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Golang audio/video library and streaming server

License: MIT License

Go 99.84% C 0.16%

joy4's Introduction

PLEASE USE joy5 INSTEAD

joy5

High performance Copy-on-write gop cache code
Better av.Packet design code

JOY4

Golang audio/video library and streaming server

JOY4 is powerful library written in golang, well-designed interface makes a few lines of code can do a lot of things such as reading, writing, transcoding among variety media formats, or setting up high-performance live streaming server.

Features

Well-designed and easy-to-use interfaces:

Muxer / Demuxer (doc example)
Audio Decoder (doc example)
Transcoding (doc example)
Streaming server (example)

Support container formats:

MP4
MPEG-TS
FLV
AAC (ADTS)

RTSP Client

High level camera bug tolerance
Support STAP-A

RTMP Client

Support publishing to nginx-rtmp-server
Support playing

RTMP / HTTP-FLV Server

Support publishing clients: OBS / ffmpeg / Flash Player (>8)
Support playing clients: Flash Player 11 / VLC / ffplay / mpv
High performance

Publisher-subscriber packet buffer queue (doc)

Customize publisher buffer time and subscriber read position
Multiple channels live streaming (example)

Packet filters (doc)

Wait first keyframe
Fix timestamp
Make A/V sync
Customize (example)

FFMPEG Golang-style binding (doc)

Audio Encoder / Decoder
Video Decoder
Audio Resampler

Support codec and container parsers:

H264 SPS/PPS/AVCDecoderConfigure parser (doc)
AAC ADTSHeader/MPEG4AudioConfig parser (doc)
MP4 Atoms parser (doc)
FLV AMF0 object parser (doc)

Requirements

Go version >= 1.6

ffmpeg version >= 3.0 (optional)

TODO

HLS / MPEG-DASH Server

ffmpeg.VideoEncoder / ffmpeg.SWScale

License

MIT

joy4's People

Contributors

Stargazers

Watchers

Forkers

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joy4's Issues

Cutting the videos based on start and end time

Define start time and end time in Filter

rtmp: send window ack

calculate FPS

Is it possible to calculate FPS on RTMP stream ?

Wirecast RTMP streaming troubles

Hello, I have an issue when streaming from Wirecast to a simple server using joy4/rtmp (server creates a file using avutil.Create, writes a header based on rtmp.Conn streams, reads from RTMP connection and writes packets to file, on closing of the connection writes trailer).
When encoding in Wirecast is set to 3 - Very fast, or slower, I have a green screen (sometimes) in output file and sound missing.

ffprobe -i output file gives me:

[h264 @ 0x2ceaa20] missing picture in access unit with size 122
[h264 @ 0x2ceaa20] non-existing PPS 1 referenced
[h264 @ 0x2ceaa20] sps_id 32 out of range
[h264 @ 0x2ceaa20] reference count 1 overflow

When using a faster encoding level - everything is fine, no errors with ffprobe.
I checked network connection (by the way it is fast enough not to cause troubles).

When I get RTMP stream from nginx-rtmp server, read it with rtmp.Conn.ReadPacket, put it to a pipe, the read a pipe with ffmpeg and read the ffmpeg output with ts.Demuxer, then write it to file with ts.Muxer, I have the same issues (sound missing and green screen on encoding 3 - Very fast or slower)

When I first get RTMP stream from nginx-rtmp server with ffmpeg, use no encoding (just "copy"), output it to stdout, set ts.Demuxer to read stdout and then use ts.Muxer to write to a file - everything is fine even with slower encoding.

May the problem be in rtmp.Conn component?

panic: flv: h264 seqhdr invalid

启动http_flv_and_rtmp_server 这个例子，执行ffmpeg -re -i movie.flv -c copy -f flv rtmp://localhost/movie，在avutil.CopyPackets(ch.que, conn)处报错：flv: h264 seqhdr invalid 。
请问该如何解决呢？谢谢。

Would be so cool if i can to get some advice how to generate rtmp stream from mp4 files...

