diff --git a/README.md b/README.md
index 0da4d9af..c91268d2 100644
--- a/README.md
+++ b/README.md
@@ -7,7 +7,7 @@ Ultimate camera streaming application with support RTSP, WebRTC, HomeKit, FFmpeg
- zero-dependency and zero-config [small app](#go2rtc-binary) for all OS (Windows, macOS, Linux, ARM)
- zero-delay for many supported protocols (lowest possible streaming latency)
- streaming from [RTSP](#source-rtsp), [RTMP](#source-rtmp), [HTTP](#source-http) (FLV/MJPEG/JPEG), [FFmpeg](#source-ffmpeg), [USB Cameras](#source-ffmpeg-device) and [other sources](#module-streams)
-- streaming to [RTSP](#module-rtsp), [WebRTC](#module-webrtc), [MSE/MP4](#module-mp4) or [MJPEG](#module-mjpeg)
+- streaming to [RTSP](#module-rtsp), [WebRTC](#module-webrtc), [MSE/MP4](#module-mp4), [HLS](#module-hls) or [MJPEG](#module-mjpeg)
- first project in the World with support streaming from [HomeKit Cameras](#source-homekit)
- first project in the World with 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)
@@ -31,8 +31,9 @@ Ultimate camera streaming application with support RTSP, WebRTC, HomeKit, FFmpeg
* [Fast start](#fast-start)
* [go2rtc: Binary](#go2rtc-binary)
- * [go2rtc: Home Assistant Add-on](#go2rtc-home-assistant-add-on)
* [go2rtc: Docker](#go2rtc-docker)
+ * [go2rtc: Home Assistant Add-on](#go2rtc-home-assistant-add-on)
+ * [go2rtc: Home Assistant Integration](#go2rtc-home-assistant-integration)
* [Configuration](#configuration)
* [Module: Streams](#module-streams)
* [Source: RTSP](#source-rtsp)
@@ -50,20 +51,21 @@ Ultimate camera streaming application with support RTSP, WebRTC, HomeKit, FFmpeg
* [Module: WebRTC](#module-webrtc)
* [Module: Ngrok](#module-ngrok)
* [Module: Hass](#module-hass)
- * [From go2rtc to Hass](#from-go2rtc-to-hass)
- * [From Hass to go2rtc](#from-hass-to-go2rtc)
* [Module: MP4](#module-mp4)
+ * [Module: HLS](#module-hls)
* [Module: MJPEG](#module-mjpeg)
* [Module: Log](#module-log)
* [Security](#security)
+* [Codecs filters](#codecs-filters)
* [Codecs madness](#codecs-madness)
* [Codecs negotiation](#codecs-negotiation)
+* [Projects using go2rtc](#projects-using-go2rtc)
* [TIPS](#tips)
* [FAQ](#faq)
## Fast start
-1. Download [binary](#go2rtc-binary) or use [Docker](#go2rtc-docker) or [Home Assistant Add-on](#go2rtc-home-assistant-add-on)
+1. Download [binary](#go2rtc-binary) or use [Docker](#go2rtc-docker) or Home Assistant [Add-on](#go2rtc-home-assistant-add-on) or [Integration](#go2rtc-home-assistant-integration)
2. Open web interface: `http://localhost:1984/`
**Optionally:**
@@ -92,6 +94,10 @@ Download binary for your OS from [latest release](https://github.com/AlexxIT/go2
Don't forget to fix the rights `chmod +x go2rtc_xxx_xxx` on Linux and Mac.
+### go2rtc: Docker
+
+Container [alexxit/go2rtc](https://hub.docker.com/r/alexxit/go2rtc) with support `amd64`, `386`, `arm64`, `arm`. This container is the same as [Home Assistant Add-on](#go2rtc-home-assistant-add-on), but can be used separately from Home Assistant. Container has preinstalled [FFmpeg](#source-ffmpeg), [Ngrok](#module-ngrok) and [Python](#source-echo).
+
### go2rtc: Home Assistant Add-on
[](https://my.home-assistant.io/redirect/supervisor_addon/?addon=a889bffc_go2rtc&repository_url=https%3A%2F%2Fgithub.com%2FAlexxIT%2Fhassio-addons)
@@ -101,9 +107,9 @@ Don't forget to fix the rights `chmod +x go2rtc_xxx_xxx` on Linux and Mac.
- go2rtc > Install > Start
2. Setup [Integration](#module-hass)
-### go2rtc: Docker
+### go2rtc: Home Assistant Integration
-Container [alexxit/go2rtc](https://hub.docker.com/r/alexxit/go2rtc) with support `amd64`, `386`, `arm64`, `arm`. This container is the same as [Home Assistant Add-on](#go2rtc-home-assistant-add-on), but can be used separately from Home Assistant. Container has preinstalled [FFmpeg](#source-ffmpeg), [Ngrok](#module-ngrok) and [Python](#source-echo).
+[WebRTC Camera](https://github.com/AlexxIT/WebRTC) custom component can be used on any [Home Assistant installation](https://www.home-assistant.io/installation/), including [HassWP](https://github.com/AlexxIT/HassWP) on Windows. It can automatically download and use the latest version of go2rtc. Or it can connect to an existing version of go2rtc. Addon installation in this case is optional.
## Configuration
@@ -122,6 +128,7 @@ Available modules:
- [rtsp](#module-rtsp) - RTSP Server (important for FFmpeg support)
- [webrtc](#module-webrtc) - WebRTC Server
- [mp4](#module-mp4) - MSE, MP4 stream and MP4 shapshot Server
+- [hls](#module-hls) - HLS TS or fMP4 stream Server
- [mjpeg](#module-mjpeg) - MJPEG Server
- [ffmpeg](#source-ffmpeg) - FFmpeg integration
- [ngrok](#module-ngrok) - Ngrok integration (external access for private network)
@@ -408,20 +415,25 @@ api:
You can get any stream as RTSP-stream: `rtsp://192.168.1.123:8554/{stream_name}`
-- you can omit the codec filters, so one first video and one first audio will be selected
-- you can set `?video=copy` or just `?video`, so only one first video without audio will be selected
-- you can set multiple video or audio, so all of them will be selected
-- 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 always disabled for localhost calls (ex. FFmpeg or Hass on same server).
```yaml
rtsp:
listen: ":8554" # RTSP Server TCP port, default - 8554
username: "admin" # optional, default - disabled
password: "pass" # optional, default - disabled
+ default_query: "video&audio" # optional, default codecs filters
```
+By default go2rtc provide RTSP-stream with only one first video and only one first audio. You can change it with the `default_query` setting:
+
+- `default_query: "mp4"` - MP4 compatible codecs (H264, H265, AAC)
+- `default_query: "video=all&audio=all"` - all tracks from all source (not all players can handle this)
+- `default_query: "video=h264,h265"` - only one video track (H264 or H265)
+- `default_query: "video&audio=all"` - only one first any video and all audio as separate tracks
+
+Read more about [codecs filters](#codecs-filters).
