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picamera2, MediaMTX (formerly rtsp-simple-server) scripts and config. More details here https://www.wtip.net/
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############################################### | |
# Global settings | |
# Settings in this section are applied anywhere. | |
############################################### | |
# Global settings -> General | |
# Verbosity of the program; available values are "error", "warn", "info", "debug". | |
logLevel: info | |
# Destinations of log messages; available values are "stdout", "file" and "syslog". | |
logDestinations: [stdout] | |
# If "file" is in logDestinations, this is the file which will receive the logs. | |
logFile: mediamtx.log | |
# Timeout of read operations. | |
readTimeout: 10s | |
# Timeout of write operations. | |
writeTimeout: 10s | |
# Size of the queue of outgoing packets. | |
# A higher value allows to increase throughput, a lower value allows to save RAM. | |
writeQueueSize: 512 | |
# Maximum size of outgoing UDP packets. | |
# This can be decreased to avoid fragmentation on networks with a low UDP MTU. | |
udpMaxPayloadSize: 1472 | |
# Command to run when a client connects to the server. | |
# This is terminated with SIGINT when a client disconnects from the server. | |
# The following environment variables are available: | |
# * RTSP_PORT: RTSP server port | |
# * MTX_CONN_TYPE: connection type | |
# * MTX_CONN_ID: connection ID | |
runOnConnect: | |
# Restart the command if it exits. | |
runOnConnectRestart: no | |
# Command to run when a client disconnects from the server. | |
# Environment variables are the same of runOnConnect. | |
runOnDisconnect: | |
############################################### | |
# Global settings -> Authentication | |
# Authentication method. Available values are: | |
# * internal: users are stored in the configuration file | |
# * http: an external HTTP URL is contacted to perform authentication | |
# * jwt: an external identity server provides authentication through JWTs | |
authMethod: internal | |
# Internal authentication. | |
# list of users. | |
authInternalUsers: | |
# Default unprivileged user. | |
# Username. 'any' means any user, including anonymous ones. | |
- user: any | |
# Password. Not used in case of 'any' user. | |
pass: | |
# IPs or networks allowed to use this user. An empty list means any IP. | |
ips: [] | |
# List of permissions. | |
permissions: | |
# Available actions are: publish, read, playback, api, metrics, pprof. | |
# Paths can be set to further restrict access to a specific path. | |
# An empty path means any path. | |
# Regular expressions can be used by using a tilde as prefix. | |
- action: read | |
path: | |
- action: playback | |
path: | |
# Default administrator. | |
# This allows to use API, metrics and PPROF without authentication, | |
# if the IP is localhost. | |
- user: any | |
pass: | |
ips: ['127.0.0.1', '::1'] | |
permissions: | |
- action: api | |
- action: metrics | |
- action: pprof | |
- user: myuser | |
pass: mypass | |
ips: [] | |
permissions: | |
- action: publish | |
# HTTP-based authentication. | |
# URL called to perform authentication. Every time a user wants | |
# to authenticate, the server calls this URL with the POST method | |
# and a body containing: | |
# { | |
# "user": "user", | |
# "password": "password", | |
# "ip": "ip", | |
# "action": "publish|read|playback|api|metrics|pprof", | |
# "path": "path", | |
# "protocol": "rtsp|rtmp|hls|webrtc|srt", | |
# "id": "id", | |
# "query": "query" | |
# } | |
# If the response code is 20x, authentication is accepted, otherwise | |
# it is discarded. | |
authHTTPAddress: | |
# Actions to exclude from HTTP-based authentication. | |
# Format is the same as the one of user permissions. | |
authHTTPExclude: | |
- action: api | |
- action: metrics | |
- action: pprof | |