Firstly ... Sorry for my English
For more information, you can visit my repo
The main point in my code:
reader.go for reading files and sending av.Packet to channel
writer.go send packets to rtmp server.
On the first file, the server work correctly, but on the next file, the twitch (for example) show a spinner
I thought, this behavior because time in packets of next files is skipped, and I started to change time in packets (reader.go (53- ~ 70)), but this decision is incorrect.
Does somebody have some idea how can I do this?
P.S Also I can`t to work with nginx-rtmp-module. Seems like server close connection.
Nginx docker image

rtsp server

no rtsp server?

golang 是否真的适合做流媒体服务？

golang 在 gc 时会引起卡顿，真的适合做流媒体服务么？请教在是否有在线上验证？

RTSP Client will panic after a long time running

joy4/format/rtsp/client.go

Lines 436 to 443 in bd41a3b

    
           if blocklen, _, ok := self.parseBlockHeader(peek); ok { 
        
           	left := blocklen+4-len(peek) 
        
           	block = append(peek, make([]byte, left)...) 
        
           	if _, err = io.ReadFull(self.brconn, block[len(peek):]); err != nil { 
        
           		return 
        
           	} 
        
           	return 
        
           }

After a long time running (13h+),

joy4/format/rtsp/client.go

Line 438 in bd41a3b

block = append(peek, make([]byte, left)...)

will throw: panic: runtime error: makeslice: len out of range error.

x265 HEVC Support

I wonder if it supports HEVC encoding?

Dead fork?

I see that no PRs have landed since March. I wanted to open a PR, but if the project is no longer being supported, is there a fork that is?

example太少啊

我想知道如何把一个rtsp流抓出来保存为flv

	file, _ := avutil.Create("projectindex.flv")
	conn, _ := rtsp.Dial("rtsp://admin:[email protected]/h264/ch1/sub/av_stream")

	demuxer := &pktque.FilterDemuxer{Demuxer: conn, Filter: &pktque.Walltime{}}
	//demuxer := &transcode.Demuxer{Demuxer: conn,}
	avutil.CopyFile(file, demuxer)

上面代码我乱猜的，不行啊，开启 rtsp.DebugRtp = true 就看见错误，不开debug就闪退：

rtsp: dollar at 11 12
panic: runtime error: invalid memory address or nil pointer dereference

how can i publish packet to other rtmp servers

hello guys, i want to use this library to receive rtmp data from one resource and publish the packet data to other rtmp servers.
i use code like this:

conn, err = rtmp.Dial(url)
conn.WritePacket(packet)

but i got error connection reset by peer, and i'm sure there is no other connections.
Any information would by helpful

Cgo compile error

examples/audio_decode$ go build

github.com/nareix/joy4/cgo/ffmpeg

could not determine kind of name for C.AV_CODEC_FLAG_GLOBAL_HEADER
could not determine kind of name for C.av_frame_alloc
could not determine kind of name for C.av_packet_unref
could not determine kind of name for C.avresample_get_out_samples

How to fix this? Help!

404 not found for multiple channels example

Sorta new to this, just kinda playing around with this for fun. I can't seem to actually get the http_flv_and_rtmp_server example to work properly. When I try to use ffplay I just get a 404 error because the channel doesn't exist. But I can't seem to figure out why

Some questions about rtmp server channels

I read the example code in example > rtmp_server_channels > main.go , but I am really confused about the meaning of delaygop waitkey framedrop delayskip cachegop ,could U add some explanation about this words ? BTW, Thanks for developing such a wonderful streaming lib ~

[Question] is it possible to convert RTMP stream (h264,mp3/aac) to Mpeg1,Mp2 format ?

i know mpeg1 and mp2 are outdated formats , but i have some ancient devices to support,
any advice would be really appreciated !

Broken pipe YouTube with bitrate > 1000

Hi, What kind of information do you need to check this case ?

有打算支持高并发的计划么？

Is it possible to send metadata?

Is there any chance to inject metadata into a RTMP stream? I'm a bit confused using the Filter functionality.

Get the one frame from the stream

Hello!

Thank you for awesome library!

I have a trivial question, maybe. I want to write very easy monitoring tool for IP-cameras (above 50 devices). The main criteria of avalability of devices is an ability of getting an image. I mean that I need more than simple check by ICMP or TCP/UDP. In this case I want to fetch (programmly) of the frame and checking that this image is non-empty, e.g. Can I do that using of your library?
I found sources of RTSP-client software but I don't know how fetch the frame.

Is this package able to playback audio?

Failed to start http_flv_and_rtmp_server

This is very good open source library for streaming.
I tried to run as given below but error thrown

/go/src/github.com/nareix/joy4/examples$ go run http_flv_and_rtmp_server/main.go

github.com/nareix/joy4/format/flv

*../format/flv/flv.go:416: self.bufr.Discard undefined (type bufio.Reader has no field or method Discard)

Could you please help me out on this issue.

example 中http_flv_and_rtmp_server根本编译不过

ch.que = pubsub.NewQueue( streams )
ch.que.SetMaxDuration(time.Minute)
这两句，NewQueue没有参数
que没有SetMaxDuration方法

h264 to mp4

I'm sorry for simply posting an issue instead of looking more deeply into the library.