+
### 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.
@@ -540,24 +552,29 @@ tunnels:
### Module: Hass
-If you install **go2rtc** as [Hass Add-on](#go2rtc-home-assistant-add-on) - you need to use localhost IP-address. In other cases you need to use IP-address of server with **go2rtc** application.
+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.
-#### From go2rtc to Hass
+But go2rtc is also compatible and can be used with [RTSPtoWebRTC](https://www.home-assistant.io/integrations/rtsp_to_webrtc/) built-in integration.
-Add any supported [stream source](#module-streams) as [Generic Camera](https://www.home-assistant.io/integrations/generic/) and view stream with built-in [Stream](https://www.home-assistant.io/integrations/stream/) integration. Technology `HLS`, supported codecs: `H264`, poor latency.
+You have several options on how to add a camera to Home Assistant:
-1. Add your stream to [go2rtc config](#configuration)
-2. Hass > Settings > Integrations > Add Integration > [Generic Camera](https://my.home-assistant.io/redirect/config_flow_start/?domain=generic) > `rtsp://127.0.0.1:8554/camera1`
+1. Camera RTSP source => [Generic Camera](https://www.home-assistant.io/integrations/generic/)
+2. Camera [any source](#module-streams) => [go2rtc config](#configuration) => [Generic Camera](https://www.home-assistant.io/integrations/generic/)
+ - Install any [go2rtc](#fast-start)
+ - Add your stream to [go2rtc config](#configuration)
+ - Hass > Settings > Integrations > Add Integration > [Generic Camera](https://my.home-assistant.io/redirect/config_flow_start/?domain=generic) > `rtsp://127.0.0.1:8554/camera1` (change to your stream name)
-#### From Hass to go2rtc
+You have several options on how to watch the stream from the cameras in Home Assistant:
-View almost any Hass camera using `WebRTC` technology, supported codecs `H264`/`PCMU`/`PCMA`/`OPUS`, best latency.
-
-When the stream starts - the camera `entity_id` will be added to go2rtc "on the fly". You don't need to add cameras manually to [go2rtc config](#configuration). Some cameras (like [Nest](https://www.home-assistant.io/integrations/nest/)) have a dynamic link to the stream, it will be updated each time a stream is started from the Hass interface.
-
-1. 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. RTSPtoWebRTC > Configure > STUN server: `stun.l.google.com:19302`
-3. Use Picture Entity or Picture Glance lovelace card
+1. `Camera Entity` => `Picture Entity Card` => Technology `HLS`, codecs: `H264/H265/AAC`, poor latency.
+2. `Camera Entity` => [RTSPtoWebRTC](https://www.home-assistant.io/integrations/rtsp_to_webrtc/) => `Picture Entity Card` => Technology `WebRTC`, codecs: `H264/PCMU/PCMA/OPUS`, best latency.
+ - 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
+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
You can add camera `entity_id` to [go2rtc config](#configuration) if you need transcoding:
@@ -574,13 +591,28 @@ 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. MP4 "file stream" - bad format for streaming because of high start delay, doesn't work in Safari
+3. 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.
API examples:
- MP4 stream: `http://192.168.1.123:1984/api/stream.mp4?src=camera1`
- MP4 snapshot: `http://192.168.1.123:1984/api/frame.mp4?src=camera1`
+Read more about [codecs filters](#codecs-filters).
+
+### Module: HLS
+
+[HLS](https://en.wikipedia.org/wiki/HTTP_Live_Streaming) is the worst technology for real-time streaming. It can only be useful on devices that do not support more modern technology, like [WebRTC](#module-webrtc), [MSE/MP4](#module-mp4).
+
+The go2rtc implementation differs from the standards and may not work with all players.
+
+API examples:
+
+- HLS/TS stream: `http://192.168.1.123:1984/api/stream.m3u8?src=camera1` (H264)
+- HLS/fMP4 stream: `http://192.168.1.123:1984/api/stream.m3u8?src=camera1&mp4` (H264, H265, AAC)
+
+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.
@@ -647,21 +679,42 @@ If you need Web interface protection without Home Assistant Add-on - you need to
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.
+## Codecs filters
+
+go2rtc can automatically detect which codecs your device supports for [WebRTC](#module-webrtc) and [MSE](#module-mp4) technologies.
+
+But it cannot be done for [RTSP](#module-rtsp), [stream.mp4](#module-mp4), [HLS](#module-hls) technologies. You can manually add a codec filter when you create a link to a stream. The filters work the same for all three technologies. Filters do not create a new codec. They only select the suitable codec from existing sources. You can add new codecs to the stream using the [FFmpeg transcoding](#source-ffmpeg).
+
+Without filters:
+
+- RTSP will provide only the first video and only the first audio
+- MP4 will include only compatible codecs (H264, H265, AAC)
+- HLS will output in the legacy TS format (H264 without audio)
+
+Some examples:
+
+- `rtsp://192.168.1.123:8554/camera1?mp4` - useful for recording as MP4 files (e.g. Hass or Frigate)
+- `rtsp://192.168.1.123:8554/camera1?video=h264,h265&audio=aac` - full version of the filter above
+- `rtsp://192.168.1.123:8554/camera1?video=h264&audio=aac&audio=opus` - H264 video codec and two separate audio tracks
+- `rtsp://192.168.1.123:8554/camera1?video&audio=all` - any video codec and all audio codecs as separate tracks
+- `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.mp4?src=camera1&video=h264,h265&audio=aac,opus,mp3,pcma,pcmu` - MP4 file with non standard audio codecs, does not work in some players
+
## Codecs madness
-`AVC/H.264` codec 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 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.