# JWT-based authentication. | |
# Users have to login through an external identity server and obtain a JWT. | |
# This JWT must contain the claim "mediamtx_permissions" with permissions, | |
# for instance: | |
# { | |
# ... | |
# "mediamtx_permissions": [ | |
# { | |
# "action": "publish", | |
# "path": "somepath" | |
# } | |
# ] | |
# } | |
# Users are expected to pass the JWT in the Authorization header or as a query parameter. | |
# This is the JWKS URL that will be used to pull (once) the public key that allows | |
# to validate JWTs. | |
authJWTJWKS: | |
############################################### | |
# Global settings -> Control API | |
# Enable controlling the server through the Control API. | |
api: no | |
# Address of the Control API listener. | |
apiAddress: :9997 | |
# Enable TLS/HTTPS on the Control API server. | |
apiEncryption: no | |
# Path to the server key. This is needed only when encryption is yes. | |
# This can be generated with: | |
# openssl genrsa -out server.key 2048 | |
# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650 | |
apiServerKey: server.key | |
# Path to the server certificate. | |
apiServerCert: server.crt | |
# Value of the Access-Control-Allow-Origin header provided in every HTTP response. | |
apiAllowOrigin: '*' | |
# List of IPs or CIDRs of proxies placed before the HTTP server. | |
# If the server receives a request from one of these entries, IP in logs | |
# will be taken from the X-Forwarded-For header. | |
apiTrustedProxies: [] | |
############################################### | |
# Global settings -> Metrics | |
# Enable Prometheus-compatible metrics. | |
metrics: no | |
# Address of the metrics HTTP listener. | |
metricsAddress: :9998 | |
# Enable TLS/HTTPS on the Metrics server. | |
metricsEncryption: no | |
# Path to the server key. This is needed only when encryption is yes. | |
# This can be generated with: | |
# openssl genrsa -out server.key 2048 | |
# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650 | |
metricsServerKey: server.key | |
# Path to the server certificate. | |
metricsServerCert: server.crt | |
# Value of the Access-Control-Allow-Origin header provided in every HTTP response. | |
metricsAllowOrigin: '*' | |
# List of IPs or CIDRs of proxies placed before the HTTP server. | |
# If the server receives a request from one of these entries, IP in logs | |
# will be taken from the X-Forwarded-For header. | |
metricsTrustedProxies: [] | |
############################################### | |
# Global settings -> PPROF | |
# Enable pprof-compatible endpoint to monitor performances. | |
pprof: no | |
# Address of the pprof listener. | |
pprofAddress: :9999 | |
# Enable TLS/HTTPS on the pprof server. | |
pprofEncryption: no | |
# Path to the server key. This is needed only when encryption is yes. | |
# This can be generated with: | |
# openssl genrsa -out server.key 2048 | |
# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650 | |
pprofServerKey: server.key | |
# Path to the server certificate. | |
pprofServerCert: server.crt | |
# Value of the Access-Control-Allow-Origin header provided in every HTTP response. | |
pprofAllowOrigin: '*' | |
# List of IPs or CIDRs of proxies placed before the HTTP server. | |
# If the server receives a request from one of these entries, IP in logs | |
# will be taken from the X-Forwarded-For header. | |
pprofTrustedProxies: [] | |
############################################### | |
# Global settings -> Playback server | |
# Enable downloading recordings from the playback server. | |
playback: no | |
# Address of the playback server listener. | |
playbackAddress: :9996 | |
# Enable TLS/HTTPS on the playback server. | |
playbackEncryption: no | |
# Path to the server key. This is needed only when encryption is yes. | |
# This can be generated with: | |
# openssl genrsa -out server.key 2048 | |
# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650 | |
playbackServerKey: server.key | |