But basically I have a server that is receiving a RAW h264 stream from a raspberry. I'm unable to send the h264 stream directly to a webpage, form what I've read it is much easier to stream it in MP4.

So my question is: Can I use this library to encapsulate my raw h264 stream in MP4 on the fly?

No RTSP client example file

Can you please provide some example file that reads RTSP video and then write the data to some other file?Currently there is no example file relating to RTSP.

format有加rtp的计划吗

RTSP digest authentication not working

It may be specific to the cameras I'm using, but RTSP digest authentication doesn't work unless I comment out this line with basic auth:

joy4/format/rtsp/client.go

Line 354 in 4b82a8c

    
           fmt.Sprintf(`Authorization: Basic %s`, base64.StdEncoding.EncodeToString([]byte(username+":"+password))),

Maybe instead of appending to headers on line 360 it should assign.

Close() on RTMP server

Right now there is no way to close the RTMP server manually. The Golang http package is an example to follow here - https://golang.org/pkg/net/http/#Server.Close

@nareix I'm happy to make a PR - keep a reference to the listener in rtmp.Server{}, and add a Close() function to call listener.Close(). Does that sound reasonable?

Not supported user defined metadata

If M2TS has user defined metadata, then joy4 can't parse it

Rtsp Server

Feature request: possibility to demux UDP MPEG-TS packets

I'm trying to demux MPEG-TS over UDP.

Code is something like that:

addr, err := net.ResolveUDPAddr("udp", "239.1.2.3:1234")
if err != nil {
	log.Fatalln(err)
}

conn, err := net.ListenMulticastUDP("udp", nil, addr)
if err != nil {
	log.Fatalln(err)
}

demux := ts.NewDemuxer(conn)
streams, err := demux.Streams()
if err != nil {
	fmt.Printf("error getting streams: %s\n", err)
	os.Exit(1)
}

It doesn't seem to work correctly.
I've looked through the formats/ts and formats/ts/tsio packages: the clue is, that TSReadPacket wants to have the pure sequence of MPEG-TS packets in the stream.

But when you join multicast stream, you never know which part of it you have started to receive. I mean, you can join in the middle of the packet. In this case, TS parser should read the stream until it encounters a valid beginning of the new MPEG-TS packet.

I could try to implement it by myself, but some piece of advice from the author how to do it properly will be really appreciated.

any plan add rtsp server?

RTMP Error: h264 seqhdr invalid

Using ffmpeg 3.3, ffmpeg -f avfoundation -framerate 30 -i "0:0" -video_size vga -f flv "rtmp://localhost:1935/stream/live"

Got error: flv: h264 seqhdr invalid. This happens during RTMP handshake.

example usage

HLS support

Hi!

First of all, Great work! awesome!

I'm currently just starting with joy4 but loving it this far. But there is this thing:

Is there any estimates on HLS support? Maybe beta to test out?

It is very needed.

publish to yotube

publish to yotube not working , but when i copy rtmp stream to other service all fine. It says remote host closed connection

avutil.CopyFile

If the function avutil.CopyFile is an analog of a io.Copy, then it should return nil if all or and not EOF

fMp4

Wondering if you could point me in the right direction. If I have a channel of packets (MPEG-TS, AAC) how do I mux that into a fMp4 stream suitable for MSE.

Thanks for open sourcing this library. Looks much easier to use than goav for gmf

forward input stream to multiple servers(push )

Hi again!, could you please point out what i'm doing wrong?

package main

import (
    "log"
    "strings"
    "sync"
    "time"