-| Device | WebRTC | MSE | MP4 |
-|---------------------|-------------|-------------|-------------|
-| *latency* | best | medium | bad |
-| Desktop Chrome 107+ | H264 | H264, H265* | H264, H265* |
-| Desktop Edge | H264 | H264, H265* | H264, H265* |
-| Desktop Safari | H264, H265* | H264, H265 | **no!** |
-| Desktop Firefox | H264 | H264 | H264 |
-| Android Chrome 107+ | H264 | H264, H265* | H264 |
-| iPad Safari 13+ | H264, H265* | H264, H265 | **no!** |
-| iPhone Safari 13+ | H264, H265* | **no!** | **no!** |
-| masOS Hass App | no | no | no |
+| Device | WebRTC | MSE | stream.mp4 |
+|---------------------|-------------------------------|------------------------|-----------------------------------------|
+| *latency* | best | medium | bad |
+| Desktop Chrome 107+ | H264, OPUS, PCMU, PCMA | H264, H265*, AAC, OPUS | H264, H265*, AAC, OPUS, PCMU, PCMA, MP3 |
+| Desktop Edge | H264, OPUS, PCMU, PCMA | H264, H265*, AAC, OPUS | H264, H265*, AAC, OPUS, PCMU, PCMA, MP3 |
+| Desktop Safari | H264, H265*, OPUS, PCMU, PCMA | H264, H265, AAC | **no!** |
+| Desktop Firefox | H264, OPUS, PCMU, PCMA | H264, AAC, OPUS | H264, AAC, OPUS |
+| Android Chrome 107+ | H264, OPUS, PCMU, PCMA | H264, H265*, AAC, OPUS | H264, ?, AAC, OPUS, PCMU, PCMA, MP3 |
+| iPad Safari 13+ | H264, H265*, OPUS, PCMU, PCMA | H264, H265, AAC | **no!** |
+| iPhone Safari 13+ | H264, H265*, OPUS, PCMU, PCMA | **no!** | **no!** |
+| masOS Hass App | no | no | no |
- 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/)
@@ -670,8 +723,9 @@ PS. Additionally WebRTC will try to use the 8555 UDP port for transmit encrypted
**Audio**
-- WebRTC audio codecs: `PCMU/8000`, `PCMA/8000`, `OPUS/48000/2`
-- MSE/MP4 audio codecs: `AAC`
+- **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)
+- `PCMU` and `PCMA` inside **MP4** isn't a standard, but some players support them, for example Chromium browsers
**Apple devices**
@@ -708,6 +762,12 @@ streams:
**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.
+## Projects using go2rtc
+
+- [Frigate 12+](https://frigate.video/) - open source NVR built around real-time AI object detection
+- [ring-mqtt](https://github.com/tsightler/ring-mqtt) - Ring devices to MQTT Bridge
+- [EufyP2PStream](https://github.com/oischinger/eufyp2pstream) - A small project that provides a Video/Audio Stream from Eufy cameras that don't directly support RTSP
+
## TIPS
**Using apps for low RTSP delay**
@@ -725,17 +785,17 @@ streams:
**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?**
+**Q. Should I use go2rtc addon or WebRTC Camera 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.
+**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.
-When **go2rtc** is released, the **WebRTC Camera** integration will be updated. And you can decide whether to use the integration or the addon.
+Basic users can use **WebRTC Camera** integration. Advanced users can use go2rtc addon or Frigate 12+ 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.
+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.
Use any config what you like.
diff --git a/cmd/app/app.go b/cmd/app/app.go
index 297b3d33..b77aed61 100644
--- a/cmd/app/app.go
+++ b/cmd/app/app.go
@@ -14,7 +14,7 @@ import (
"time"
)
-var Version = "1.0.1"
+var Version = "1.1.1"
var UserAgent = "go2rtc/" + Version
var ConfigPath string
diff --git a/cmd/hls/hls.go b/cmd/hls/hls.go
new file mode 100644
index 00000000..32618c64
--- /dev/null
+++ b/cmd/hls/hls.go
@@ -0,0 +1,261 @@
+package hls
+
+import (
+ "fmt"
+ "github.com/AlexxIT/go2rtc/cmd/api"
+ "github.com/AlexxIT/go2rtc/cmd/streams"
+ "github.com/AlexxIT/go2rtc/pkg/mp4"
+ "github.com/AlexxIT/go2rtc/pkg/streamer"
+ "github.com/AlexxIT/go2rtc/pkg/ts"
+ "github.com/rs/zerolog/log"
+ "net/http"
+ "strconv"
+ "sync"
+ "time"
+)
+
+func Init() {
+ api.HandleFunc("api/stream.m3u8", handlerStream)
+ api.HandleFunc("api/hls/playlist.m3u8", handlerPlaylist)
+
+ // HLS (TS)
+ api.HandleFunc("api/hls/segment.ts", handlerSegmentTS)
+
+ // HLS (fMP4)
+ api.HandleFunc("api/hls/init.mp4", handlerInit)
+ api.HandleFunc("api/hls/segment.m4s", handlerSegmentMP4)
+}
+
+type Consumer interface {
+ streamer.Consumer
+ Init() ([]byte, error)
+ MimeCodecs() string
+ Start()
+}
+
+type Session struct {
+ cons Consumer
+ playlist string
+ init []byte
+ segment []byte
+ seq int
+ alive *time.Timer
+ mu sync.Mutex
+}
+
+const keepalive = 5 * time.Second
+
+var sessions = map[string]*Session{}
+
+func handlerStream(w http.ResponseWriter, r *http.Request) {
+ // CORS important for Chromecast
+ w.Header().Set("Access-Control-Allow-Origin", "*")
+ w.Header().Set("Content-Type", "application/vnd.apple.mpegurl")
+
+ if r.Method == "OPTIONS" {
+ w.Header().Set("Access-Control-Allow-Methods", "GET")
+ return
+ }
+
+ src := r.URL.Query().Get("src")
+ stream := streams.GetOrNew(src)
+ if stream == nil {
+ http.Error(w, api.StreamNotFound, http.StatusNotFound)
+ return
+ }
+
+ var cons Consumer
+
+ // use fMP4 with codecs filter and TS without
+ medias := mp4.ParseQuery(r.URL.Query())
+ if medias != nil {
+ cons = &mp4.Consumer{
+ RemoteAddr: r.RemoteAddr,
+ UserAgent: r.UserAgent(),
+ Medias: medias,
+ }
+ } else {
+ cons = &ts.Consumer{
+ RemoteAddr: r.RemoteAddr,
+ UserAgent: r.UserAgent(),
+ }
+ }
+
+ session := &Session{cons: cons}
+
+ cons.Listen(func(msg interface{}) {
+ if data, ok := msg.([]byte); ok {
+ session.mu.Lock()
+ session.segment = append(session.segment, data...)
