yutopp / go-rtmp

RTMP 1.0 server/client library written in Go.

go-rtmp

RTMP 1.0 server/client library written in Go.

Work in progress

Installation

go get github.com/yutopp/go-rtmp

See also server_demo and client_demo.

Documentation

• GoDoc
• REAL-TIME MESSAGING PROTOCOL (RTMP) SPECIFICATION

NOTES

How to limit bitrates or set timeouts

• Please use yutopp/go-iowrap.

License

Boost Software License - Version 1.0

go-rtmp's Issues

Change the Handler interface to reduce callbacks which have increased

Add OnError to handler

is it possible to publish audio data every 0.5 second?

original code of rtmp OnAudio function like this,

func (h *Handler) OnAudio(timestamp uint32, payload io.Reader) error {
	var audio flvtag.AudioData
	if err := flvtag.DecodeAudioData(payload, &audio); err != nil {
		return err
	}

	flvBody := new(bytes.Buffer)
	if _, err := io.Copy(flvBody, audio.Data); err != nil {
		return err
	}
	audio.Data = flvBody
	_ = h.pub.Publish(&flvtag.FlvTag{
		TagType:   flvtag.TagTypeAudio,
		Timestamp: timestamp,
		Data:      &audio,
	})

	return nil
}

and i want to chage the code that publish audio data every 0.5 second
so trying like this

func (h *Handler) OnAudio(timestamp uint32, payload io.Reader) error {

	var audio flvtag.AudioData
	if err := flvtag.DecodeAudioData(payload, &audio); err != nil {
		return err
	}
       // if flag == true, startTimeStamp Control
	if h.Flag == true {
		h.startTimeStamp = timestamp
		//h.Buf = new(bytes.Buffer)
		h.Buf.Reset()
	}
	h.Flag = false

	buf := new(bytes.Buffer)
	if _, err := io.Copy(buf, audio.Data); err != nil {
		return err
	}

	if audio.AACPacketType == 0 {
		audio.Data = buf
		tag := &flvtag.FlvTag{
			TagType:   flvtag.TagTypeAudio,
			Timestamp: h.startTimeStamp,
			Data:      &audio,
		}
		timestamp, (timestamp - h.startTimeStamp), buf.Len())
		_ = h.pub.Publish(tag)
		h.Flag = true
		return nil
	}

	h.Buf.Write(buf.Bytes())

        // collect audio data into h.Buf and then publish every 0.5 seconds
	if timestamp-h.startTimeStamp >= 500 {
		h.lastTimeStamp = timestamp
		audio.Data = h.Buf
		tag := &flvtag.FlvTag{
			TagType:   flvtag.TagTypeAudio,
			Timestamp: h.startTimeStamp,
			Data:      &audio,
		}

		_ = h.pub.Publish(tag)
		h.Flag = true
	}

	return nil
}

when i check the audio data size There is no Data loss but Audio Data is Broken.
Is there any problem with my code?

Support all fmt

can the stream be converted to hls ?

the stream keeps increasing the flv file size other than that hls is capable for transferring media on lots of platforms

should we convert the flv stream to hls or is there a way to implement it in the code

Support ext timestamp

Fix timing problems for Server.Serve and Close

Calling Server.Close method before the listener registration in Server.Serve cause a timing problem which failed to call listener.Close.

Add tests for handshake logics

Rtmp Server to save flv file

I want to continuously save the recording from rtmp
I want to change the filename dynamically according to PublishingName periodically.
Can i do that

Limit output size of UnknownDataBodyDecodeError and UnknownCmdBodyDecodeError

Error() function may generate a large string when there are many elements in Objs.

Is it possible to determine the FPS?

Is it possible to determine the FPS of the incoming RTMP stream?

Is this still maintained?

How to get started with using this library?

Is there any documentation I can refer to?

support handshake with digest

How do take path meaasge

For example：rtmp://video-center.alivecdn.com/record/3518?vhost=rec.livecourse.com"

path is ： /record/3518?vhost=rec.livecourse.com

stream_id always be 1

Support timeout

Support play command completely

add OnUnpublish Handler

How to implement Play within current architecture

Great Go code first of all!

However, it is hard to implement Play within current code architecture. The server is basically setup to be a passive handler. For example, after a rtmp player sends out play command and gets back the play succeed response. It sits there and waits for server to send video and audio chunks. But in server_data_play_handler.go, it is still waiting for message send from client. How could we write a play function that fits nicely in current architecture?

Use %T instead of reflect.TypeOf

Add client message handlers

BUG? not compatible with https://github.com/ossrs/srs

ffmpeg -> srs -> go-rtmp server demo

failed with handshake

srs:

Support FCPublish, FCUnpublish

Support deleteStream(removeStream?)

Revert: hander interfaces

Currently, handler provides methods OnCommand, OnData, etc, and a user uses a type switch to assign logic per commands/data.
This method has an advantage to reduce methods of the interface, however it is too dynamic.

Maybe it is better to provide methods of all commands/data as a interface, and a default implementation of the interface, then let users use it by mixin.

[help] How to get "App" in "On*" hooks?

Hi!

Thanks a lot for your work!

My use cases:

1. I want to publish via ffmpeg ffmpeg -i file.mp4 -c copy -f flv rtmp://0.0.0.0:1935/xxx/yyy
2. I want to read from VLC rtmp://0.0.0.0:1935/xxx/yyy
3. I want to read from HTTP http://0.0.0.0:8080/xxx/yyy

---

My questions:

1. Do I need to create a buffer per app?
2. How to access to App in OnPlay or OnPublish hooks of handler?

Stream to YouTube

Hi
Before i dig in, is it possible to use this lib to Push and h264 Stream to YouTube. The example for Client is Limited some advice would be great.

Thx

What is the best way for me to take the stream and push it to AWS S3?

I am successfully receiving stream on my server from OBS studio. I am poking my head around but I figure someone should have the answer. How to I take stream and throw it up to AWS S3?

Add a RTMP client interface