# Path to the server certificate. | |
playbackServerCert: server.crt | |
# Value of the Access-Control-Allow-Origin header provided in every HTTP response. | |
playbackAllowOrigin: '*' | |
# List of IPs or CIDRs of proxies placed before the HTTP server. | |
# If the server receives a request from one of these entries, IP in logs | |
# will be taken from the X-Forwarded-For header. | |
playbackTrustedProxies: [] | |
############################################### | |
# Global settings -> RTSP server | |
# Enable publishing and reading streams with the RTSP protocol. | |
rtsp: yes | |
# List of enabled RTSP transport protocols. | |
# UDP is the most performant, but doesn't work when there's a NAT/firewall between | |
# server and clients, and doesn't support encryption. | |
# UDP-multicast allows to save bandwidth when clients are all in the same LAN. | |
# TCP is the most versatile, and does support encryption. | |
# The handshake is always performed with TCP. | |
protocols: [udp, multicast, tcp] | |
# Encrypt handshakes and TCP streams with TLS (RTSPS). | |
# Available values are "no", "strict", "optional". | |
encryption: "no" | |
# Address of the TCP/RTSP listener. This is needed only when encryption is "no" or "optional". | |
rtspAddress: :8554 | |
# Address of the TCP/TLS/RTSPS listener. This is needed only when encryption is "strict" or "optional". | |
rtspsAddress: :8322 | |
# Address of the UDP/RTP listener. This is needed only when "udp" is in protocols. | |
rtpAddress: :8000 | |
# Address of the UDP/RTCP listener. This is needed only when "udp" is in protocols. | |
rtcpAddress: :8001 | |
# IP range of all UDP-multicast listeners. This is needed only when "multicast" is in protocols. | |
multicastIPRange: 224.1.0.0/16 | |
# Port of all UDP-multicast/RTP listeners. This is needed only when "multicast" is in protocols. | |
multicastRTPPort: 8002 | |
# Port of all UDP-multicast/RTCP listeners. This is needed only when "multicast" is in protocols. | |
multicastRTCPPort: 8003 | |
# Path to the server key. This is needed only when encryption is "strict" or "optional". | |
# This can be generated with: | |
# openssl genrsa -out server.key 2048 | |
# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650 | |
serverKey: server.key | |
# Path to the server certificate. This is needed only when encryption is "strict" or "optional". | |
serverCert: server.crt | |
# Authentication methods. Available are "basic" and "digest". | |
# "digest" doesn't provide any additional security and is available for compatibility only. | |
rtspAuthMethods: [basic] | |
############################################### | |
# Global settings -> RTMP server | |
# Enable publishing and reading streams with the RTMP protocol. | |
rtmp: no | |
# Address of the RTMP listener. This is needed only when encryption is "no" or "optional". | |
rtmpAddress: :1935 | |
# Encrypt connections with TLS (RTMPS). | |
# Available values are "no", "strict", "optional". | |
rtmpEncryption: "no" | |
# Address of the RTMPS listener. This is needed only when encryption is "strict" or "optional". | |
rtmpsAddress: :1936 | |
# Path to the server key. This is needed only when encryption is "strict" or "optional". | |
# This can be generated with: | |
# openssl genrsa -out server.key 2048 | |
# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650 | |
rtmpServerKey: server.key | |
# Path to the server certificate. This is needed only when encryption is "strict" or "optional". | |
rtmpServerCert: server.crt | |
############################################### | |
# Global settings -> HLS server | |
# Enable reading streams with the HLS protocol. | |
hls: no | |
# Address of the HLS listener. | |
hlsAddress: :8888 | |
# Enable TLS/HTTPS on the HLS server. | |
# This is required for Low-Latency HLS. | |
hlsEncryption: no | |
# Path to the server key. This is needed only when encryption is yes. | |
# This can be generated with: | |
# openssl genrsa -out server.key 2048 | |
# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650 | |
