app.
@@ -103,7 +115,7 @@ Create file `go2rtc.yaml` next to the app.
- `api` server will start on default **1984 port**
- `rtsp` server will start on default **8554 port**
- `webrtc` will use random UDP port for each connection
- `ffmpeg` will use default transcoding options (you need to install it [manually](https://ffmpeg.org/))
- `ffmpeg` will use default transcoding options (you may install it [manually](https://ffmpeg.org/))
Available modules:
@@ -118,7 +130,7 @@ Available modules:
### Module: Streams
**go2rtc** support different stream source types. You can config only one link as stream source or multiple.
**go2rtc** support different stream source types. You can config one or multiple links of any type as stream source.
Available source types:
@@ -128,12 +140,14 @@ Available source types:
- [exec](#source-exec) - advanced FFmpeg and GStreamer integration
- [hass](#source-hass) - Home Assistant integration
**PS.** You can use sources like `MJPEG`, `HLS` and others via FFmpeg integration.
#### Source: RTSP
- Support **RTSP and RTSPS** links with multiple video and audio tracks
- Support **2-way audio** ONLY for [ONVIF Profile T](https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf) cameras (back channel connection)
**Attention:** proprietary 2-way audio standards are not supported!
**Attention:** other 2-way audio standards are not supported! ONVIF without Profile T is not supported!
```yaml
streams:
@@ -164,18 +178,21 @@ streams:
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.
- 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
Format: `ffmpeg:{input}#{param1}#{param2}#{param3}`. Examples:
```yaml
streams:
# [FILE] all tracks will be copied without transcoding codecs
file1: ffmpeg:~/media/BigBuckBunny.mp4
file1: ffmpeg:/media/BigBuckBunny.mp4
# [FILE] video will be transcoded to H264, audio will be skipped
file2: ffmpeg:~/media/BigBuckBunny.mp4#video=h264
file2: ffmpeg:/media/BigBuckBunny.mp4#video=h264
# [FILE] video will be copied, audio will be transcoded to pcmu
file3: ffmpeg:~/media/BigBuckBunny.mp4#video=copy#audio=pcmu
file3: ffmpeg:/media/BigBuckBunny.mp4#video=copy#audio=pcmu
# [HLS] video will be copied, audio will be skipped
hls: ffmpeg:https://devstreaming-cdn.apple.com/videos/streaming/examples/bipbop_16x9/gear5/prog_index.m3u8#video=copy
@@ -187,28 +204,15 @@ streams:
rtsp: ffmpeg:rtsp://rtsp:12345678@192.168.1.123/av_stream/ch0#video=copy#audio=copy
```
All trascoding formats has built-in templates. But you can override them via YAML config. You can also add your own formats to config and use them with source params.
All trascoding formats has [built-in templates](https://github.com/AlexxIT/go2rtc/blob/master/cmd/ffmpeg/ffmpeg.go): `h264`, `h264/ultra`, `h264/high`, `h265`, `opus`, `pcmu`, `pcmu/16000`, `pcmu/48000`, `pcma`, `pcma/16000`, `pcma/48000`, `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.
```yaml
ffmpeg:
bin: ffmpeg # path to ffmpeg binary
link: -hide_banner -i {input} # if input is link
file: -hide_banner -re -stream_loop -1 -i {input} # if input not link
rtsp: -hide_banner -fflags nobuffer -flags low_delay -rtsp_transport tcp -i {input} # if input is RTSP link
output: -rtsp_transport tcp -f rtsp {output} # output
h264: "-codec:v libx264 -g 30 -preset superfast -tune zerolatency -profile main -level 4.1"
h264/ultra: "-codec:v libx264 -g 30 -preset ultrafast -tune zerolatency"
h264/high: "-codec:v libx264 -g 30 -preset superfast -tune zerolatency"
h265: "-codec:v libx265 -g 30 -preset ultrafast -tune zerolatency"
opus: "-codec:a libopus -ar 48000 -ac 2"
pcmu: "-codec:a pcm_mulaw -ar 8000 -ac 1"
pcmu/16000: "-codec:a pcm_mulaw -ar 16000 -ac 1"
pcmu/48000: "-codec:a pcm_mulaw -ar 48000 -ac 1"
pcma: "-codec:a pcm_alaw -ar 8000 -ac 1"
pcma/16000: "-codec:a pcm_alaw -ar 16000 -ac 1"
pcma/48000: "-codec:a pcm_alaw -ar 48000 -ac 1"
aac/16000: "-codec:a aac -ar 16000 -ac 1"
h264: "-codec:v libx264 -g 30 -preset superfast -tune zerolatency -profile main -level 4.1"
mycodec: "-any args that support ffmpeg..."
```
#### Source: Exec
@@ -217,7 +221,7 @@ FFmpeg source just a shortcut to exec source. You can get any stream or file or
```yaml
streams:
stream1: exec:ffmpeg -hide_banner -re -stream_loop -1 -i ~/media/BigBuckBunny.mp4 -c copy -rtsp_transport tcp -f rtsp {output}
stream1: exec:ffmpeg -hide_banner -re -stream_loop -1 -i /media/BigBuckBunny.mp4 -c copy -rtsp_transport tcp -f rtsp {output}
```
#### Source: Hass
@@ -228,7 +232,7 @@ Support import camera links from [Home Assistant](https://www.home-assistant.io/
```yaml
hass:
config: "~/.homeassistant"
config: "/config" # skip this setting if you Hass Add-on user
streams:
generic_camera: hass:Camera1 # Settings > Integrations > Integration Name
@@ -274,9 +278,9 @@ rtsp:
### Module: WebRTC
WebRTC usually works without problems in the local network. But external access may require additional settings. It depends on what type of internet do you have.
WebRTC usually works without problems in the local network. But external access may require additional settings. It depends on what type of Internet do you have.
- by default, WebRTC use two random UDP ports for each connection (for video and audio)
- by default, WebRTC use two random UDP ports for each connection (video and audio)
- you can enable one additional TCP port for all connections and use it for external access
**Static public IP**
@@ -340,6 +344,7 @@ webrtc:
With Ngrok integration you can get external access to your streams in situation when you have Internet with private IP-address.
- Ngrok preistalled 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)
@@ -404,11 +409,10 @@ In other cases you need to use IP-address of server with **go2rtc** application.
- 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/`
2. Add generic camera with RTSP link:
- Hass > Settings > Integrations > Add Integration > [Generic Camera](https://my.home-assistant.io/redirect/config_flow_start/?domain=generic) > `rtsp://...` or `rtmp://...`
- you can use either direct RTSP links to cameras or take RTSP streams from **go2rtc**
3. Use Picture Entity or Picture Glance lovelace card
4. Open full screen card - this is should be WebRTC stream
- you can use either direct RTSP links to cameras or take RTSP streams from **go2rtc**
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.