+ session.mu.Unlock()
+ }
+ })
+
+ if err := stream.AddConsumer(cons); err != nil {
+ log.Error().Err(err).Caller().Send()
+ return
+ }
+
+ session.alive = time.AfterFunc(keepalive, func() {
+ stream.RemoveConsumer(cons)
+ })
+ session.init, _ = cons.Init()
+
+ cons.Start()
+
+ sid := strconv.FormatInt(time.Now().UnixNano(), 10)
+
+ // two segments important for Chromecast
+ if medias != nil {
+ session.playlist = `#EXTM3U
+#EXT-X-VERSION:6
+#EXT-X-TARGETDURATION:1
+#EXT-X-MEDIA-SEQUENCE:%d
+#EXT-X-MAP:URI="init.mp4?id=` + sid + `"
+#EXTINF:0.500,
+segment.m4s?id=` + sid + `&n=%d
+#EXTINF:0.500,
+segment.m4s?id=` + sid + `&n=%d`
+ } else {
+ session.playlist = `#EXTM3U
+#EXT-X-VERSION:3
+#EXT-X-TARGETDURATION:1
+#EXT-X-MEDIA-SEQUENCE:%d
+#EXTINF:0.500,
+segment.ts?id=` + sid + `&n=%d
+#EXTINF:0.500,
+segment.ts?id=` + sid + `&n=%d`
+ }
+
+ sessions[sid] = session
+
+ // bandwidth important for Safari, codecs useful for smooth playback
+ data := []byte(`#EXTM3U
+#EXT-X-STREAM-INF:BANDWIDTH=1000000,CODECS="` + cons.MimeCodecs() + `"
+hls/playlist.m3u8?id=` + sid)
+
+ if _, err := w.Write(data); err != nil {
+ log.Error().Err(err).Caller().Send()
+ }
+}
+
+func handlerPlaylist(w http.ResponseWriter, r *http.Request) {
+ w.Header().Set("Access-Control-Allow-Origin", "*")
+ w.Header().Set("Content-Type", "application/vnd.apple.mpegurl")
+
+ if r.Method == "OPTIONS" {
+ w.Header().Set("Access-Control-Allow-Methods", "GET")
+ return
+ }
+
+ sid := r.URL.Query().Get("id")
+ session := sessions[sid]
+ if session == nil {
+ http.NotFound(w, r)
+ return
+ }
+
+ s := fmt.Sprintf(session.playlist, session.seq, session.seq, session.seq+1)
+
+ if _, err := w.Write([]byte(s)); err != nil {
+ log.Error().Err(err).Caller().Send()
+ }
+}
+
+func handlerSegmentTS(w http.ResponseWriter, r *http.Request) {
+ w.Header().Set("Access-Control-Allow-Origin", "*")
+ w.Header().Set("Content-Type", "video/mp2t")
+
+ if r.Method == "OPTIONS" {
+ w.Header().Set("Access-Control-Allow-Methods", "GET")
+ return
+ }
+
+ sid := r.URL.Query().Get("id")
+ session := sessions[sid]
+ if session == nil {
+ http.NotFound(w, r)
+ return
+ }
+
+ session.alive.Reset(keepalive)
+
+ var i byte
+ for len(session.segment) == 0 {
+ if i++; i > 10 {
+ http.NotFound(w, r)
+ return
+ }
+ time.Sleep(time.Millisecond * 100)
+ }
+
+ session.mu.Lock()
+ data := session.segment
+ // important to start new segment with init
+ session.segment = session.init
+ session.seq++
+ session.mu.Unlock()
+
+ if _, err := w.Write(data); err != nil {
+ log.Error().Err(err).Caller().Send()
+ }
+}
+
+func handlerInit(w http.ResponseWriter, r *http.Request) {
+ w.Header().Set("Access-Control-Allow-Origin", "*")
+ w.Header().Add("Content-Type", "video/mp4")
+
+ if r.Method == "OPTIONS" {
+ w.Header().Set("Access-Control-Allow-Methods", "GET")
+ return
+ }
+
+ sid := r.URL.Query().Get("id")
+ session := sessions[sid]
+ if session == nil {
+ http.NotFound(w, r)
+ return
+ }
+
+ if _, err := w.Write(session.init); err != nil {
+ log.Error().Err(err).Caller().Send()
+ }
+}
+
+func handlerSegmentMP4(w http.ResponseWriter, r *http.Request) {
+ w.Header().Set("Access-Control-Allow-Origin", "*")
+ w.Header().Add("Content-Type", "video/iso.segment")
+
+ if r.Method == "OPTIONS" {
+ w.Header().Set("Access-Control-Allow-Methods", "GET, OPTIONS")
+ return
+ }
+
+ sid := r.URL.Query().Get("id")
+ session := sessions[sid]
+ if session == nil {
+ http.NotFound(w, r)
+ return
+ }
+
+ session.alive.Reset(keepalive)
+
+ var i byte
+ for len(session.segment) == 0 {
+ if i++; i > 10 {
+ http.NotFound(w, r)
+ return
+ }
+ time.Sleep(time.Millisecond * 100)
+ }
+
+ session.mu.Lock()
+ data := session.segment
+ session.segment = nil
+ session.seq++
+ session.mu.Unlock()
+
+ if _, err := w.Write(data); err != nil {
+ log.Error().Err(err).Caller().Send()
+ }
+}
diff --git a/cmd/mp4/mp4.go b/cmd/mp4/mp4.go
index abcedef4..d98cae80 100644
--- a/cmd/mp4/mp4.go
+++ b/cmd/mp4/mp4.go
@@ -26,8 +26,14 @@ func Init() {
var log zerolog.Logger
func handlerKeyframe(w http.ResponseWriter, r *http.Request) {
- if isChromeFirst(w, r) {
- return
+ // Chrome 105 does two requests: without Range and with `Range: bytes=0-`
+ ua := r.UserAgent()
+ if strings.Contains(ua, " Chrome/") {
+ if r.Header.Values("Range") == nil {
+ w.Header().Set("Content-Type", "video/mp4")
+ w.WriteHeader(http.StatusOK)
+ return
+ }
}
src := r.URL.Query().Get("src")
@@ -68,7 +74,22 @@ func handlerKeyframe(w http.ResponseWriter, r *http.Request) {
func handlerMP4(w http.ResponseWriter, r *http.Request) {
log.Trace().Msgf("[mp4] %s %+v", r.Method, r.Header)
- if isChromeFirst(w, r) || isSafari(w, r) {
+ // Chrome has Safari in UA, so check first Chrome and later Safari
+ ua := r.UserAgent()
+ if strings.Contains(ua, " Chrome/") {
+ if r.Header.Values("Range") == nil {