hlsServerKey: server.key | |
# Path to the server certificate. | |
hlsServerCert: server.crt | |
# Value of the Access-Control-Allow-Origin header provided in every HTTP response. | |
# This allows to play the HLS stream from an external website. | |
hlsAllowOrigin: '*' | |
# List of IPs or CIDRs of proxies placed before the HLS server. | |
# If the server receives a request from one of these entries, IP in logs | |
# will be taken from the X-Forwarded-For header. | |
hlsTrustedProxies: [] | |
# By default, HLS is generated only when requested by a user. | |
# This option allows to generate it always, avoiding the delay between request and generation. | |
hlsAlwaysRemux: no | |
# Variant of the HLS protocol to use. Available options are: | |
# * mpegts - uses MPEG-TS segments, for maximum compatibility. | |
# * fmp4 - uses fragmented MP4 segments, more efficient. | |
# * lowLatency - uses Low-Latency HLS. | |
hlsVariant: lowLatency | |
# Number of HLS segments to keep on the server. | |
# Segments allow to seek through the stream. | |
# Their number doesn't influence latency. | |
hlsSegmentCount: 7 | |
# Minimum duration of each segment. | |
# A player usually puts 3 segments in a buffer before reproducing the stream. | |
# The final segment duration is also influenced by the interval between IDR frames, | |
# since the server changes the duration in order to include at least one IDR frame | |
# in each segment. | |
hlsSegmentDuration: 1s | |
# Minimum duration of each part. | |
# A player usually puts 3 parts in a buffer before reproducing the stream. | |
# Parts are used in Low-Latency HLS in place of segments. | |
# Part duration is influenced by the distance between video/audio samples | |
# and is adjusted in order to produce segments with a similar duration. | |
hlsPartDuration: 200ms | |
# Maximum size of each segment. | |
# This prevents RAM exhaustion. | |
hlsSegmentMaxSize: 50M | |
# Directory in which to save segments, instead of keeping them in the RAM. | |
# This decreases performance, since reading from disk is less performant than | |
# reading from RAM, but allows to save RAM. | |
hlsDirectory: '' | |
# The muxer will be closed when there are no | |
# reader requests and this amount of time has passed. | |
hlsMuxerCloseAfter: 60s | |
############################################### | |
# Global settings -> WebRTC server | |
# Enable publishing and reading streams with the WebRTC protocol. | |
webrtc: no | |
# Address of the WebRTC HTTP listener. | |
webrtcAddress: :8889 | |
# Enable TLS/HTTPS on the WebRTC server. | |
webrtcEncryption: no | |
# Path to the server key. | |
# This can be generated with: | |
# openssl genrsa -out server.key 2048 | |
# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650 | |
webrtcServerKey: server.key | |
# Path to the server certificate. | |
webrtcServerCert: server.crt | |
# Value of the Access-Control-Allow-Origin header provided in every HTTP response. | |
# This allows to play the WebRTC stream from an external website. | |
webrtcAllowOrigin: '*' | |
# List of IPs or CIDRs of proxies placed before the WebRTC server. | |
# If the server receives a request from one of these entries, IP in logs | |
# will be taken from the X-Forwarded-For header. | |
webrtcTrustedProxies: [] | |
# Address of a local UDP listener that will receive connections. | |
# Use a blank string to disable. | |
webrtcLocalUDPAddress: :8189 | |
# Address of a local TCP listener that will receive connections. | |
# This is disabled by default since TCP is less efficient than UDP and | |
# introduces a progressive delay when network is congested. | |
webrtcLocalTCPAddress: '' | |
# WebRTC clients need to know the IP of the server. | |
# Gather IPs from interfaces and send them to clients. | |
webrtcIPsFromInterfaces: yes | |