    "github.com/nareix/joy4/av/avutil"
    "github.com/nareix/joy4/av/pktque"
    "github.com/nareix/joy4/av/pubsub"
    "github.com/nareix/joy4/format"
    "github.com/nareix/joy4/format/rtmp"
)

func init() {
    format.RegisterAll()
}

var sm = map[string][]string{
    "azorestv": []string{"rtmp://opinion.azorestv.com:1936/proxy/one", "rtmp://opinion.azorestv.com:1936/proxy/two"},
}

func main() {
    proxyServer()
}

func proxyServer() {
    server := &rtmp.Server{}

    server.HandlePublish = func(conn *rtmp.Conn) {

        segs := strings.Split(conn.URL.Path, "/")
        key := segs[2]
        streamMap, ok := sm[key]
        if !ok {
            log.Println("invalid key")
            return
        }

        que := &pubsub.Queue{}
        que.Close()

        streams, err := conn.Streams()
        if err != nil {
            log.Println(err)
            return
        }

        que = pubsub.NewQueue(streams)
        que.SetMaxDuration(time.Second * 10)

        go func() {
            err := avutil.CopyPackets(que, conn)
            if err != nil {
                log.Println(err)
            }
        }()

        wait := &sync.WaitGroup{}

        for i := range streamMap {
            go func() {
                err := pushStream(que, streamMap[i], wait)
                if err != nil {
                    log.Println(err)
                }

            }()
        }
        wait.Wait()

    }

    server.ListenAndServe()

}

func pushStream(sub *pubsub.Queue, dsturl string, wait *sync.WaitGroup) error {

    defer wait.Done()
    wait.Add(1)

    origin := sub.Latest()
    filters := pktque.Filters{}

    filters = append(filters, &pktque.FixTime{StartFromZero: true, MakeIncrement: true})
    demuxer := &pktque.FilterDemuxer{
        Filter:  filters,
        Demuxer: origin,
    }

    dst, err := rtmp.Dial(dsturl)
    if err != nil {
        return err
    }
    defer dst.Close()

    return avutil.CopyFile(dst, demuxer)
}

thanks again!

example error

root@ubuntu-HP-ProDesk-680-G2-MT:/usr/local/lib# ffmpeg
ffmpeg version 3.1.11 Copyright (c) 2000-2017 the FFmpeg developers
built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.9) 20160609
configuration: --prefix=/usr/local --enable-shared
libavutil 55. 28.100 / 55. 78.100
libavcodec 57. 48.101 / 57.107.100
libavformat 57. 41.100 / 57. 83.100
libavdevice 57. 0.101 / 57. 10.100
libavfilter 6. 47.100 / 6.107.100
libswscale 4. 1.100 / 4. 8.100
libswresample 2. 1.100 / 2. 9.100
Hyper fast Audio and Video encoder
usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}...

root@ubuntu-HP-ProDesk-680-G2-MT:~/go/src/github.com/nareix/joy4/examples/audio_decode# go run main.go
/# github.com/nareix/joy4/cgo/ffmpeg
../../cgo/ffmpeg/audio.go: In function ‘wrap_avcodec_decode_audio4’:
../../cgo/ffmpeg/audio.go:7:2: warning: ‘avcodec_decode_audio4’ is deprecated [-Wdeprecated-declarations]
return avcodec_decode_audio4(ctx, frame, got, &pkt);
^
In file included from /usr/local/include/libavformat/avformat.h:317:0,
from ../../cgo/ffmpeg/ffmpeg.h:2,
from ../../cgo/ffmpeg/audio.go:4:
/usr/local/include/libavcodec/avcodec.h:4714:5: note: declared here
int avcodec_decode_audio4(AVCodecContext *avctx, AVFrame *frame,
^
../../cgo/ffmpeg/audio.go: In function ‘_cgo_77bb8e1dcb21_Cfunc_avcodec_encode_audio2’:
../../cgo/ffmpeg/audio.go:124:2: warning: ‘avcodec_encode_audio2’ is deprecated [-Wdeprecated-declarations]
}
^
In file included from /usr/local/include/libavformat/avformat.h:317:0,
from ../../cgo/ffmpeg/ffmpeg.h:2,
from ../../cgo/ffmpeg/audio.go:4:
/usr/local/include/libavcodec/avcodec.h:5232:5: note: declared here
int avcodec_encode_audio2(AVCodecContext *avctx, AVPacket *avpkt,
^
/# github.com/nareix/joy4/cgo/ffmpeg
../../cgo/ffmpeg/video.go: In function ‘wrap_avcodec_decode_video2’:
../../cgo/ffmpeg/video.go:7:2: warning: ‘avcodec_decode_video2’ is deprecated [-Wdeprecated-declarations]
return avcodec_decode_video2(ctx, frame, got, &pkt);
^
In file included from /usr/local/include/libavformat/avformat.h:317:0,
from ../../cgo/ffmpeg/ffmpeg.h:2,
from ../../cgo/ffmpeg/video.go:4:
/usr/local/include/libavcodec/avcodec.h:4763:5: note: declared here
int avcodec_decode_video2(AVCodecContext *avctx, AVFrame *picture,
^
panic: runtime error: index out of range

goroutine 1 [running]:
panic(0x6666a0, 0xc82000a0c0)
/usr/lib/go-1.6/src/runtime/panic.go:481 +0x3e6
main.main()
/home/ubuntu/go/src/github.com/nareix/joy4/examples/audio_decode/main.go:30 +0x3ae
exit status 2