### Module: Log
@@ -425,3 +429,53 @@ log:
streams: error
webrtc: fatal
```
## Security
By default `go2rtc` start Web interface on port `1984` and RTSP on port `8554`. Both 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:
```yaml
api:
listen: "127.0.0.1:1984" # localhost
rtsp:
listen: "127.0.0.1:8554" # localhost
webrtc:
listen: ":8555" # external TCP port
```
- 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
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.
PS. Additionally WebRTC opens a lot of random UDP ports for transmit encrypted media. They work without problems on the local network. And sometimes work for external access, even if you haven't opened ports on your router. But for stable external WebRTC access, you need to configure the TCP port.
## FAQ
**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.
**Q. Why go2rtc is an addon and not an integration?**
Because **go2rtc** is more than just viewing your stream online with WebRTC. 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.
When **go2rtc** is released, the **WebRTC Camera** integration will be updated. And you can decide whether to use the integration or the addon.
**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**.
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 protocol.
Use any config what you like.
**Q. What about lovelace card with support 2-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).
+7 -5
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@@ -5,16 +5,18 @@ RUN apk add --no-cache git go ffmpeg
ARG BUILD_ARCH
WORKDIR app
RUN git clone https://github.com/AlexxIT/go2rtc .
RUN CGO_ENABLED=0 go build -ldflags "-s -w" -trimpath
RUN git clone https://github.com/AlexxIT/go2rtc \
&& cd go2rtc \
&& CGO_ENABLED=0 go build -ldflags "-s -w" -trimpath -o /usr/local/bin
# https://github.com/home-assistant/docker-base/blob/master/alpine/Dockerfile
RUN if [ "${BUILD_ARCH}" = "aarch64" ]; then BUILD_ARCH="arm64"; \
elif [ "${BUILD_ARCH}" = "armv7" ]; then BUILD_ARCH="arm"; fi \
&& cd go2rtc \
&& curl $(curl -s "https://raw.githubusercontent.com/ngrok/docker-ngrok/main/releases.json" | jq -r ".${BUILD_ARCH}.url") -o ngrok.zip \
&& unzip ngrok
&& unzip ngrok -d /usr/local/bin
RUN rm -r /go2rtc
COPY run.sh /
RUN chmod a+x /run.sh
+7 -6
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@@ -2,12 +2,13 @@
set +e
while true; do
if [ -x /config/go2rtc ]; then
/config/go2rtc -config /config/go2rtc.yaml
else
/app/go2rtc -config /config/go2rtc.yaml
fi
# set cwd for go2rtc (for config file, Hass itegration, etc)
cd /config
# add the feature to override go2rtc binary from Hass config folder
export PATH="/config:$PATH"
while true; do
go2rtc
sleep 5
done
+2 -11
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@@ -9,8 +9,6 @@ import (
"github.com/rs/zerolog"
"net"
"net/http"
"os"
"strconv"
)
func Init() {
@@ -39,15 +37,14 @@ func Init() {
HandleFunc("/api/frame.mp4", frameHandler)
HandleFunc("/api/frame.raw", frameHandler)
HandleFunc("/api/stack", stackHandler)
HandleFunc("/api/streams", streamsHandler)
HandleFunc("/api/exit", exitHandler)
HandleFunc("/api/ws", apiWS)
// ensure we can listen without errors
listener, err := net.Listen("tcp", cfg.Mod.Listen)
if err != nil {
log.Fatal().Err(err).Msg("[api] listen")
return
}
log.Info().Str("addr", cfg.Mod.Listen).Msg("[api] listen")
@@ -55,7 +52,7 @@ func Init() {
go func() {
s := http.Server{}
if err = s.Serve(listener); err != nil {
log.Fatal().Err(err).Msg("[api] Serve")
log.Fatal().Err(err).Msg("[api] serve")
}
}()
}
@@ -99,12 +96,6 @@ func streamsHandler(w http.ResponseWriter, r *http.Request) {
}
}
func exitHandler(w http.ResponseWriter, r *http.Request) {
s := r.URL.Query().Get("code")
code, _ := strconv.Atoi(s)
os.Exit(code)
}
func apiWS(w http.ResponseWriter, r *http.Request) {
ctx := new(Context)
if err := ctx.Upgrade(w, r); err != nil {
+2 -2
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@@ -8,8 +8,8 @@ import (
)
func frameHandler(w http.ResponseWriter, r *http.Request) {
url := r.URL.Query().Get("url")
stream := streams.Get(url)
src := r.URL.Query().Get("src")
stream := streams.Get(src)
if stream == nil {
return
}
+8 -2
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@@ -24,7 +24,11 @@ func Init() {
Mod map[string]string `yaml:"log"`
}
LoadConfig(&cfg)
if data != nil {
if err := yaml.Unmarshal(data, &cfg); err != nil {
println("ERROR: " + err.Error())
}
}
var writer io.Writer = os.Stdout
@@ -48,7 +52,9 @@ func Init() {
modules = cfg.Mod
log.Info().Msgf("go2rtc %s/%s", runtime.GOOS, runtime.GOARCH)
path, _ := os.Getwd()
log.Debug().Str("os", runtime.GOOS).Str("arch", runtime.GOARCH).
Str("cwd", path).Int("conf_size", len(data)).Msgf("[app]")
}
func LoadConfig(v interface{}) {
+27
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@@ -0,0 +1,27 @@
package debug
import (
"github.com/AlexxIT/go2rtc/cmd/api"
"github.com/AlexxIT/go2rtc/cmd/streams"
"github.com/AlexxIT/go2rtc/pkg/streamer"
"net/http"
"os"
"strconv"
)
func Init() {
api.HandleFunc("/api/stack", stackHandler)
api.HandleFunc("/api/exit", exitHandler)
streams.HandleFunc("null", nullHandler)
}
func exitHandler(_ http.ResponseWriter, r *http.Request) {
s := r.URL.Query().Get("code")
code, _ := strconv.Atoi(s)
os.Exit(code)
}
func nullHandler(string) (streamer.Producer, error) {
return nil, nil
}
+1 -1
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@@ -1,4 +1,4 @@
package api
package debug
import (
"bytes"
+37 -1
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@@ -49,7 +49,7 @@ func Handle(url string) (streamer.Producer, error) {
)
// remove `exec:`
args := strings.Split(url[5:], " ")
args := QuoteSplit(url[5:])
cmd := exec.Command(args[0], args[1:]...)