+ w.Header().Set("Content-Type", "video/mp4")
+ w.WriteHeader(http.StatusOK)
+ return
+ }
+ } else if strings.Contains(ua, " Safari/") {
+ // auto redirect to HLS/fMP4 format, because Safari not support MP4 stream
+ url := "stream.m3u8?" + r.URL.RawQuery
+ if !r.URL.Query().Has("mp4") {
+ url += "&mp4"
+ }
+
+ http.Redirect(w, r, url, http.StatusMovedPermanently)
return
}
@@ -84,8 +105,8 @@ func handlerMP4(w http.ResponseWriter, r *http.Request) {
cons := &mp4.Consumer{
RemoteAddr: r.RemoteAddr,
UserAgent: r.UserAgent(),
+ Medias: streamer.ParseQuery(r.URL.Query()),
}
- cons.Medias = streamer.ParseQuery(r.URL.Query())
cons.Listen(func(msg interface{}) {
if data, ok := msg.([]byte); ok {
@@ -138,23 +159,3 @@ func handlerMP4(w http.ResponseWriter, r *http.Request) {
duration.Stop()
}
}
-
-func isChromeFirst(w http.ResponseWriter, r *http.Request) bool {
- // Chrome 105 does two requests: without Range and with `Range: bytes=0-`
- if strings.Contains(r.UserAgent(), " Chrome/") {
- if r.Header.Values("Range") == nil {
- w.Header().Set("Content-Type", "video/mp4")
- w.WriteHeader(http.StatusOK)
- return true
- }
- }
- return false
-}
-
-func isSafari(w http.ResponseWriter, r *http.Request) bool {
- if r.Header.Get("Range") == "bytes=0-1" {
- handlerKeyframe(w, r)
- return true
- }
- return false
-}
diff --git a/cmd/rtsp/rtsp.go b/cmd/rtsp/rtsp.go
index aa910a98..ce9a723b 100644
--- a/cmd/rtsp/rtsp.go
+++ b/cmd/rtsp/rtsp.go
@@ -9,20 +9,23 @@ import (
"github.com/AlexxIT/go2rtc/pkg/tcp"
"github.com/rs/zerolog"
"net"
+ "net/url"
"strings"
)
func Init() {
var conf struct {
Mod struct {
- Listen string `yaml:"listen" json:"listen"`
- Username string `yaml:"username" json:"-"`
- Password string `yaml:"password" json:"-"`
+ Listen string `yaml:"listen" json:"listen"`
+ Username string `yaml:"username" json:"-"`
+ Password string `yaml:"password" json:"-"`
+ DefaultQuery string `yaml:"default_query" json:"default_query"`
} `yaml:"rtsp"`
}
// default config
conf.Mod.Listen = ":8554"
+ conf.Mod.DefaultQuery = "video&audio"
app.LoadConfig(&conf)
app.Info["rtsp"] = conf.Mod
@@ -50,6 +53,10 @@ func Init() {
log.Info().Str("addr", address).Msg("[rtsp] listen")
+ if query, err := url.ParseQuery(conf.Mod.DefaultQuery); err == nil {
+ defaultMedias = mp4.ParseQuery(query)
+ }
+
go func() {
for {
conn, err := ln.Accept()
@@ -79,6 +86,7 @@ var Port string
var log zerolog.Logger
var handlers []Handler
+var defaultMedias []*streamer.Media
func rtspHandler(url string) (streamer.Producer, error) {
backchannel := true
@@ -104,6 +112,8 @@ func rtspHandler(url string) (streamer.Producer, error) {
log.Trace().Msgf("[rtsp] client request:\n%s", msg)
case *tcp.Response:
log.Trace().Msgf("[rtsp] client response:\n%s", msg)
+ case string:
+ log.Trace().Msgf("[rtsp] client msg: %s", msg)
}
})
}
@@ -165,7 +175,12 @@ func tcpHandler(conn *rtsp.Conn) {
conn.SessionName = app.UserAgent
- conn.Medias = ParseQuery(conn.URL.Query())
+ conn.Medias = mp4.ParseQuery(conn.URL.Query())
+ if conn.Medias == nil {
+ for _, media := range defaultMedias {
+ conn.Medias = append(conn.Medias, media.Clone())
+ }
+ }
if err := stream.AddConsumer(conn); err != nil {
log.Warn().Err(err).Str("stream", name).Msg("[rtsp]")
@@ -229,12 +244,3 @@ func tcpHandler(conn *rtsp.Conn) {
_ = conn.Close()
}
-
-func ParseQuery(query map[string][]string) []*streamer.Media {
- if query["mp4"] != nil {
- cons := mp4.Consumer{}
- return cons.GetMedias()
- }
-
- return streamer.ParseQuery(query)
-}
diff --git a/main.go b/main.go
index 63826914..63b7072e 100644
--- a/main.go
+++ b/main.go
@@ -8,6 +8,7 @@ import (
"github.com/AlexxIT/go2rtc/cmd/exec"
"github.com/AlexxIT/go2rtc/cmd/ffmpeg"
"github.com/AlexxIT/go2rtc/cmd/hass"
+ "github.com/AlexxIT/go2rtc/cmd/hls"
"github.com/AlexxIT/go2rtc/cmd/homekit"
"github.com/AlexxIT/go2rtc/cmd/http"
"github.com/AlexxIT/go2rtc/cmd/ivideon"
@@ -42,6 +43,7 @@ func main() {
webrtc.Init()
mp4.Init()
+ hls.Init()
mjpeg.Init()
http.Init()
diff --git a/pkg/h264/rtp.go b/pkg/h264/rtp.go
index 6044caae..024ccf74 100644
--- a/pkg/h264/rtp.go
+++ b/pkg/h264/rtp.go
@@ -10,6 +10,8 @@ import (
const RTPPacketVersionAVC = 0
+const PSMaxSize = 128 // the biggest SPS I've seen is 48 (EZVIZ CS-CV210)
+
func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
depack := &codecs.H264Packet{IsAVC: true}
@@ -29,7 +31,7 @@ func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
// Fix TP-Link Tapo TC70: sends SPS and PPS with packet.Marker = true
// Reolink Duo 2: sends SPS with Marker and PPS without
- if packet.Marker && len(payload) < 128 {
+ if packet.Marker && len(payload) < PSMaxSize {
switch NALUType(payload) {
case NALUTypeSPS, NALUTypePPS:
buf = append(buf, payload...)
@@ -70,7 +72,10 @@ func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
if len(buf) > 0 {
payload = append(buf, payload...)
