Files

T

Sergey Krashevich 2b7682cdb3 refactor(dvrip): simplify broadcast loop structure

- replace traditional for loop with range-based for loop for clarity

refactor(ffmpeg): simplify cut function loop
- utilize range-based for loop instead of traditional for loop

refactor(ring): update API response mapping type
- change map type from `interface{}` to `any` for better type safety

refactor(stream): handle nil source in NewStream
- add nil check for source elements before processing

refactor(webrtc): unify payload handling in GetToken
- change map type from `interface{}` to `any` for consistency

refactor(ascii): optimize nested loops in Write function
- replace traditional for loops with range-based for loops for readability

refactor(bits): enhance readability in Reader methods
- replace traditional for loops with range-based for loops in Read functions

refactor(h264): modernize loop structures in DecodeConfig
- switch to range-based for loops for cleaner code

refactor(h265): streamline profile_tier_level loops
- utilize range-based for loops instead of traditional for loops

chore(core): remove commented-out test function for clarity

refactor(core): simplify RandString function loop
- change traditional for loop to range-based for loop

refactor(flvt): optimize timestamp handling in TestTimeToRTP
- switch to range-based for loop for iterating frames

refactor(nest): improve error handling in ExchangeSDP
- format error message with printf-style formatting for clarity

refactor(tapo): enhance securityEncode function
- change traditional for loop to range-based for loop for readability

fix(tcp): correct masking in websocket Write method
- replace traditional for loop with range-based for loop

refactor(tutk): modernize encoding loops in crypto functions
- utilize range-based for loops for better readability

refactor(tuya): unify data types in MQTT message struct
- change map type from `interface{}` to `any` for consistency

refactor(webrtc): standardize codec registration
- change map type from `interface{}` to `any` for type safety

refactor(yaml): simplify Unmarshal function signature
- update parameter type from `interface{}` to `any` for better clarity

2026-03-10 23:26:45 +03:00

device

BIG docs refactoring

2026-02-06 21:45:55 +03:00

hardware

refactor(dvrip): simplify broadcast loop structure

2026-03-10 23:26:45 +03:00

virtual

Update ffmpeg pkg for reading files and parsing ffmpeg version

2024-05-24 11:06:51 +03:00

api.go

Add /api/ffmpeg for playing files and tts on cameras with two-way audio

2024-05-24 15:57:18 +03:00

ffmpeg_test.go

Add #timeout param for ffmpeg source

2025-11-25 16:37:08 -03:00

ffmpeg.go

Code refactoring for #1956

2025-12-27 08:36:40 +03:00

jpeg_test.go

Fix tests

2024-04-30 14:25:48 +03:00

jpeg.go

Rewrite JPEG snapshot consumer

2023-08-30 05:57:00 +03:00

producer.go

Change ffmpeg transcoder from opus to opus/16000

2025-12-14 17:59:07 +03:00

README.md

BIG docs refactoring

2026-02-06 21:45:55 +03:00

version.go

Code refactoring for app module

2024-06-02 07:00:29 +03:00

README.md

FFmpeg

You can get any stream, file or device via FFmpeg and push it to go2rtc. The app will automatically start FFmpeg with the proper arguments when someone starts watching the stream.

FFmpeg preinstalled for Docker and Home Assistant add-on users
Home Assistant add-on users can target files from /media folder

Configuration

Format: ffmpeg:{input}#{param1}#{param2}#{param3}. Examples:

streams:
  # [FILE] all tracks will be copied without transcoding codecs
  file1: ffmpeg:/media/BigBuckBunny.mp4

  # [FILE] video will be transcoded to H264, audio will be skipped
  file2: ffmpeg:/media/BigBuckBunny.mp4#video=h264

  # [FILE] video will be copied, audio will be transcoded to PCMU
  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

  # [MJPEG] video will be transcoded to H264
  mjpeg: ffmpeg:http://185.97.122.128/cgi-bin/faststream.jpg#video=h264

  # [RTSP] video with rotation, should be transcoded, so select H264
  rotate: ffmpeg:rtsp://12345678@192.168.1.123/av_stream/ch0#video=h264#rotate=90

All transcoding formats have built-in templates: h264, h265, opus, pcmu, pcmu/16000, pcmu/48000, pcma, pcma/16000, pcma/48000, aac, aac/16000.

But you can override them via YAML config. You can also add your own formats to the config and use them with source params.

ffmpeg:
  bin: ffmpeg  # path to ffmpeg binary
  global: "-hide_banner"
  timeout: 5  # default timeout in seconds for rtsp inputs
  h264: "-codec:v libx264 -g:v 30 -preset:v superfast -tune:v zerolatency -profile:v main -level:v 4.1"
  mycodec: "-any args that supported by ffmpeg..."
  myinput: "-fflags nobuffer -flags low_delay -timeout {timeout} -i {input}"
  myraw: "-ss 00:00:20"

You can use go2rtc stream name as ffmpeg input (ex. ffmpeg:camera1#video=h264)
You can use video and audio params multiple times (ex. #video=copy#audio=copy#audio=pcmu)
You can use rotate param with 90, 180, 270 or -90 values, important with transcoding (ex. #video=h264#rotate=90)
You can use width and/or height params, important with transcoding (ex. #video=h264#width=1280)
You can use drawtext to add a timestamp (ex. drawtext=x=2:y=2:fontsize=12:fontcolor=white:box=1:boxcolor=black)
- This will greatly increase the CPU of the server, even with hardware acceleration
You can use timeout param to set RTSP input timeout in seconds (ex. #timeout=10)
You can use raw param for any additional FFmpeg arguments (ex. #raw=-vf transpose=1)
You can use input param to override default input template (ex. #input=rtsp/udp will change RTSP transport from TCP to UDP+TCP)
- You can use raw input value (ex. #input=-timeout {timeout} -i {input})
- You can add your own input templates