if log.Trace().Enabled() {
@@ -83,3 +83,39 @@ func Handle(url string) (streamer.Producer, error) {
var log zerolog.Logger
var waiters map[string]chan streamer.Producer
func QuoteSplit(s string) []string {
var a []string
for len(s) > 0 {
is := strings.IndexByte(s, ' ')
if is >= 0 {
// skip prefix and double spaces
if is == 0 {
// goto next symbol
s = s[1:]
continue
}
// check if quote in word
if i := strings.IndexByte(s[:is], '"'); i >= 0 {
// search quote end
if is = strings.Index(s, `" `); is > 0 {
is += 1
} else {
is = -1
}
}
}
if is >= 0 {
a = append(a, strings.ReplaceAll(s[:is], `"`, ""))
s = s[is+1:]
} else {
//add last word
a = append(a, s)
break
}
}
return a
}
+36 -1
View File
@@ -1,6 +1,41 @@
## Devices Windows
```
>ffmpeg -hide_banner -f dshow -list_options true -i video="VMware Virtual USB Video Device"
[dshow @ 0000025695e52900] DirectShow video device options (from video devices)
[dshow @ 0000025695e52900] Pin "Record" (alternative pin name "0")
[dshow @ 0000025695e52900] pixel_format=yuyv422 min s=1280x720 fps=1 max s=1280x720 fps=10
[dshow @ 0000025695e52900] pixel_format=yuyv422 min s=1280x720 fps=1 max s=1280x720 fps=10 (tv, bt470bg/bt709/unknown, topleft)
[dshow @ 0000025695e52900] pixel_format=nv12 min s=1280x720 fps=1 max s=1280x720 fps=23
[dshow @ 0000025695e52900] pixel_format=nv12 min s=1280x720 fps=1 max s=1280x720 fps=23 (tv, bt470bg/bt709/unknown, topleft)
```
## Devices Mac
```
% ./ffmpeg -hide_banner -f avfoundation -list_devices true -i ""
[AVFoundation indev @ 0x7f8b1f504d80] AVFoundation video devices:
[AVFoundation indev @ 0x7f8b1f504d80] [0] FaceTime HD Camera
[AVFoundation indev @ 0x7f8b1f504d80] [1] Capture screen 0
[AVFoundation indev @ 0x7f8b1f504d80] AVFoundation audio devices:
[AVFoundation indev @ 0x7f8b1f504d80] [0] Soundflower (2ch)
[AVFoundation indev @ 0x7f8b1f504d80] [1] Built-in Microphone
[AVFoundation indev @ 0x7f8b1f504d80] [2] Soundflower (64ch)
```
## Devices Linux
```
# ffmpeg -hide_banner -f v4l2 -list_formats all -i /dev/video0
[video4linux2,v4l2 @ 0x7f7de7c58bc0] Raw : yuyv422 : YUYV 4:2:2 : 640x480 160x120 176x144 320x176 320x240 352x288 432x240 544x288 640x360 752x416 800x448 800x600 864x480 960x544 960x720 1024x576 1184x656 1280x720 1280x960
[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
```
## Useful links
- https://superuser.com/questions/564402/explanation-of-x264-tune
- https://stackoverflow.com/questions/33624016/why-sliced-thread-affect-so-much-on-realtime-encoding-using-ffmpeg-x264
- https://codec.fandom.com/ru/wiki/X264_-_%D0%BE%D0%BF%D0%B8%D1%81%D0%B0%D0%BD%D0%B8%D0%B5_%D0%BA%D0%BB%D1%8E%D1%87%D0%B5%D0%B9_%D0%BA%D0%BE%D0%B4%D0%B8%D1%80%D0%BE%D0%B2%D0%B0%D0%BD%D0%B8%D1%8F
- https://codec.fandom.com/ru/wiki/X264_-_описание_ключей_кодирования
- https://html5test.com/
- https://trac.ffmpeg.org/wiki/Capture/Webcam
- https://trac.ffmpeg.org/wiki/DirectShow
+63
View File
@@ -0,0 +1,63 @@
package ffmpeg
import (
"bytes"
"github.com/AlexxIT/go2rtc/pkg/streamer"
"os/exec"
"strings"
)
// https://trac.ffmpeg.org/wiki/Capture/Webcam
const deviceInputPrefix = "-f avfoundation"
func deviceInputSuffix(videoIdx, audioIdx int) string {
video := findMedia(streamer.KindVideo, videoIdx)
audio := findMedia(streamer.KindAudio, audioIdx)
switch {
case video != nil && audio != nil:
return `"` + video.Title + `:` + audio.Title + `"`
case video != nil:
return `"` + video.Title + `"`
case audio != nil:
return `"` + audio.Title + `"`
}
return ""
}
func loadMedias() {
cmd := exec.Command(
tpl["bin"], "-hide_banner", "-list_devices", "true", "-f", "avfoundation", "-i", "dummy",
)
var buf bytes.Buffer
cmd.Stderr = &buf
_ = cmd.Run()
var kind string
lines := strings.Split(buf.String(), "\n")
process:
for _, line := range lines {
switch {
case strings.HasSuffix(line, "video devices:"):
kind = streamer.KindVideo
continue
case strings.HasSuffix(line, "audio devices:"):
kind = streamer.KindAudio
continue
case strings.HasPrefix(line, "dummy"):
break process
}
// [AVFoundation indev @ 0x7fad54604380] [0] FaceTime HD Camera
name := line[42:]
media := loadMedia(kind, name)
medias = append(medias, media)
}
}
func loadMedia(kind, name string) *streamer.Media {
return &streamer.Media{
Kind: kind, Title: name,
}
}
+36
View File
@@ -0,0 +1,36 @@
package ffmpeg
import (
"github.com/AlexxIT/go2rtc/pkg/streamer"
"github.com/rs/zerolog/log"
"io/ioutil"
"strings"
)
// https://trac.ffmpeg.org/wiki/Capture/Webcam
const deviceInputPrefix = "-f v4l2"
func deviceInputSuffix(videoIdx, audioIdx int) string {
video := findMedia(streamer.KindVideo, videoIdx)
return video.Title
}
func loadMedias() {
files, err := ioutil.ReadDir("/dev")
if err != nil {
return
}
for _, file := range files {
log.Trace().Msg("[ffmpeg] " + file.Name())
if strings.HasPrefix(file.Name(), streamer.KindVideo) {
media := loadMedia(streamer.KindVideo, "/dev/"+file.Name())
medias = append(medias, media)
}
}
}
func loadMedia(kind, name string) *streamer.Media {
return &streamer.Media{
Kind: kind, Title: name,
}
}
+59
View File
@@ -0,0 +1,59 @@
package ffmpeg
import (
"bytes"
"github.com/AlexxIT/go2rtc/pkg/streamer"
"os/exec"
"strings"
)
// https://trac.ffmpeg.org/wiki/DirectShow
const deviceInputPrefix = "-f dshow"
func deviceInputSuffix(videoIdx, audioIdx int) string {
video := findMedia(streamer.KindVideo, videoIdx)
audio := findMedia(streamer.KindAudio, audioIdx)
switch {
case video != nil && audio != nil:
return `video="` + video.Title + `":audio=` + audio.Title + `"`
case video != nil:
return `video="` + video.Title + `"`
case audio != nil:
return `audio="` + audio.Title + `"`
}
return ""
}
func loadMedias() {
cmd := exec.Command(
tpl["bin"], "-hide_banner", "-list_devices", "true", "-f", "dshow", "-i", "",
)
var buf bytes.Buffer
cmd.Stderr = &buf
_ = cmd.Run()
lines := strings.Split(buf.String(), "\r\n")
for _, line := range lines {
var kind string
if strings.HasSuffix(line, "(video)") {
kind = streamer.KindVideo
} else if strings.HasSuffix(line, "(audio)") {