buf = buf[:0]
- } else {
+ }
+
+ // should not be that huge SPS
+ if NALUType(payload) == NALUTypeSPS && binary.BigEndian.Uint32(payload) >= PSMaxSize {
// some Chinese buggy cameras has single packet with SPS+PPS+IFrame separated by 00 00 00 01
// https://github.com/AlexxIT/WebRTC/issues/391
// https://github.com/AlexxIT/WebRTC/issues/392
diff --git a/pkg/mp4/consumer.go b/pkg/mp4/consumer.go
index 8d5fbf2d..496ae65b 100644
--- a/pkg/mp4/consumer.go
+++ b/pkg/mp4/consumer.go
@@ -24,6 +24,16 @@ type Consumer struct {
send uint32
}
+// ParseQuery - like usual parse, but with mp4 param handler
+func ParseQuery(query map[string][]string) []*streamer.Media {
+ if query["mp4"] != nil {
+ cons := Consumer{}
+ return cons.GetMedias()
+ }
+
+ return streamer.ParseQuery(query)
+}
+
const (
waitNone byte = iota
waitKeyframe
@@ -161,8 +171,12 @@ func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *strea
panic("unsupported codec")
}
+func (c *Consumer) MimeCodecs() string {
+ return c.muxer.MimeCodecs(c.codecs)
+}
+
func (c *Consumer) MimeType() string {
- return c.muxer.MimeType(c.codecs)
+ return `video/mp4; codecs="` + c.MimeCodecs() + `"`
}
func (c *Consumer) Init() ([]byte, error) {
diff --git a/pkg/mp4/muxer.go b/pkg/mp4/muxer.go
index 83bb33c5..cd7d9762 100644
--- a/pkg/mp4/muxer.go
+++ b/pkg/mp4/muxer.go
@@ -24,8 +24,8 @@ const (
MimeOpus = "opus"
)
-func (m *Muxer) MimeType(codecs []*streamer.Codec) string {
- s := `video/mp4; codecs="`
+func (m *Muxer) MimeCodecs(codecs []*streamer.Codec) string {
+ var s string
for i, codec := range codecs {
if i > 0 {
@@ -46,7 +46,7 @@ func (m *Muxer) MimeType(codecs []*streamer.Codec) string {
}
}
- return s + `"`
+ return s
}
func (m *Muxer) GetInit(codecs []*streamer.Codec) ([]byte, error) {
diff --git a/pkg/mp4/segment.go b/pkg/mp4/segment.go
index 9cc3a88a..6fac7a40 100644
--- a/pkg/mp4/segment.go
+++ b/pkg/mp4/segment.go
@@ -50,7 +50,7 @@ func (c *Segment) AddTrack(media *streamer.Media, track *streamer.Track) *stream
return nil
}
- c.MimeType = muxer.MimeType(codecs)
+ c.MimeType = `video/mp4; codecs="` + muxer.MimeCodecs(codecs) + `"`
switch track.Codec.Name {
case streamer.CodecH264:
diff --git a/pkg/mp4/v1/consumer.go b/pkg/mp4/v1/consumer.go
index 75dbd59e..491b668a 100644
--- a/pkg/mp4/v1/consumer.go
+++ b/pkg/mp4/v1/consumer.go
@@ -1,6 +1,8 @@
package mp4
import (
+ "encoding/hex"
+ "github.com/AlexxIT/go2rtc/pkg/aac"
"github.com/AlexxIT/go2rtc/pkg/h264"
"github.com/AlexxIT/go2rtc/pkg/streamer"
"github.com/deepch/vdk/av"
@@ -101,7 +103,17 @@ func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *strea
return track.Bind(push)
case streamer.CodecAAC:
- stream, _ := aacparser.NewCodecDataFromMPEG4AudioConfigBytes([]byte{20, 8})
+ s := streamer.Between(codec.FmtpLine, "config=", ";")
+
+ b, err := hex.DecodeString(s)
+ if err != nil {
+ return nil
+ }
+
+ stream, err := aacparser.NewCodecDataFromMPEG4AudioConfigBytes(b)
+ if err != nil {
+ return nil
+ }
c.mimeType += ",mp4a.40.2"
c.streams = append(c.streams, stream)
@@ -131,6 +143,11 @@ func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *strea
return nil
}
+ if codec.IsRTP() {
+ wrapper := aac.RTPDepay(track)
+ push = wrapper(push)
+ }
+
return track.Bind(push)
}
diff --git a/pkg/rtsp/conn.go b/pkg/rtsp/conn.go
index 39703673..40c07fd2 100644
--- a/pkg/rtsp/conn.go
+++ b/pkg/rtsp/conn.go
@@ -304,6 +304,12 @@ func (c *Conn) Describe() error {
req.Header.Set("Require", "www.onvif.org/ver20/backchannel")
}
+ if c.UserAgent != "" {
+ // this camera will answer with 401 on DESCRIBE without User-Agent
+ // https://github.com/AlexxIT/go2rtc/issues/235
+ req.Header.Set("User-Agent", c.UserAgent)
+ }
+
res, err := c.Do(req)
if err != nil {
return err
@@ -347,7 +353,7 @@ func (c *Conn) Describe() error {
func (c *Conn) Setup() error {
for _, media := range c.Medias {
- _, err := c.SetupMedia(media, media.Codecs[0])
+ _, err := c.SetupMedia(media, media.Codecs[0], true)
if err != nil {
return err
}
@@ -356,11 +362,12 @@ func (c *Conn) Setup() error {
return nil
}
-func (c *Conn) SetupMedia(
- media *streamer.Media, codec *streamer.Codec,
-) (*streamer.Track, error) {
- c.stateMu.Lock()
- defer c.stateMu.Unlock()
+func (c *Conn) SetupMedia(media *streamer.Media, codec *streamer.Codec, first bool) (*streamer.Track, error) {
+ // TODO: rewrite recoonection and first flag
+ if first {
+ c.stateMu.Lock()
+ defer c.stateMu.Unlock()
+ }
if c.state != StateConn && c.state != StateSetup {
return nil, fmt.Errorf("RTSP SETUP from wrong state: %s", c.state)
@@ -412,7 +419,7 @@ func (c *Conn) SetupMedia(
for _, newMedia := range c.Medias {
if newMedia.Control == media.Control {
- return c.SetupMedia(newMedia, newMedia.Codecs[0])
+ return c.SetupMedia(newMedia, newMedia.Codecs[0], false)
}
}
}
@@ -742,6 +749,9 @@ func (c *Conn) Handle() (err error) {
return
}
+ var channelID byte
+ var size uint16
+
if buf4[0] != '$' {
switch string(buf4) {
case "RTSP":
@@ -750,26 +760,62 @@ func (c *Conn) Handle() (err error) {
return
}
c.Fire(res)
+ continue
+
case "OPTI", "TEAR", "DESC", "SETU", "PLAY", "PAUS", "RECO", "ANNO", "GET_", "SET_":
var req *tcp.Request
if req, err = tcp.ReadRequest(c.reader); err != nil {
return
}
c.Fire(req)
+ continue
+
default:
- return fmt.Errorf("RTSP wrong input")
+ for i := 0; ; i++ {
+ // search next start symbol
+ if _, err = c.reader.ReadBytes('$'); err != nil {
+ return err
+ }
+
+ if channelID, err = c.reader.ReadByte(); err != nil {
+ return err
+ }
+
+ // check if channel ID exists
+ if c.channels[channelID] == nil {
+ continue
+ }
+
+ buf4 = make([]byte, 2)
+ if _, err = io.ReadFull(c.reader, buf4); err != nil {
+ return err
+ }
+