# List of interfaces whose IPs will be sent to clients. | |
# An empty value means to use all available interfaces. | |
webrtcIPsFromInterfacesList: [] | |
# List of additional hosts or IPs to send to clients. | |
webrtcAdditionalHosts: [] | |
# ICE servers. Needed only when local listeners can't be reached by clients. | |
# STUN servers allows to obtain and share the public IP of the server. | |
# TURN/TURNS servers forces all traffic through them. | |
webrtcICEServers2: [] | |
# - url: stun:stun.l.google.com:19302 | |
# if user is "AUTH_SECRET", then authentication is secret based. | |
# the secret must be inserted into the password field. | |
# username: '' | |
# password: '' | |
# clientOnly: false | |
# Time to wait for the WebRTC handshake to complete. | |
webrtcHandshakeTimeout: 10s | |
# Maximum time to gather video tracks. | |
webrtcTrackGatherTimeout: 2s | |
############################################### | |
# Global settings -> SRT server | |
# Enable publishing and reading streams with the SRT protocol. | |
srt: no | |
# Address of the SRT listener. | |
srtAddress: :8890 | |
############################################### | |
# Default path settings | |
# Settings in "pathDefaults" are applied anywhere, | |
# unless they are overridden in "paths". | |
pathDefaults: | |
############################################### | |
# Default path settings -> General | |
# Source of the stream. This can be: | |
# * publisher -> the stream is provided by a RTSP, RTMP, WebRTC or SRT client | |
# * rtsp://existing-url -> the stream is pulled from another RTSP server / camera | |
# * rtsps://existing-url -> the stream is pulled from another RTSP server / camera with RTSPS | |
# * rtmp://existing-url -> the stream is pulled from another RTMP server / camera | |
# * rtmps://existing-url -> the stream is pulled from another RTMP server / camera with RTMPS | |
# * http://existing-url/stream.m3u8 -> the stream is pulled from another HLS server / camera | |
# * https://existing-url/stream.m3u8 -> the stream is pulled from another HLS server / camera with HTTPS | |
# * udp://ip:port -> the stream is pulled with UDP, by listening on the specified IP and port | |
# * srt://existing-url -> the stream is pulled from another SRT server / camera | |
# * whep://existing-url -> the stream is pulled from another WebRTC server / camera | |
# * wheps://existing-url -> the stream is pulled from another WebRTC server / camera with HTTPS | |
# * redirect -> the stream is provided by another path or server | |
# * rpiCamera -> the stream is provided by a Raspberry Pi Camera | |
# If path name is a regular expression, $G1, G2, etc will be replaced | |
# with regular expression groups. | |
source: publisher | |
# If the source is a URL, and the source certificate is self-signed | |
# or invalid, you can provide the fingerprint of the certificate in order to | |
# validate it anyway. It can be obtained by running: | |
# openssl s_client -connect source_ip:source_port </dev/null 2>/dev/null | sed -n '/BEGIN/,/END/p' > server.crt | |
# openssl x509 -in server.crt -noout -fingerprint -sha256 | cut -d "=" -f2 | tr -d ':' | |
sourceFingerprint: | |
# If the source is a URL, it will be pulled only when at least | |
# one reader is connected, saving bandwidth. | |
sourceOnDemand: no | |
# If sourceOnDemand is "yes", readers will be put on hold until the source is | |
# ready or until this amount of time has passed. | |
sourceOnDemandStartTimeout: 10s | |
# If sourceOnDemand is "yes", the source will be closed when there are no | |
# readers connected and this amount of time has passed. | |
sourceOnDemandCloseAfter: 10s | |
# Maximum number of readers. Zero means no limit. | |
maxReaders: 0 | |
# SRT encryption passphrase require to read from this path | |
srtReadPassphrase: | |
# If the stream is not available, redirect readers to this path. | |
# It can be can be a relative path (i.e. /otherstream) or an absolute RTSP URL. | |
fallback: | |