kind = streamer.KindAudio
} else {
continue
}
// hope we have constant prefix and suffix sizes
// [dshow @ 00000181e8d028c0] "VMware Virtual USB Video Device" (video)
name := line[28 : len(line)-9]
media := loadMedia(kind, name)
medias = append(medias, media)
}
}
func loadMedia(kind, name string) *streamer.Media {
return &streamer.Media{
Kind: kind, Title: name,
}
}
+73
View File
@@ -0,0 +1,73 @@
package ffmpeg
import (
"encoding/json"
"github.com/AlexxIT/go2rtc/pkg/streamer"
"github.com/rs/zerolog/log"
"net/http"
"net/url"
"strconv"
"strings"
)
func getDevice(src string) (string, error) {
if medias == nil {
loadMedias()
}
input := deviceInputPrefix
var videoIdx, audioIdx int
if i := strings.IndexByte(src, '?'); i > 0 {
query, err := url.ParseQuery(src[i+1:])
if err != nil {
return "", err
}
for key, value := range query {
switch key {
case "video":
videoIdx, _ = strconv.Atoi(value[0])
case "audio":
audioIdx, _ = strconv.Atoi(value[0])
case "framerate":
input += " -framerate " + value[0]
case "resolution":
input += " -video_size " + value[0]
}
}
}
input += " -i " + deviceInputSuffix(videoIdx, audioIdx)
return input, nil
}
var medias []*streamer.Media
func findMedia(kind string, index int) *streamer.Media {
for _, media := range medias {
if media.Kind != kind {
continue
}
if index == 0 {
return media
}
index--
}
return nil
}
func handleDevices(w http.ResponseWriter, r *http.Request) {
if medias == nil {
loadMedias()
}
data, err := json.Marshal(medias)
if err != nil {
log.Error().Err(err).Msg("[api.ffmpeg]")
return
}
if _, err = w.Write(data); err != nil {
log.Error().Err(err).Msg("[api.ffmpeg]")
}
}
+14 -6
View File
@@ -1,6 +1,7 @@
package ffmpeg
import (
"github.com/AlexxIT/go2rtc/cmd/api"
"github.com/AlexxIT/go2rtc/cmd/app"
"github.com/AlexxIT/go2rtc/cmd/exec"
"github.com/AlexxIT/go2rtc/cmd/streams"
@@ -20,9 +21,9 @@ func Init() {
"bin": "ffmpeg",
// inputs
"link": "-hide_banner -i {input}",
"rtsp": "-hide_banner -fflags nobuffer -flags low_delay -rtsp_transport tcp -i {input}",
"file": "-hide_banner -re -stream_loop -1 -i {input}",
"link": "-i {input}",
"rtsp": "-fflags nobuffer -flags low_delay -rtsp_transport tcp -i {input}",
"file": "-re -stream_loop -1 -i {input}",
// output
"out": "-rtsp_transport tcp -f rtsp {output}",
@@ -31,7 +32,8 @@ func Init() {
// `-preset superfast` - we can't use ultrafast because it doesn't support `-profile main -level 4.1`
// `-tune zerolatency` - for minimal latency
// `-profile main -level 4.1` - most used streaming profile
"h264": "-codec:v libx264 -g 30 -preset superfast -tune zerolatency -profile main -level 4.1",
// `-pix_fmt yuv420p` - if input pix format 4:2:2
"h264": "-codec:v libx264 -g 30 -preset superfast -tune zerolatency -profile main -level 4.1 -pix_fmt yuv420p",
"h264/ultra": "-codec:v libx264 -g 30 -preset ultrafast -tune zerolatency",
"h264/high": "-codec:v libx264 -g 30 -preset superfast -tune zerolatency",
"h265": "-codec:v libx265 -g 30 -preset ultrafast -tune zerolatency",
@@ -47,7 +49,7 @@ func Init() {
app.LoadConfig(&cfg)
tpl := cfg.Mod
tpl = cfg.Mod
streams.HandleFunc("ffmpeg", func(s string) (streamer.Producer, error) {
s = s[7:] // remove `ffmpeg:`
@@ -62,13 +64,15 @@ func Init() {
switch {
case strings.HasPrefix(s, "rtsp"):
template = tpl["rtsp"]
case strings.HasPrefix(s, "device"):
template, _ = getDevice(s)
case strings.Contains(s, "://"):
template = tpl["link"]
default:
template = tpl["file"]
}
s = "exec:" + tpl["bin"] + " " +
s = "exec:" + tpl["bin"] + " -hide_banner " +
strings.Replace(template, "{input}", s, 1)
if query != nil {
@@ -109,8 +113,12 @@ func Init() {
return exec.Handle(s)
})
api.HandleFunc("/api/devices", handleDevices)
}
var tpl map[string]string
func parseQuery(s string) map[string][]string {
query := map[string][]string{}
for _, key := range strings.Split(s, "#") {
+28 -11
View File
@@ -41,28 +41,45 @@ func Init() {
return
}
ent := new(entries)
if err = json.Unmarshal(data, ent); err != nil {
storage := new(entries)
if err = json.Unmarshal(data, storage); err != nil {
return
}
urls := map[string]string{}
for _, entrie := range ent.Data.Entries {
switch entrie.Domain {
case "generic":
if entrie.Options.StreamSource != "" {
urls[entrie.Title] = entrie.Options.StreamSource
}
}
}
streams.HandleFunc("hass", func(url string) (streamer.Producer, error) {
if hurl := urls[url[5:]]; hurl != "" {
return streams.GetProducer(hurl)
}
return nil, fmt.Errorf("can't get url: %s", url)
})
for _, entrie := range storage.Data.Entries {
switch entrie.Domain {
case "generic":
if entrie.Options.StreamSource == "" {
continue
}
urls[entrie.Title] = entrie.Options.StreamSource
//case "homekit_controller":
// if entrie.Data.ClientID == "" {
// continue
// }
// urls[entrie.Title] = fmt.Sprintf(
// "homekit://%s:%d?client_id=%s&client_private=%s%s&device_id=%s&device_public=%s",
// entrie.Data.DeviceHost, entrie.Data.DevicePort,
// entrie.Data.ClientID, entrie.Data.ClientPrivate, entrie.Data.ClientPublic,
// entrie.Data.DeviceID, entrie.Data.DevicePublic,
// )
default:
continue
}
streams.Get("hass:" + entrie.Title)
}
}
var log zerolog.Logger
+2 -2
View File
@@ -13,8 +13,8 @@ func Init() {
}
func handler(ctx *api.Context, msg *streamer.Message) {
url := ctx.Request.URL.Query().Get("url")
stream := streams.Get(url)
src := ctx.Request.URL.Query().Get("src")
stream := streams.Get(src)
if stream == nil {
return
}
+10 -1
View File
@@ -65,8 +65,17 @@ func rtspHandler(url string) (streamer.Producer, error) {
if err = conn.Dial(); err != nil {
return nil, err
}
conn.Backchannel = true
if err = conn.Describe(); err != nil {
return nil, err
// second try without backchannel, we need to reconnect
if err = conn.Dial(); err != nil {
return nil, err
}
conn.Backchannel = false
if err = conn.Describe(); err != nil {
return nil, err
}
}
return conn, nil
+3
View File
@@ -19,6 +19,9 @@ func HandleFunc(scheme string, handler Handler) {
func HasProducer(url string) bool {
i := strings.IndexByte(url, ':')
if i <= 0 { // TODO: i < 4 ?
