How to use go language for audio and video processing and streaming media development

How to use Go language for audio and video processing and streaming media development

Introduction:
With the rapid development of the Internet and the continuous improvement of network bandwidth, the application of audio and video is becoming more and more widespread. As a high-concurrency and high-performance programming language, Go language has gradually attracted the attention of developers. This article will introduce how to use Go language for audio and video processing and streaming media development, including the following: audio and video format processing, audio and video encoding and decoding, audio and video transmission and streaming, construction of streaming media servers, etc.

1. Processing of audio and video formats
In audio and video processing, common audio and video formats include MP3, AAC, WAV, FLV, MP4, etc. The Go language provides some excellent libraries that can easily handle these audio and video formats. The following takes processing MP3 files as an example.

In Go language, we can use the third-party library "github.com/hajimehoshi/go-mp3" to process MP3 files. We first need to install the library:

go get github.com/hajimehoshi/go-mp3/mp3

Next, we use the following code example to read MP3 files and output audio Data:

package main

import (

"fmt"
"github.com/hajimehoshi/go-mp3/mp3"
"github.com/hajimehoshi/oto"
"os"

)

func main() {

file, err := os.Open("test.mp3")
if err != nil {
    fmt.Println("Open file failed:", err)
    return
}
defer file.Close()

decoder, err := mp3.NewDecoder(file)
if err != nil {
    fmt.Println("NewDecoder failed:", err)
    return
}

pcm, err := oto.NewPlayer(decoder.SampleRate(), 2, 2, 8192)
if err != nil {
    fmt.Println("NewPlayer failed:", err)
    return
}
defer pcm.Close()

fmt.Println("Playing...")

buffer := make([]byte, 8192)
for {
    n, err := decoder.Read(buffer)
    if err != nil {
        fmt.Println("Read failed:", err)
        break
    }
    if n == 0 {
        break
    }
    pcm.Write(buffer[:n])
}

fmt.Println("Done.")

}

In the above code, we create an MP3 decoder using the mp3.NewDecoder function, and create an audio player through the oto.NewPlayer function. Then, read the audio data through the Read method, and write the audio data to the player for playback through the Write method.

2. Audio and video encoding and decoding
In audio and video processing, encoding and decoding is a very important part. Go language provides some excellent encoding and decoding libraries, such as ffmpeg, opus, x264, etc. Most of these libraries provide encapsulation of the Go language and are relatively simple to use.

The following takes the ffmpeg library as an example to introduce how to use Go language for audio and video encoding and decoding. First, we need to install the ffmpeg library:

go get github.com/giorgisio/goav/avcodec
go get github.com/giorgisio/goav/avformat

Then, through the following Code example to encode MP3 files into AAC files:

package main

import (

"github.com/giorgisio/goav/avcodec"
"github.com/giorgisio/goav/avformat"
"github.com/giorgisio/goav/avutil"
"os"

)

func main() {

inputFile := "input.mp3"
outputFile := "output.aac"

// 注册所有的编解码器
avcodec.AvcodecRegisterAll()

inputContext := avformat.AvformatAllocContext()
if avformat.AvformatOpenInput(&inputContext, inputFile, nil, nil) < 0 {
    panic("Open input file failed.")
}
defer avformat.AvformatFreeContext(inputContext)

if avformat.AvformatFindStreamInfo(inputContext, nil) < 0 {
    panic("Find stream info failed.")
}

audioStreamIndex := -1
for i := 0; i < len(inputContext.Streams()); i++ {
    if inputContext.Streams()[i].CodecParameters().CodecType() == avutil.AVMEDIA_TYPE_AUDIO {
        audioStreamIndex = i
        break
    }
}

codecParameters := inputContext.Streams()[audioStreamIndex].CodecParameters()
codecId := codecParameters.CodecId()
codec := avcodec.AvcodecFindDecoder(codecId)
if codec == nil {
    panic("Find decoder failed.")
}

codecContext := avcodec.AvcodecAllocContext3(codec)
if codecContext == nil {
    panic("Allocate codec context failed.")
}
defer avcodec.AvcodecFreeContext(codecContext)

if avcodec.AvcodecParametersToContext(codecContext, codecParameters) < 0 {
    panic("Parameters to context failed.")
}

if avcodec.AvcodecOpen2(codecContext, codec, nil) < 0 {
    panic("Open codec failed.")
}
defer avcodec.AvcodecClose(codecContext)

outputFileContext := avformat.AvformatAllocOutputContext2()
if avformat.AvformatAllocOutputContext2(&outputFileContext, nil, "adts", outputFile) < 0 {
    panic("Allocate output context failed.")
}
defer avformat.AvformatFreeContext(outputFileContext)

outputStream := avformat.AvformatNewStream(outputFileContext, nil)
if outputStream == nil {
    panic("New stream failed.")
}