############################################### | |
# Default path settings -> Record | |
# Record streams to disk. | |
record: no | |
# Path of recording segments. | |
# Extension is added automatically. | |
# Available variables are %path (path name), %Y %m %d %H %M %S %f %s (time in strftime format) | |
recordPath: ./recordings/%path/%Y-%m-%d_%H-%M-%S-%f | |
# Format of recorded segments. | |
# Available formats are "fmp4" (fragmented MP4) and "mpegts" (MPEG-TS). | |
recordFormat: fmp4 | |
# fMP4 segments are concatenation of small MP4 files (parts), each with this duration. | |
# MPEG-TS segments are concatenation of 188-bytes packets, flushed to disk with this period. | |
# When a system failure occurs, the last part gets lost. | |
# Therefore, the part duration is equal to the RPO (recovery point objective). | |
recordPartDuration: 1s | |
# Minimum duration of each segment. | |
recordSegmentDuration: 1h | |
# Delete segments after this timespan. | |
# Set to 0s to disable automatic deletion. | |
recordDeleteAfter: 24h | |
############################################### | |
# Default path settings -> Publisher source (when source is "publisher") | |
# Allow another client to disconnect the current publisher and publish in its place. | |
overridePublisher: yes | |
# SRT encryption passphrase required to publish to this path | |
srtPublishPassphrase: | |
############################################### | |
# Default path settings -> RTSP source (when source is a RTSP or a RTSPS URL) | |
# Transport protocol used to pull the stream. available values are "automatic", "udp", "multicast", "tcp". | |
rtspTransport: automatic | |
# Support sources that don't provide server ports or use random server ports. This is a security issue | |
# and must be used only when interacting with sources that require it. | |
rtspAnyPort: no | |
# Range header to send to the source, in order to start streaming from the specified offset. | |
# available values: | |
# * clock: Absolute time | |
# * npt: Normal Play Time | |
# * smpte: SMPTE timestamps relative to the start of the recording | |
rtspRangeType: | |
# Available values: | |
# * clock: UTC ISO 8601 combined date and time string, e.g. 20230812T120000Z | |
# * npt: duration such as "300ms", "1.5m" or "2h45m", valid time units are "ns", "us" (or "µs"), "ms", "s", "m", "h" | |
# * smpte: duration such as "300ms", "1.5m" or "2h45m", valid time units are "ns", "us" (or "µs"), "ms", "s", "m", "h" | |
rtspRangeStart: | |
############################################### | |
# Default path settings -> Redirect source (when source is "redirect") | |
# RTSP URL which clients will be redirected to. | |
sourceRedirect: | |
############################################### | |
# Default path settings -> Raspberry Pi Camera source (when source is "rpiCamera") | |
# ID of the camera | |
rpiCameraCamID: 0 | |
# width of frames | |
rpiCameraWidth: 1920 | |
# height of frames | |
rpiCameraHeight: 1080 | |
# flip horizontally | |
rpiCameraHFlip: false | |
# flip vertically | |
rpiCameraVFlip: false | |
# brightness [-1, 1] | |
rpiCameraBrightness: 0 | |
# contrast [0, 16] | |
rpiCameraContrast: 1 | |
# saturation [0, 16] | |
rpiCameraSaturation: 1 | |
# sharpness [0, 16] | |
rpiCameraSharpness: 1 | |
# exposure mode. | |
# values: normal, short, long, custom | |
rpiCameraExposure: normal | |
# auto-white-balance mode. | |
# values: auto, incandescent, tungsten, fluorescent, indoor, daylight, cloudy, custom | |
rpiCameraAWB: auto | |
# auto-white-balance fixed gains. This can be used in place of rpiCameraAWB. | |
# format: [red,blue] | |
rpiCameraAWBGains: [0, 0] | |
# denoise operating mode. | |
# values: off, cdn_off, cdn_fast, cdn_hq | |
rpiCameraDenoise: "off" | |
# fixed shutter speed, in microseconds. | |
rpiCameraShutter: 0 | |
# metering mode of the AEC/AGC algorithm. | |
# values: centre, spot, matrix, custom | |
rpiCameraMetering: centre | |