Unable to proxy rtmp (EOF)

test server rtmp://opinion.azorestv.com/clients/57040dd71756e561cece4900

关于视频转码问题

我在运用该项目进行视频转码的时候，发现只有音频编码解码，而缺少视频编码解码，不知道是否我看没找到，还是暂时不支持？（感谢该项目）

Using CFLAGS to change library path

I am trying to build joy4 with custom ffmpeg library files.
I've added:
#cgo CFLAGS: -I/path/to/include/
in ffmpeg.go file and I am getting a file not found error:
./ffmpeg.h:1:10: fatal error: 'libavutil/samplefmt.h' file not found

The files are all there - these are ffmpeg pre-build library for ARM processors.
What am I missing?

ffmpeg bindings error

# bitbucket.org/andboson/cams-back/vendor/github.com/nareix/joy4/cgo/ffmpeg vendor/github.com/nareix/joy4/cgo/ffmpeg/audio.go:298: cannot use C.int64_t(self.Bitrate) (type C.int64_t) as type C.int in assignment

when i run

/github.com/nareix/joy4/examples/transcode/main.go

Ubuntu 16.04 x64

TS file generated contains invalid aac sample rate

I am generating TS files, and the aac track sample rate is not stored correctly. The sample rate is set to 48000, 1 channel, AOT_AAC_LC.

	mPEG4AudioConfig := aacparser.MPEG4AudioConfig{
		SampleRate:      48000,
		SampleRateIndex: 0, // if 0 will be calculated
		ChannelConfig: 0,                  // if 0 it will be calculated
		ChannelLayout: av.CH_FRONT_CENTER,
		ObjectType: aacparser.AOT_AAC_LC,
	}

ffprobe version 4.0 Copyright (c) 2007-2018 the FFmpeg developers
  built with gcc 6.3.0 (GCC)
  configuration: --prefix=/home/alexj/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/alexj/ffmpeg_build/include --extra-ldflags=-L/home/alexj/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/alexj/bin --enable-gpl --enable-libaom --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-nonfree
  libavutil      56. 14.100 / 56. 14.100
  libavcodec     58. 18.100 / 58. 18.100
  libavformat    58. 12.100 / 58. 12.100
  libavdevice    58.  3.100 / 58.  3.100
  libavfilter     7. 16.100 /  7. 16.100
  libswscale      5.  1.100 /  5.  1.100
  libswresample   3.  1.100 /  3.  1.100
  libpostproc    55.  1.100 / 55.  1.100
[aac @ 0x4b2fc40] Multiple frames in a packet.
[aac @ 0x4b2fc40] channel element 2.6 is not allocated
    Last message repeated 166 times
[mpegts @ 0x4b29ec0] decoding for stream 1 failed
[mpegts @ 0x4b29ec0] Could not find codec parameters for stream 1 (Audio: aac (LC) ([15][0][0][0] / 0x000F), mono, fltp, 52 kb/s): unspecified sample rate
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, mpegts, from 'tmp.ts':
  Duration: 00:00:03.57, start: 56548.874933, bitrate: 266 kb/s
  Program 1 
    Stream #0:0[0x100]: Video: h264 (Baseline) ([27][0][0][0] / 0x001B), yuv420p(progressive), 360x640, 90k tbr, 90k tbn, 180k tbc
    Stream #0:1[0x101]: Audio: aac (LC) ([15][0][0][0] / 0x000F), mono, fltp, 52 kb/s

	if blocklen, _, ok := self.parseBlockHeader(peek); ok {
	left := blocklen+4-len(peek)
	block = append(peek, make([]byte, left)...)
	if _, err = io.ReadFull(self.brconn, block[len(peek):]); err != nil {
	return
	}
	return
	}

nareix / joy4 Goto Github PK

joy4's Introduction

PLEASE USE joy5 INSTEAD

JOY4

Features

Requirements

TODO

License

joy4's People

Contributors

Stargazers

Watchers

Forkers

joy4's Issues

github.com/nareix/joy4/cgo/ffmpeg

github.com/nareix/joy4/format/flv

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