+ // check if size good for RTP
+ size = binary.BigEndian.Uint16(buf4)
+ if size <= 1500 {
+ break
+ }
+
+ // 10 tries to find good packet
+ if i >= 10 {
+ return fmt.Errorf("RTSP wrong input")
+ }
+ }
+
+ c.Fire("RTSP wrong input")
}
- continue
- }
+ } else {
+ // hope that the odd channels are always RTCP
+ channelID = buf4[1]
- // hope that the odd channels are always RTCP
- channelID := buf4[1]
+ // get data size
+ size = binary.BigEndian.Uint16(buf4[2:])
- // get data size
- size := int(binary.BigEndian.Uint16(buf4[2:]))
-
- if _, err = c.reader.Discard(4); err != nil {
- return
+ // skip 4 bytes from c.reader.Peek
+ if _, err = c.reader.Discard(4); err != nil {
+ return
+ }
}
// init memory for data
@@ -778,7 +824,7 @@ func (c *Conn) Handle() (err error) {
return
}
- c.receive += size
+ c.receive += int(size)
if channelID&1 == 0 {
packet := &rtp.Packet{}
@@ -789,10 +835,8 @@ func (c *Conn) Handle() (err error) {
track := c.channels[channelID]
if track != nil {
_ = track.WriteRTP(packet)
- //return fmt.Errorf("wrong channelID: %d", channelID)
} else {
- continue // TODO: maybe fix this
- //panic("wrong channelID")
+ c.Fire("wrong channelID: " + strconv.Itoa(int(channelID)))
}
} else {
msg := &RTCP{Channel: channelID}
diff --git a/pkg/rtsp/helpers.go b/pkg/rtsp/helpers.go
index f748c9a2..893f3b87 100644
--- a/pkg/rtsp/helpers.go
+++ b/pkg/rtsp/helpers.go
@@ -5,6 +5,7 @@ import (
"github.com/AlexxIT/go2rtc/pkg/streamer"
"github.com/pion/rtcp"
"net/url"
+ "regexp"
"strings"
)
@@ -22,9 +23,12 @@ t=0 0`
func UnmarshalSDP(rawSDP []byte) ([]*streamer.Media, error) {
medias, err := streamer.UnmarshalSDP(rawSDP)
if err != nil {
+ // fix multiple `s=` https://github.com/AlexxIT/WebRTC/issues/417
+ re, _ := regexp.Compile("\ns=[^\n]+")
+ rawSDP = re.ReplaceAll(rawSDP, nil)
+
// fix SDP header for some cameras
- i := bytes.Index(rawSDP, []byte("\nm="))
- if i > 0 {
+ if i := bytes.Index(rawSDP, []byte("\nm=")); i > 0 {
rawSDP = append([]byte(sdpHeader), rawSDP[i:]...)
medias, err = streamer.UnmarshalSDP(rawSDP)
}
diff --git a/pkg/rtsp/rtsp_test.go b/pkg/rtsp/rtsp_test.go
index 7e32b272..e7024f4e 100644
--- a/pkg/rtsp/rtsp_test.go
+++ b/pkg/rtsp/rtsp_test.go
@@ -18,3 +18,35 @@ func TestURLParse(t *testing.T) {
assert.Empty(t, err)
assert.Equal(t, "turret2-cam.lan:554", u.Host)
}
+
+func TestMultipleSinSDP(t *testing.T) {
+ s := `v=0
+o=- 91674849066 1 IN IP4 192.168.1.123
+s=RtspServer
+i=live
+t=0 0
+a=control:*
+a=range:npt=0-
+m=video 0 RTP/AVP 96
+c=IN IP4 0.0.0.0
+s=RtspServer
+i=live
+a=control:track0
+a=range:npt=0-
+a=rtpmap:96 H264/90000
+a=fmtp:96 packetization-mode=1;profile-level-id=42001E;sprop-parameter-sets=Z0IAHvQCgC3I,aM48gA==
+a=control:track0
+m=audio 0 RTP/AVP 97
+c=IN IP4 0.0.0.0
+s=RtspServer
+i=live
+a=control:track1
+a=range:npt=0-
+a=rtpmap:97 MPEG4-GENERIC/8000/1
+a=fmtp:97 profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3;config=1588
+a=control:track1
+`
+ medias, err := UnmarshalSDP([]byte(s))
+ assert.Nil(t, err)
+ assert.NotNil(t, medias)
+}
diff --git a/pkg/rtsp/streamer.go b/pkg/rtsp/streamer.go
index 2af46629..7fea6d97 100644
--- a/pkg/rtsp/streamer.go
+++ b/pkg/rtsp/streamer.go
@@ -45,7 +45,7 @@ func (c *Conn) GetTrack(media *streamer.Media, codec *streamer.Codec) *streamer.
return streamer.NewTrack(codec, media.Direction)
}
- track, err := c.SetupMedia(media, codec)
+ track, err := c.SetupMedia(media, codec, true)
if err != nil {
return nil
}
diff --git a/pkg/ts/ts.go b/pkg/ts/ts.go
new file mode 100644
index 00000000..ad80e0ca
--- /dev/null
+++ b/pkg/ts/ts.go
@@ -0,0 +1,200 @@
+package ts
+
+import (
+ "bytes"
+ "encoding/hex"
+ "github.com/AlexxIT/go2rtc/pkg/aac"
+ "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/ts"
+ "github.com/pion/rtp"
+ "sync/atomic"
+ "time"
+)
+
+type Consumer struct {
+ streamer.Element
+
+ UserAgent string
+ RemoteAddr string
+
+ buf *bytes.Buffer
+ muxer *ts.Muxer
+ mimeType string
+ streams []av.CodecData
+ start bool
+ init []byte
+
+ send uint32
+}
+
+func (c *Consumer) GetMedias() []*streamer.Media {
+ return []*streamer.Media{
+ {
+ Kind: streamer.KindVideo,
+ Direction: streamer.DirectionRecvonly,
+ Codecs: []*streamer.Codec{
+ {Name: streamer.CodecH264},
+ },
+ },
+ //{
+ // Kind: streamer.KindAudio,
+ // Direction: streamer.DirectionRecvonly,
+ // Codecs: []*streamer.Codec{
+ // {Name: streamer.CodecAAC},
+ // },
+ //},
+ }
+}
+
+func (c *Consumer) AddTrack(media *streamer.Media, track *streamer.Track) *streamer.Track {
+ codec := track.Codec
+ trackID := int8(len(c.streams))
+
+ switch codec.Name {
+ case streamer.CodecH264:
+ sps, pps := h264.GetParameterSet(codec.FmtpLine)
+ stream, err := h264parser.NewCodecDataFromSPSAndPPS(sps, pps)
+ if err != nil {
+ return nil
+ }
+
+ if len(c.mimeType) > 0 {
+ c.mimeType += ","
+ }
+
+ // TODO: fixme
+ // some devices won't play high level
+ if stream.RecordInfo.AVCLevelIndication <= 0x29 {
+ c.mimeType += "avc1." + h264.GetProfileLevelID(codec.FmtpLine)
+ } else {
+ c.mimeType += "avc1.640029"
+ }
+
+ c.streams = append(c.streams, stream)
+
+ pkt := av.Packet{Idx: trackID, CompositionTime: time.Millisecond}
+
+ ts2time := time.Second / time.Duration(codec.ClockRate)
+
+ push := func(packet *rtp.Packet) error {
+ if packet.Version != h264.RTPPacketVersionAVC {
+ return nil
+ }
+
+ if !c.start {
+ return nil
+ }
+
+ pkt.Data = packet.Payload
+ newTime := time.Duration(packet.Timestamp) * ts2time
+ if pkt.Time > 0 {
+ pkt.Duration = newTime - pkt.Time
+ }
+ pkt.Time = newTime
+
+ if err = c.muxer.WritePacket(pkt); err != nil {
+ return err
+ }
+
+ // clone bytes from buffer, so next packet won't overwrite it
+ buf := append([]byte{}, c.buf.Bytes()...)