# fixed gain | |
rpiCameraGain: 0 | |
# EV compensation of the image [-10, 10] | |
rpiCameraEV: 0 | |
# Region of interest, in format x,y,width,height | |
rpiCameraROI: | |
# whether to enable HDR on Raspberry Camera 3. | |
rpiCameraHDR: false | |
# tuning file | |
rpiCameraTuningFile: | |
# sensor mode, in format [width]:[height]:[bit-depth]:[packing] | |
# bit-depth and packing are optional. | |
rpiCameraMode: | |
# frames per second | |
rpiCameraFPS: 30 | |
# period between IDR frames | |
rpiCameraIDRPeriod: 60 | |
# bitrate | |
rpiCameraBitrate: 1000000 | |
# H264 profile | |
rpiCameraProfile: main | |
# H264 level | |
rpiCameraLevel: '4.1' | |
# Autofocus mode | |
# values: auto, manual, continuous | |
rpiCameraAfMode: continuous | |
# Autofocus range | |
# values: normal, macro, full | |
rpiCameraAfRange: normal | |
# Autofocus speed | |
# values: normal, fast | |
rpiCameraAfSpeed: normal | |
# Lens position (for manual autofocus only), will be set to focus to a specific distance | |
# calculated by the following formula: d = 1 / value | |
# Examples: 0 moves the lens to infinity. | |
# 0.5 moves the lens to focus on objects 2m away. | |
# 2 moves the lens to focus on objects 50cm away. | |
rpiCameraLensPosition: 0.0 | |
# Specifies the autofocus window, in the form x,y,width,height where the coordinates | |
# are given as a proportion of the entire image. | |
rpiCameraAfWindow: | |
# enables printing text on each frame. | |
rpiCameraTextOverlayEnable: false | |
# text that is printed on each frame. | |
# format is the one of the strftime() function. | |
rpiCameraTextOverlay: '%Y-%m-%d %H:%M:%S - MediaMTX' | |
############################################### | |
# Default path settings -> Hooks | |
# Command to run when this path is initialized. | |
# This can be used to publish a stream when the server is launched. | |
# This is terminated with SIGINT when the program closes. | |
# The following environment variables are available: | |
# * MTX_PATH: path name | |
# * RTSP_PORT: RTSP server port | |
# * G1, G2, ...: regular expression groups, if path name is | |
# a regular expression. | |
runOnInit: | |
# Restart the command if it exits. | |
runOnInitRestart: no | |
# Command to run when this path is requested by a reader | |
# and no one is publishing to this path yet. | |
# This can be used to publish a stream on demand. | |
# This is terminated with SIGINT when there are no readers anymore. | |
# The following environment variables are available: | |
# * MTX_PATH: path name | |
# * MTX_QUERY: query parameters (passed by first reader) | |
# * RTSP_PORT: RTSP server port | |
# * G1, G2, ...: regular expression groups, if path name is | |
# a regular expression. | |
runOnDemand: | |
# Restart the command if it exits. | |
runOnDemandRestart: no | |
# Readers will be put on hold until the runOnDemand command starts publishing | |
# or until this amount of time has passed. | |
runOnDemandStartTimeout: 10s | |
# The command will be closed when there are no | |
# readers connected and this amount of time has passed. | |
runOnDemandCloseAfter: 10s | |
# Command to run when there are no readers anymore. | |
# Environment variables are the same of runOnDemand. | |
runOnUnDemand: | |
# Command to run when the stream is ready to be read, whenever it is | |
# published by a client or pulled from a server / camera. | |
# This is terminated with SIGINT when the stream is not ready anymore. | |
# The following environment variables are available: | |
# * MTX_PATH: path name | |
# * MTX_QUERY: query parameters (passed by publisher) | |
# * RTSP_PORT: RTSP server port | |
# * G1, G2, ...: regular expression groups, if path name is | |
# a regular expression. | |
# * MTX_SOURCE_TYPE: source type | |
# * MTX_SOURCE_ID: source ID | |
runOnReady: | |
# Restart the command if it exits. | |
runOnReadyRestart: no | |