return false
}
return handlers[url[:i]] != nil
}
+22 -3
View File
@@ -2,6 +2,7 @@ package streams
import (
"github.com/AlexxIT/go2rtc/pkg/streamer"
"sync"
)
type state byte
@@ -21,15 +22,19 @@ type Producer struct {
tracks []*streamer.Track
state state
mx sync.Mutex
}
func (p *Producer) GetMedias() []*streamer.Media {
p.mx.Lock()
defer p.mx.Unlock()
if p.state == stateNone {
log.Debug().Str("url", p.url).Msg("[streams] probe producer")
var err error
p.element, err = GetProducer(p.url)
if err != nil {
if err != nil || p.element == nil {
log.Error().Err(err).Str("url", p.url).Msg("[streams] probe producer")
return nil
}
@@ -41,6 +46,9 @@ func (p *Producer) GetMedias() []*streamer.Media {
}
func (p *Producer) GetTrack(media *streamer.Media, codec *streamer.Codec) *streamer.Track {
p.mx.Lock()
defer p.mx.Unlock()
if p.state == stateMedias {
p.state = stateTracks
}
@@ -61,6 +69,9 @@ func (p *Producer) GetTrack(media *streamer.Media, codec *streamer.Codec) *strea
// internals
func (p *Producer) start() {
p.mx.Lock()
defer p.mx.Unlock()
if p.state != stateTracks {
return
}
@@ -72,10 +83,18 @@ func (p *Producer) start() {
}
func (p *Producer) stop() {
p.mx.Lock()
log.Debug().Str("url", p.url).Msg("[streams] stop producer")
_ = p.element.Stop()
p.element = nil
if p.element != nil {
_ = p.element.Stop()
p.element = nil
} else {
log.Warn().Str("url", p.url).Msg("[streams] stop empty producer")
}
p.tracks = nil
p.state = stateNone
p.mx.Unlock()
}
+11 -3
View File
@@ -2,6 +2,7 @@ package streams
import (
"encoding/json"
"errors"
"github.com/AlexxIT/go2rtc/pkg/streamer"
)
@@ -29,6 +30,7 @@ func NewStream(source interface{}) *Stream {
}
case map[string]interface{}:
return NewStream(source["url"])
case nil:
default:
panic("wrong source type")
}
@@ -78,7 +80,7 @@ func (s *Stream) AddConsumer(cons streamer.Consumer) (err error) {
// can't match tracks for consumer
if len(consumer.tracks) == 0 {
return nil
return errors.New("couldn't find the matching tracks")
}
s.consumers = append(s.consumers, consumer)
@@ -117,11 +119,17 @@ func (s *Stream) RemoveConsumer(cons streamer.Consumer) {
}
func (s *Stream) AddProducer(prod streamer.Producer) {
panic("not implemented")
producer := &Producer{element: prod, state: stateTracks}
s.producers = append(s.producers, producer)
}
func (s *Stream) RemoveProducer(prod streamer.Producer) {
panic("not implemented")
for i, producer := range s.producers {
if producer.element == prod {
s.removeProducer(i)
break
}
}
}
func (s *Stream) Active() bool {
+2 -1
View File
@@ -2,6 +2,7 @@ package streams
import (
"github.com/AlexxIT/go2rtc/pkg/fake"
"github.com/AlexxIT/go2rtc/pkg/rtsp"
"github.com/AlexxIT/go2rtc/pkg/streamer"
"github.com/stretchr/testify/assert"
"testing"
@@ -103,7 +104,7 @@ a=control:streamid=0
func TestRouting(t *testing.T) {
prod := &fake.Producer{}
prod.Medias, _ = streamer.UnmarshalRTSPSDP([]byte(dahuaSimple))
prod.Medias, _ = rtsp.UnmarshalSDP([]byte(dahuaSimple))
assert.Len(t, prod.Medias, 3)
HandleFunc("fake", func(url string) (streamer.Producer, error) {
+5 -3
View File
@@ -63,13 +63,13 @@ var log zerolog.Logger
var NewPConn func() (*pion.PeerConnection, error)
func offerHandler(ctx *api.Context, msg *streamer.Message) {
name := ctx.Request.URL.Query().Get("url")
stream := streams.Get(name)
src := ctx.Request.URL.Query().Get("src")
stream := streams.Get(src)
if stream == nil {
return
}
log.Debug().Str("stream", name).Msg("[webrtc] new consumer")
log.Debug().Str("src", src).Msg("[webrtc] new consumer")
var err error
@@ -108,6 +108,7 @@ func offerHandler(ctx *api.Context, msg *streamer.Message) {
// 2. AddConsumer, so we get new tracks
if err = stream.AddConsumer(conn); err != nil {
log.Warn().Err(err).Msg("[api.webrtc] add consumer")
_ = conn.Conn.Close()
ctx.Error(err)
return
}
@@ -168,6 +169,7 @@ func ExchangeSDP(
// 2. AddConsumer, so we get new tracks
if err = stream.AddConsumer(conn); err != nil {
log.Warn().Err(err).Msg("[api.webrtc] add consumer")
_ = conn.Conn.Close()
return
}
+10 -1
View File
@@ -38,4 +38,13 @@ require (
golang.org/x/sys v0.0.0-20220622161953-175b2fd9d664 // indirect
)
replace github.com/deepch/vdk v0.0.19 => github.com/AlexxIT/vdk v0.0.18-0.20220616041030-b0d122807b2e
replace (
// windows support: https://github.com/brutella/dnssd/pull/35
github.com/brutella/dnssd v1.2.2 => github.com/rblenkinsopp/dnssd v1.2.3-0.20220516082132-0923f3c787a1
// RTP tlv8 fix
github.com/brutella/hap v0.0.17 => github.com/AlexxIT/hap v0.0.15-0.20220823033740-ce7d1564e657
// MSE update
github.com/deepch/vdk v0.0.19 => github.com/AlexxIT/vdk v0.0.18-0.20220616041030-b0d122807b2e