if avcodec.AvcodecParametersFromContext(outputStream.CodecParameters(), codecContext) < 0 {
    panic("Parameters from context failed.")
}

if outputStream.CodecParameters().CodecType() != avutil.AVMEDIA_TYPE_AUDIO {
    panic("Codec type is not audio.")
}

if avformat.AvioOpen(&outputFileContext.Pb(), outputFile, avformat.AVIO_FLAG_WRITE) < 0 {
    panic("Open output file failed.")
}

if avformat.AvformatWriteHeader(outputFileContext, nil) < 0 {
    panic("Write header failed.")
}
defer avformat.AvWriteTrailer(outputFileContext)

packet := avcodec.AvPacketAlloc()
defer avcodec.AvPacketFree(packet)

for avcodec.AvReadFrame(inputContext, packet) >= 0 {
    if packet.StreamIndex() == audioStreamIndex {
        packet.SetPts(packet.Pts() * 2)
        packet.SetDts(packet.Dts() * 2)
        packet.SetDuration(packet.Duration() * 2)
        packet.SetPos(-1)

        if avcodec.AvInterleavedWriteFrame(outputFileContext, packet) < 0 {
            panic("Interleaved write frame failed.")
        }
    }
    avcodec.AvPacketUnref(packet)
}

}

In the above code, we use the ffmpeg library to decode the input MP3 file, then encode the decoded audio data, and write the encoded data to the output file .

3. Audio and video transmission and streaming
Audio and video transmission and streaming are the key to realizing real-time audio and video transmission and streaming media services, and it is also a very complex link. Currently, the most commonly used audio and video transmission protocols are RTMP and HLS. The Go language provides some excellent libraries that can easily implement push and pull streaming of RTMP and HLS protocols.

The following takes the RTMP protocol as an example to introduce how to use Go language for audio and video transmission and streaming. First, we need to install the rtmp library:

go get github.com/gwuhaolin/livego/protocol/rtmp
go get github.com/gwuhaolin/livego/av/codec
go get github. com/gwuhaolin/livego/container

Then, use the following code example to push the camera’s video data to the RTMP server:

package main

import (

"github.com/gwuhaolin/livego/protocol/rtmp"
"github.com/gwuhaolin/livego/av/codec"
"github.com/gwuhaolin/livego/container"
"os"

)

func main() {

inputFile := "/dev/video0"
outputURL := "rtmp://localhost/live/stream"

inputCodec := codec.NewVideoCodec(codec.H264)
outputCodec := codec.NewVideoCodec(codec.H264)

container := container.NewPushContainer(inputFile, inputCodec, outputURL, outputCodec)
container.Push()

}

In the above code, we use the RTPMPusher class provided by the rtmp library to implement the camera's Video data is pushed to RTMP server. Among them, inputFile is the input file (camera device file), and outputURL is the push address.

4. Construction of streaming media server
In streaming media development, the streaming media server is the core component for realizing real-time audio and video transmission and on-demand functions. Currently, commonly used streaming media servers include Nginx-rtmp, FFmpeg, GStreamer, etc.

This section will take Nginx-rtmp as an example to introduce how to use Nginx-rtmp to build a streaming media server. Nginx-rtmp can push audio and video data to the RTMP server, and can also pull audio and video data from the RTMP server.

1. First, we need to install Nginx and Nginx-rtmp module:

wget http://nginx.org/download/nginx-1.18.0.tar.gz
tar zxf nginx-1.18.0.tar.gz
cd nginx-1.18.0
./configure --add-module=/path/to/nginx-rtmp-module
make
make install

2. Modify the Nginx configuration file:

rtmp {

server {
    listen 1935;
    chunk_size 4000;
    application live {
        live on;
        record off;
    }
    application hls {
        live on;
        hls on;
        hls_path /path/to/hls;
        hls_fragment 5s;
        hls_playlist_length 30s;
    }
}

}

In the above configuration, we define There are two applications: live and hls. Among them, the live application is used for real-time audio and video transmission, and the hls application is used for on-demand services.

3. Start Nginx-rtmp service:

/path/to/nginx/sbin/nginx -c /path/to/nginx/conf/nginx.conf

4. Push and play:

Push:
ffmpeg -re -i /path/to/source -c:v copy -c:a copy -f flv rtmp://localhost/live/stream

Play:
ffplay rtmp://localhost/live/stream

Summary:
This article introduces how to use Go language Audio and video processing and streaming media development. By learning audio and video format processing, audio and video encoding and decoding, audio and video transmission and streaming, and the construction of streaming media servers, we can better understand and apply audio and video technology, and implement a variety of rich audio and video applications. I hope this article can be helpful to readers interested in audio and video development.

The above is the detailed content of How to use go language for audio and video processing and streaming media development. For more information, please follow other related articles on the PHP Chinese website!