# Command to run when the stream is not available anymore. | |
# Environment variables are the same of runOnReady. | |
runOnNotReady: | |
# Command to run when a client starts reading. | |
# This is terminated with SIGINT when a client stops reading. | |
# The following environment variables are available: | |
# * MTX_PATH: path name | |
# * MTX_QUERY: query parameters (passed by reader) | |
# * RTSP_PORT: RTSP server port | |
# * G1, G2, ...: regular expression groups, if path name is | |
# a regular expression. | |
# * MTX_READER_TYPE: reader type | |
# * MTX_READER_ID: reader ID | |
runOnRead: | |
# Restart the command if it exits. | |
runOnReadRestart: no | |
# Command to run when a client stops reading. | |
# Environment variables are the same of runOnRead. | |
runOnUnread: | |
# Command to run when a recording segment is created. | |
# The following environment variables are available: | |
# * MTX_PATH: path name | |
# * RTSP_PORT: RTSP server port | |
# * G1, G2, ...: regular expression groups, if path name is | |
# a regular expression. | |
# * MTX_SEGMENT_PATH: segment file path | |
runOnRecordSegmentCreate: | |
# Command to run when a recording segment is complete. | |
# The following environment variables are available: | |
# * MTX_PATH: path name | |
# * RTSP_PORT: RTSP server port | |
# * G1, G2, ...: regular expression groups, if path name is | |
# a regular expression. | |
# * MTX_SEGMENT_PATH: segment file path | |
# * MTX_SEGMENT_DURATION: segment duration | |
runOnRecordSegmentComplete: | |
############################################### | |
# Path settings | |
# Settings in "paths" are applied to specific paths, and the map key | |
# is the name of the path. | |
# Any setting in "pathDefaults" can be overridden here. | |
# It's possible to use regular expressions by using a tilde as prefix, | |
# for example "~^(test1|test2)$" will match both "test1" and "test2", | |
# for example "~^prefix" will match all paths that start with "prefix". | |
paths: | |
# example: | |
# my_camera: | |
# source: rtsp://my_camera | |
hqstream: | |
runOnInit: python3 /root/picam_stream.py | |
runOnInitRestart: yes | |
# Settings under path "all_others" are applied to all paths that | |
# do not match another entry. | |
all_others: |
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#!/usr/bin/python3 | |
from picamera2 import Picamera2 | |
from picamera2.encoders import H264Encoder, Quality | |
from picamera2.outputs import FfmpegOutput | |
import os, time | |
picam2 = Picamera2() | |
frame_rate = 30 | |
# max resolution is (3280, 2464) for full FoV at 15FPS | |
video_config = picam2.create_video_configuration(main={"size": (1640, 1232), "format": "RGB888"}, | |
lores={"size": (640, 480), "format": "YUV420"}, | |
controls={'FrameRate': frame_rate}) | |
picam2.align_configuration(video_config) | |
picam2.configure(video_config) | |
# FFMPEG output config | |
HQoutput = FfmpegOutput("-f rtsp -rtsp_transport udp rtsp://myuser:mypass@localhost:8554/hqstream", audio=False) | |
LQoutput = FfmpegOutput("-f rtsp -rtsp_transport udp rtsp://myuser:mypass@localhost:8554/lqstream", audio=False) | |
# Encoder settings | |
encoder_HQ = H264Encoder(repeat=True, iperiod=30, framerate=frame_rate, enable_sps_framerate=True) | |
encoder_LQ = H264Encoder(repeat=True, iperiod=30, framerate=frame_rate, enable_sps_framerate=True) | |
try: | |
print("trying to start camera streams") | |
picam2.start_recording(encoder_HQ, HQoutput, quality=Quality.LOW) | |
picam2.start_recording(encoder_LQ, LQoutput, quality=Quality.LOW, name="lores") | |
print("Started camera streams") | |
while True: | |
time.sleep(5) | |
still = picam2.capture_request() | |
still.save("main", "/dev/shm/camera-tmp.jpg") | |
still.release() | |
os.rename('/dev/shm/camera-tmp.jpg', '/dev/shm/camera.jpg') # make image replacement atomic operation | |
except : | |
print("exiting picamera2 streamer") | |
picam2.stop_recording() |
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