// AES_256_CM_HMAC_SHA1_80 support
github.com/pion/srtp/v2 v2.0.10 => github.com/AlexxIT/srtp/v2 v2.0.10-0.20220608200505-3191d4f19c10
)
+2 -2
View File
@@ -1,3 +1,5 @@
github.com/AlexxIT/srtp/v2 v2.0.10-0.20220608200505-3191d4f19c10 h1:4aKRthhmkYcStKuk1hcyvkeNJ/BDx5BTIvYmDO9ZJvg=
github.com/AlexxIT/srtp/v2 v2.0.10-0.20220608200505-3191d4f19c10/go.mod h1:5TtM9yw6lsH0ppNCehB/EjEUli7VkUgKSPJqWVqbhQ4=
github.com/AlexxIT/vdk v0.0.18-0.20220616041030-b0d122807b2e h1:NAgHHZB+JUN3/J4/yq1q1EAc8xwJ8bb/Qp0AcjkfzAA=
github.com/AlexxIT/vdk v0.0.18-0.20220616041030-b0d122807b2e/go.mod h1:KqQ/KU3hOc4a62l/jPRH5Hiz5fhTq5cGCl8IqeCxWQI=
github.com/cheekybits/genny v1.0.0/go.mod h1:+tQajlRqAUrPI7DOSpB0XAqZYtQakVtB7wXkRAgjxjQ=
@@ -90,8 +92,6 @@ github.com/pion/sdp/v3 v3.0.5/go.mod h1:iiFWFpQO8Fy3S5ldclBkpXqmWy02ns78NOKoLLL0
github.com/pion/srtp v1.5.1 h1:9Q3jAfslYZBt+C69SI/ZcONJh9049JUHZWYRRf5KEKw=
github.com/pion/srtp v1.5.1/go.mod h1:B+QgX5xPeQTNc1CJStJPHzOlHK66ViMDWTT0HZTCkcA=
github.com/pion/srtp/v2 v2.0.9/go.mod h1:5TtM9yw6lsH0ppNCehB/EjEUli7VkUgKSPJqWVqbhQ4=
github.com/pion/srtp/v2 v2.0.10 h1:b8ZvEuI+mrL8hbr/f1YiJFB34UMrOac3R3N1yq2UN0w=
github.com/pion/srtp/v2 v2.0.10/go.mod h1:XEeSWaK9PfuMs7zxXyiN252AHPbH12NX5q/CFDWtUuA=
github.com/pion/stun v0.3.5 h1:uLUCBCkQby4S1cf6CGuR9QrVOKcvUwFeemaC865QHDg=
github.com/pion/stun v0.3.5/go.mod h1:gDMim+47EeEtfWogA37n6qXZS88L5V6LqFcf+DZA2UA=
github.com/pion/transport v0.6.0/go.mod h1:iWZ07doqOosSLMhZ+FXUTq+TamDoXSllxpbGcfkCmbE=
+2
View File
@@ -3,6 +3,7 @@ package main
import (
"github.com/AlexxIT/go2rtc/cmd/api"
"github.com/AlexxIT/go2rtc/cmd/app"
"github.com/AlexxIT/go2rtc/cmd/debug"
"github.com/AlexxIT/go2rtc/cmd/exec"
"github.com/AlexxIT/go2rtc/cmd/ffmpeg"
"github.com/AlexxIT/go2rtc/cmd/hass"
@@ -33,6 +34,7 @@ func main() {
mse.Init()
ngrok.Init()
debug.Init()
sigs := make(chan os.Signal, 1)
signal.Notify(sigs, syscall.SIGINT, syscall.SIGTERM)
+32 -2
View File
@@ -3,9 +3,12 @@ package rtmp
import (
"encoding/base64"
"encoding/binary"
"encoding/hex"
"fmt"
"github.com/AlexxIT/go2rtc/pkg/h264"
"github.com/AlexxIT/go2rtc/pkg/streamer"
"github.com/deepch/vdk/av"
"github.com/deepch/vdk/codec/aacparser"
"github.com/deepch/vdk/codec/h264parser"
"github.com/deepch/vdk/format/rtmp"
"github.com/pion/rtp"
@@ -70,9 +73,36 @@ func (c *Client) Dial() (err error) {
c.tracks = append(c.tracks, track)
case av.AAC:
panic("not implemented")
// TODO: fix support
cd := stream.(aacparser.CodecData)
// a=fmtp:97 streamtype=5;profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3;config=1588
fmtp := fmt.Sprintf(
"config=%s",
hex.EncodeToString(cd.ConfigBytes),
)
codec := &streamer.Codec{
Name: streamer.CodecAAC,
ClockRate: uint32(cd.Config.SampleRate),
Channels: uint16(cd.Config.ChannelConfig),
FmtpLine: fmtp,
}
media := &streamer.Media{
Kind: streamer.KindAudio,
Direction: streamer.DirectionSendonly,
Codecs: []*streamer.Codec{codec},
}
c.medias = append(c.medias, media)
track := &streamer.Track{
Codec: codec, Direction: media.Direction,
}
c.tracks = append(c.tracks, track)
default:
panic("unsupported codec")
fmt.Printf("[rtmp] unsupported codec %+v\n", stream)
}
}
+2 -1
View File
@@ -2,6 +2,7 @@ package rtmp
import (
"encoding/json"
"fmt"
"github.com/AlexxIT/go2rtc/pkg/streamer"
"strconv"
)
@@ -16,7 +17,7 @@ func (c *Client) GetTrack(media *streamer.Media, codec *streamer.Codec) *streame
return track
}
}
panic("wrong codec")
panic(fmt.Sprintf("wrong media/codec: %+v %+v", media, codec))
}
func (c *Client) Start() error {
+69 -36
View File
@@ -43,11 +43,15 @@ const (
ModeServerConsumer
)
const KeepAlive = time.Second * 25
type Conn struct {
streamer.Element
// public
Backchannel bool
Medias []*streamer.Media
Session string
UserAgent string
@@ -104,6 +108,9 @@ func (c *Conn) Dial() (err error) {
//if c.state != StateClientInit {
// panic("wrong state")
//}
if c.conn != nil && c.auth != nil {
c.auth.Reset()
}
c.conn, err = net.DialTimeout(
"tcp", c.URL.Host, 10*time.Second,
@@ -144,7 +151,9 @@ func (c *Conn) Request(req *tcp.Request) error {
}
c.sequence++
req.Header.Set("CSeq", strconv.Itoa(c.sequence))
// important to send case sensitive CSeq
// https://github.com/AlexxIT/go2rtc/issues/7
req.Header["CSeq"] = []string{strconv.Itoa(c.sequence)}
c.auth.Write(req)
@@ -254,21 +263,17 @@ func (c *Conn) Describe() error {
Method: MethodDescribe,
URL: c.URL,
Header: map[string][]string{
"Accept": {"application/sdp"},
"Require": {"www.onvif.org/ver20/backchannel"},
"Accept": {"application/sdp"},
},
}
if c.Backchannel {