+ atomic.AddUint32(&c.send, uint32(len(buf)))
+ c.Fire(buf)
+
+ c.buf.Reset()
+
+ return nil
+ }
+
+ if codec.IsRTP() {
+ wrapper := h264.RTPDepay(track)
+ push = wrapper(push)
+ }
+
+ return track.Bind(push)
+
+ case streamer.CodecAAC:
+ s := streamer.Between(codec.FmtpLine, "config=", ";")
+
+ b, err := hex.DecodeString(s)
+ if err != nil {
+ return nil
+ }
+
+ stream, err := aacparser.NewCodecDataFromMPEG4AudioConfigBytes(b)
+ if err != nil {
+ return nil
+ }
+
+ if len(c.mimeType) > 0 {
+ c.mimeType += ","
+ }
+
+ c.mimeType += "mp4a.40.2"
+ c.streams = append(c.streams, stream)
+
+ pkt := av.Packet{Idx: trackID, CompositionTime: time.Millisecond}
+
+ ts2time := time.Second / time.Duration(codec.ClockRate)
+
+ push := func(packet *rtp.Packet) error {
+ if !c.start {
+ return nil
+ }
+
+ pkt.Data = packet.Payload
+ newTime := time.Duration(packet.Timestamp) * ts2time
+ if pkt.Time > 0 {
+ pkt.Duration = newTime - pkt.Time
+ }
+ pkt.Time = newTime
+
+ if err := c.muxer.WritePacket(pkt); err != nil {
+ return err
+ }
+
+ // clone bytes from buffer, so next packet won't overwrite it
+ buf := append([]byte{}, c.buf.Bytes()...)
+ atomic.AddUint32(&c.send, uint32(len(buf)))
+ c.Fire(buf)
+
+ c.buf.Reset()
+
+ return nil
+ }
+
+ if codec.IsRTP() {
+ wrapper := aac.RTPDepay(track)
+ push = wrapper(push)
+ }
+
+ return track.Bind(push)
+ }
+
+ panic("unsupported codec")
+}
+
+func (c *Consumer) MimeCodecs() string {
+ return c.mimeType
+}
+
+func (c *Consumer) Init() ([]byte, error) {
+ c.buf = bytes.NewBuffer(nil)
+ c.muxer = ts.NewMuxer(c.buf)
+
+ // first packet will be with header, it's ok
+ if err := c.muxer.WriteHeader(c.streams); err != nil {
+ return nil, err
+ }
+ data := append([]byte{}, c.buf.Bytes()...)
+
+ return data, nil
+}
+
+func (c *Consumer) Start() {
+ c.start = true
+}
diff --git a/www/README.md b/www/README.md
index 79d93b45..cb34a782 100644
--- a/www/README.md
+++ b/www/README.md
@@ -48,6 +48,18 @@ pc.ontrack = ev => {
}
```
+## Chromecast 1
+
+2023-02-02. Error:
+
+```
+InvalidStateError: Failed to execute 'addTransceiver' on 'RTCPeerConnection': This operation is only supported in 'unified-plan'. 'unified-plan' will become the default behavior in the future, but it is currently experimental. To try it out, construct the RTCPeerConnection with sdpSemantics:'unified-plan' present in the RTCConfiguration argument.
+```
+
+User-Agent: `Mozilla/5.0 (X11; Linux armv7l) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/70.0.3538.47 Safari/537.36 CrKey/1.36.159268`
+
+https://webrtc.org/getting-started/unified-plan-transition-guide?hl=en
+
## Useful links
- https://www.webrtc-experiment.com/DetectRTC/
@@ -58,3 +70,4 @@ pc.ontrack = ev => {
- https://chromium.googlesource.com/external/w3c/web-platform-tests/+/refs/heads/master/media-source/mediasource-is-type-supported.html
- https://googlechrome.github.io/samples/media/sourcebuffer-changetype.html
- https://chromestatus.com/feature/5100845653819392
+- https://developer.apple.com/documentation/webkit/delivering_video_content_for_safari
diff --git a/www/index.html b/www/index.html
index 4781b57f..86497bae 100644
--- a/www/index.html
+++ b/www/index.html
@@ -89,9 +89,7 @@
const templates = [
'stream',
'2-way-aud',
- 'mp4',
- 'mjpeg',
- 'info',
+ 'links',
'delete',
];
@@ -138,15 +136,17 @@
for (const [name, value] of Object.entries(data)) {
const online = value && value.consumers ? value.consumers.length : 0;
+ const src = encodeURIComponent(name);
const links = templates.map(link => {
- return link.replace("{name}", encodeURIComponent(name));
+ return link.replace("{name}", src);
}).join(" ");
const tr = document.createElement("tr");
tr.dataset["id"] = name;
tr.innerHTML =
` | ` +
- `${online} | ${links} | `;
+ `${online} / info | ` +
+ `${links} | `;
tbody.appendChild(tr);
}
});
@@ -156,17 +156,9 @@
fetch(url, {cache: 'no-cache'}).then(r => r.json()).then(data => {
const info = document.querySelector(".info");
info.innerText = `Version: ${data.version}, Config: ${data.config_path}`;
-
- try {
- const host = data.host.match(/^[^:]+/)[0];
- const port = data.rtsp.listen.match(/[0-9]+$/)[0];
- templates.splice(4, 0, `rtsp`);
- } catch (e) {
- templates.splice(4, 0, `rtsp`);
- }
-
- reload();
});
+
+ reload();