req.Header.Set("Require", "www.onvif.org/ver20/backchannel")
}
res, err := c.Do(req)
if err != nil {
if res != nil {
// if we have answer - give second chanse without onvif header
req.Header.Del("Require")
res, err = c.Do(req)
}
if err != nil {
return err
}
return err
}
if val := res.Header.Get("Content-Base"); val != "" {
@@ -278,13 +283,7 @@ func (c *Conn) Describe() error {
}
}
// fix bug in Sonoff camera SDP "o=- 1 1 IN IP4 rom t_rtsplin"
// TODO: make some universal fix
if i := bytes.Index(res.Body, []byte("rom t_rtsplin")); i > 0 {
res.Body[i+3] = '_'
}
c.Medias, err = streamer.UnmarshalRTSPSDP(res.Body)
c.Medias, err = UnmarshalSDP(res.Body)
if err != nil {
return err
}
@@ -370,9 +369,10 @@ func (c *Conn) SetupMedia(
// Transport: RTP/AVP/TCP;unicast;interleaved=10-11;ssrc=10117CB7
// Transport: RTP/AVP/TCP;unicast;destination=192.168.1.123;source=192.168.10.12;interleaved=0
// Transport: RTP/AVP/TCP;ssrc=22345682;interleaved=0-1
s := res.Header.Get("Transport")
// TODO: rewrite
if !strings.HasPrefix(s, "RTP/AVP/TCP;unicast") {
if !strings.HasPrefix(s, "RTP/AVP/TCP;") {
return nil, fmt.Errorf("wrong transport: %s", s)
}
@@ -451,15 +451,17 @@ func (c *Conn) Accept() error {
return err
}
if c.URL == nil {
c.URL = req.URL
c.UserAgent = req.Header.Get("User-Agent")
}
c.Fire(req)
// Receiver: OPTIONS > DESCRIBE > SETUP... > PLAY > TEARDOWN
// Sender: OPTIONS > ANNOUNCE > SETUP... > RECORD > TEARDOWN
switch req.Method {
case MethodOptions:
c.URL = req.URL
c.UserAgent = req.Header.Get("User-Agent")
res := &tcp.Response{
Header: map[string][]string{
"Public": {"OPTIONS, SETUP, TEARDOWN, DESCRIBE, PLAY, PAUSE, ANNOUNCE, RECORD"},
@@ -475,7 +477,7 @@ func (c *Conn) Accept() error {
return errors.New("wrong content type")
}
c.Medias, err = streamer.UnmarshalRTSPSDP(req.Body)
c.Medias, err = UnmarshalSDP(req.Body)
if err != nil {
return err
}
@@ -575,6 +577,7 @@ func (c *Conn) Handle() (err error) {
}()
//c.Fire(streamer.StatePlaying)
ts := time.Now().Add(KeepAlive)
for {
// we can read:
@@ -629,7 +632,7 @@ func (c *Conn) Handle() (err error) {
if channelID&1 == 0 {
packet := &rtp.Packet{}
if err = packet.Unmarshal(buf); err != nil {
return errors.New("wrong RTP data")
return
}
track := c.channels[channelID]
@@ -637,22 +640,34 @@ func (c *Conn) Handle() (err error) {
_ = track.WriteRTP(packet)
//return fmt.Errorf("wrong channelID: %d", channelID)
} else {
panic("wrong channelID")
continue // TODO: maybe fix this
//panic("wrong channelID")
}
} else {
msg := &RTCP{Channel: channelID}
if err = msg.Header.Unmarshal(buf); err != nil {
return errors.New("wrong RTCP data")
return
}
msg.Packets, err = rtcp.Unmarshal(buf)
if err != nil {
return errors.New("wrong RTCP data")
return
}
c.Fire(msg)
}
// keep-alive
now := time.Now()
if now.After(ts) {
req := &tcp.Request{Method: MethodOptions, URL: c.URL}
// don't need to wait respose on this request
if err = c.Request(req); err != nil {
return err
}
ts = now.Add(KeepAlive)
}
}
}
@@ -712,17 +727,35 @@ type RTCP struct {
Packets []rtcp.Packet
}
func between(s, sub1, sub2 string) (res string, ok1 bool, ok2 bool) {
i := strings.Index(s, sub1)
if i >= 0 {
ok1 = true
s = s[i+len(sub1):]
const sdpHeader = `v=0
o=- 0 0 IN IP4 0.0.0.0
s=-
t=0 0`
func UnmarshalSDP(rawSDP []byte) ([]*streamer.Media, error) {
medias, err := streamer.UnmarshalSDP(rawSDP)
if err != nil {
// fix SDP header for some cameras
i := bytes.Index(rawSDP, []byte("\nm="))
if i > 0 {
rawSDP = append([]byte(sdpHeader), rawSDP[i:]...)
medias, err = streamer.UnmarshalSDP(rawSDP)
}
if err != nil {
return nil, err
}
}
i = strings.Index(s, sub2)
if i >= 0 {
return s[:i], ok1, true
// fix bug in ONVIF spec
// https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec-v241.pdf
for _, media := range medias {
switch media.Direction {
case streamer.DirectionRecvonly, "":
media.Direction = streamer.DirectionSendonly
case streamer.DirectionSendonly:
media.Direction = streamer.DirectionRecvonly
}
}
return s, ok1, false
return medias, nil
}
+6 -25
View File
@@ -46,12 +46,13 @@ func GetKind(name string) string {
// - deepch/vdk/format/rtsp/sdp.Media
// - pion/sdp.MediaDescription
type Media struct {
Kind string // video, audio
Direction string
Codecs []*Codec
Kind string `json:"kind,omitempty"` // video or audio
Direction string `json:"direction,omitempty"`
Codecs []*Codec `json:"codecs,omitempty"`
MID string // TODO: fixme?
Control string // TODO: fixme?
MID string `json:"mid,omitempty"` // TODO: fixme?
Control string `json:"control,omitempty"` // TODO: fixme?
Title string `json:"title,omitempty"` // TODO: fixme?
}
func (m *Media) String() string {
@@ -180,26 +181,6 @@ func UnmarshalSDP(rawSDP []byte) ([]*Media, error) {
return medias, nil
}
func UnmarshalRTSPSDP(rawSDP []byte) ([]*Media, error) {
medias, err := UnmarshalSDP(rawSDP)
if err != nil {
return nil, err
}
// fix bug in ONVIF spec
// https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec-v241.pdf
for _, media := range medias {
switch media.Direction {
case DirectionRecvonly, "":
media.Direction = DirectionSendonly
case DirectionSendonly:
media.Direction = DirectionRecvonly
}
}
return medias, nil
}
func MarshalSDP(medias []*Media) ([]byte, error) {
sd := &sdp.SessionDescription{}
+6
View File
@@ -80,6 +80,12 @@ func (a *Auth) Write(req *Request) {
}
}
func (a *Auth) Reset() {
if a.Method == AuthDigest {
a.Method = AuthUnknown
}
}
func Between(s, sub1, sub2 string) string {
i := strings.Index(s, sub1)
if i < 0 {
+4 -1
View File
@@ -47,10 +47,13 @@ func ReadResponse(r *bufio.Reader) (*Response, error) {
if err != nil {
return nil, err
}
if line == "" {
return nil, errors.New("empty response on RTSP request")
}
ss := strings.SplitN(line, " ", 3)
if len(ss) != 3 {
return nil, errors.New("malformed response")
return nil, fmt.Errorf("malformed response: %s", line)
}
res := &Response{
+20 -20
View File
@@ -1,6 +1,7 @@
package webrtc
import (
"github.com/pion/ice/v2"
"github.com/pion/interceptor"
"github.com/pion/webrtc/v3"
"net"
@@ -21,31 +22,30 @@ func NewAPI(address string) (*webrtc.API, error) {
return nil, err
}
if address == "" {
return webrtc.NewAPI(
webrtc.WithMediaEngine(m),
webrtc.WithInterceptorRegistry(i),
), nil
}
ln, err := net.Listen("tcp", address)
if err != nil {
return webrtc.NewAPI(
webrtc.WithMediaEngine(m),
webrtc.WithInterceptorRegistry(i),
), err
}
s := webrtc.SettingEngine{
//LoggerFactory: customLoggerFactory{},
}
s.SetNetworkTypes([]webrtc.NetworkType{
webrtc.NetworkTypeUDP4, webrtc.NetworkTypeUDP6,
webrtc.NetworkTypeTCP4, webrtc.NetworkTypeTCP6,
// disable listen on Hassio docker interfaces
s.SetInterfaceFilter(func(name string) bool {
return name != "hassio" && name != "docker0"
})
tcpMux := webrtc.NewICETCPMux(nil, ln, 8)
s.SetICETCPMux(tcpMux)
// disable mDNS listener
s.SetICEMulticastDNSMode(ice.MulticastDNSModeDisabled)
if address != "" {
ln, err := net.Listen("tcp", address)
if err == nil {
s.SetNetworkTypes([]webrtc.NetworkType{
webrtc.NetworkTypeUDP4, webrtc.NetworkTypeUDP6,
webrtc.NetworkTypeTCP4, webrtc.NetworkTypeTCP6,
})
tcpMux := webrtc.NewICETCPMux(nil, ln, 8)
s.SetICETCPMux(tcpMux)
}
}
return webrtc.NewAPI(
webrtc.WithMediaEngine(m),
+7
View File
@@ -2,11 +2,18 @@
- UPX-3.96 pack broken bin for `linux_mipsel`
- UPX-3.95 pack broken bin for `mac_amd64`
- UPX windows pack is recognised by anti-viruses as malicious
- `aarch64` = `arm64`
- `armv7` = `arm`
## Virus
- https://go.dev/doc/faq#virus
- https://groups.google.com/g/golang-nuts/c/lPwiWYaApSU
## Useful links
- https://github.com/golang/go/wiki/GoArm
- https://gist.github.com/asukakenji/f15ba7e588ac42795f421b48b8aede63
- https://en.wikipedia.org/wiki/AArch64
- https://stackoverflow.com/questions/22267189/what-does-the-w-flag-mean-when-passed-in-via-the-ldflags-option-to-the-go-comman
+4 -4
View File
@@ -2,13 +2,13 @@
@SET GOOS=windows
@SET GOARCH=amd64
@SET FILENAME=go2rtc_win64.exe
go build -ldflags "-s -w" -trimpath -o %FILENAME% && upx-3.96 %FILENAME%
@SET FILENAME=go2rtc_win64.zip
go build -ldflags "-s -w" -trimpath && 7z a -sdel %FILENAME% go2rtc.exe
@SET GOOS=windows
@SET GOARCH=386
@SET FILENAME=go2rtc_win32.exe
go build -ldflags "-s -w" -trimpath -o %FILENAME% && upx-3.96 %FILENAME%
@SET FILENAME=go2rtc_win32.zip
go build -ldflags "-s -w" -trimpath && 7z a -sdel %FILENAME% go2rtc.exe
@SET GOOS=linux
@SET GOARCH=amd64
+3 -2
View File
@@ -1,4 +1,5 @@
@SET GOOS=linux
@SET GOARCH=amd64
cd ..
go build -ldflags "-s -w" -trimpath && upx-3.96 go2rtc
@cd ..
del go2rtc
go build -ldflags "-s -w" -trimpath
+5
View File
@@ -0,0 +1,5 @@
@ECHO OFF
@SET GOOS=linux
@SET GOARCH=mipsle
cd ..
go build -ldflags "-s -w" -trimpath && upx-3.95 go2rtc
+5
View File
@@ -0,0 +1,5 @@
@SET GOOS=darwin
@SET GOARCH=amd64
@cd ..
del go2rtc
go build -ldflags "-s -w" -trimpath
+4
View File
@@ -0,0 +1,4 @@
@SET GOOS=windows
@SET GOARCH=amd64
cd ..
go build -ldflags "-w -s" -trimpath
+82
View File
@@ -0,0 +1,82 @@
<!DOCTYPE html>
<html>
<head>
<meta charset="utf-8">
<meta name="viewport" content="width=device-width, user-scalable=yes, initial-scale=1, maximum-scale=1">
<meta http-equiv="X-UA-Compatible" content="ie=edge">
<title>go2rtc</title>
<style>
table {
background-color: white;
text-align: left;
border-collapse: collapse;
}
table td, table th {
border: 1px solid black;
padding: 5px 5px;
}
table tbody td {
font-size: 13px;
}
table thead {
background: #CFCFCF;
background: linear-gradient(to bottom, #dbdbdb 0%, #d3d3d3 66%, #CFCFCF 100%);
border-bottom: 3px solid black;
}
table thead th {
font-size: 15px;
font-weight: bold;
color: black;
text-align: center;
}
.header {
padding: 5px 5px;
}
</style>
</head>
<body>
<div class="header">
<input id="src" type="text" placeholder="url">
<a id="add" href="#">add</a>
</div>
<table id="devices">
<thead>
<tr>
<th>Kind</th>
<th>Name</th>
</tr>
</thead>
<tbody>
</tbody>
</table>
<script>
const baseUrl = location.origin + location.pathname.substr(
0, location.pathname.lastIndexOf("/")
);
function reload() {
fetch(`${baseUrl}/api/devices`).then(r => {
r.json().then(data => {
let html = '';
data.forEach(function (item) {
html += `<tr><td>${item.kind}</td><td>${item.title}</td></tr>`;
})
let content = document.getElementById('devices').getElementsByTagName('tbody')[0];
content.innerHTML = html
});
})
}
reload();
</script>
</body>
</html>
+4 -3
View File
@@ -45,6 +45,7 @@
<div class="header">
<input id="src" type="text" placeholder="url">
<a id="add" href="#">add</a>
<a href="devices.html">devices</a>
</div>
<table id="streams">
<thead>
@@ -63,9 +64,9 @@
);
const links = [
'<a href="webrtc.html?url={name}">webrtc</a>',
'<a href="mse.html?url={name}">mse</a>',
'<a href="api/frame.mp4?url={name}">frame.mp4</a>',
'<a href="webrtc.html?src={name}">webrtc</a>',
'<a href="mse.html?src={name}">mse</a>',
'<a href="api/frame.mp4?src={name}">frame.mp4</a>',
'<a href="api/streams?src={name}">info</a>',
];