Golang and FFmpeg: How to implement audio synthesis and variable speed

Golang and FFmpeg: How to implement audio synthesis and speed change, specific code examples are needed

Audio synthesis and speed change are one of the common requirements in audio processing, and Golang is one of the common requirements in audio processing. A powerful programming language, combined with the FFmpeg tool, can easily implement these functions. This article will introduce how to use Golang and FFmpeg to implement audio synthesis and speed change, and give specific code examples.

1. Installing FFmpeg

First, we need to install the FFmpeg tool. Execute the following command in the terminal to install FFmpeg:

sudo apt-get install ffmpeg

2. Introduce the GoFFmpeg library

GoFFmpeg is a Golang library that provides an encapsulation of FFmpeg functions, we can use it To achieve audio synthesis and speed changing. In the Go project, execute the following command to introduce the GoFFmpeg library:

go get -u github.com/goodiebag/go-libav

3. Audio synthesis

Audio synthesis is to merge two or more audio files into one audio file the process of. The following is a simple example that demonstrates how to use the GoFFmpeg library to implement audio synthesis:

package main

import (
    "fmt"

    "github.com/goodiebag/go-libav/avcodec"
    "github.com/goodiebag/go-libav/avformat"
    "github.com/goodiebag/go-libav/avutil"
)

func main() {
    formatContext1 := avformat.AvformatAllocContext() // 创建AVFormatContext对象
    formatContext2 := avformat.AvformatAllocContext()

    filename1 := "audio1.mp3" // 第一个音频文件的文件名
    filename2 := "audio2.mp3" // 第二个音频文件的文件名
    outputFilename := "output.mp3" // 合成音频的输出文件名

    avformat.AvformatOpenInput(&formatContext1, filename1, nil, nil) // 打开第一个音频文件
    avformat.AvformatOpenInput(&formatContext2, filename2, nil, nil) // 打开第二个音频文件

    avformat.AvformatFindStreamInfo(formatContext1, nil) // 获取第一个音频文件的流信息
    avformat.AvformatFindStreamInfo(formatContext2, nil) // 获取第二个音频文件的流信息

    stream1 := formatContext1.Streams()[0] // 获取第一个音频文件的流
    stream2 := formatContext2.Streams()[0] // 获取第二个音频文件的流

    formatContextOut := avformat.AvformatAllocContext() // 创建输出格式的AVFormatContext对象
    avformat.AvformatAllocOutputContext2(&formatContextOut, nil, nil, outputFilename) // 创建输出格式的AVFormatContext对象

    outputStream := avutil.AvformatNewStream(formatContextOut, nil) // 创建输出流
    outputStream.SetCodecParameters(stream1.CodecParameters()) // 设置输出流的编解码参数

    if err := formatContextOut.WriteHeader(nil); err != nil { // 写入文件头
        fmt.Println("Error writing header:", err)
        return
    }

    packet := avcodec.AvPacketAlloc()

    for {
        if ret := avformat.AvReadFrame(formatContext1, packet); ret < 0 { // 读取第一个音频文件的音频帧
            break
        }

        packet.SetStreamIndex(outputStream.Index()) // 设置音频帧的流索引
        if err := avformat.AvInterleavedWriteFrame(formatContextOut, packet); err != nil { // 写入音频帧
            fmt.Println("Error writing frame:", err)
            break
        }

        avformat.AvPacketUnref(packet)
    }

    for {
        if ret := avformat.AvReadFrame(formatContext2, packet); ret < 0 { // 读取第二个音频文件的音频帧
            break
        }

        packet.SetStreamIndex(outputStream.Index()) // 设置音频帧的流索引
        if err := avformat.AvInterleavedWriteFrame(formatContextOut, packet); err != nil { // 写入音频帧
            fmt.Println("Error writing frame:", err)
            break
        }

        avformat.AvPacketUnref(packet)
    }

    if err := avformat.AvWriteTrailer(formatContextOut); err != nil { // 写入文件尾
        fmt.Println("Error writing trailer:", err)
        return
    }

    fmt.Println("Audio files merged successfully!")
}

This code first creates two AVFormatContext objects, which are used to open two audio files to be synthesized. Then, obtain the audio stream information through the AvformatFindStreamInfo function. After that, create a new AVFormatContext object for managing synthesized audio files. In this new AVFormatContext object, create a new output stream and set the corresponding codec parameters.

Then, we enter a loop to read the audio frames of the first audio file and write them to the output stream. Then, enter a loop again to read the audio frames of the second audio file and write them to the output stream. Finally, the end of the file is written to complete the audio synthesis.

4. Audio speed changing

Audio speed changing is the process of changing the audio playback speed. The following is a simple example that demonstrates how to use the GoFFmpeg library to implement audio speed changing:

package main

import (
    "fmt"

    "github.com/goodiebag/go-libav/avcodec"
    "github.com/goodiebag/go-libav/avformat"
    "github.com/goodiebag/go-libav/avutil"
)

func main() {
    formatContext := avformat.AvformatAllocContext() // 创建AVFormatContext对象

    filename := "input.mp3" // 需要变速的音频文件的文件名
    outputFilename := "output.mp3" // 变速后的音频文件的输出文件名

    if err := avformat.AvformatOpenInput(&formatContext, filename, nil, nil); err != nil { // 打开音频文件
        fmt.Println("Error opening input:", err)
        return
    }

    if err := avformat.AvformatFindStreamInfo(formatContext, nil); err != nil { // 获取音频流信息
        fmt.Println("Error finding stream info:", err)
        return
    }

    stream := formatContext.Streams()[0] // 获取音频流

    formatContextOut := avformat.AvformatAllocContext() // 创建输出格式的AVFormatContext对象
    avformat.AvformatAllocOutputContext2(&formatContextOut, nil, nil, outputFilename) // 创建输出格式的AVFormatContext对象

    outputStream := avutil.AvformatNewStream(formatContextOut, nil) // 创建输出流
    outputStream.SetCodecParameters(stream.CodecParameters()) // 设置输出流的编解码参数

    if err := formatContextOut.WriteHeader(nil); err != nil { // 写入文件头
        fmt.Println("Error writing header:", err)
        return
    }

    ptsDelta := avcodec.AvRescaleDelta(1, 2, stream.R(TB().Den*1000), stream.TimeBase()) // 设置时间戳间隔

    packet := avcodec.AvPacketAlloc()

    for {
        if ret := avformat.AvReadFrame(formatContext, packet); ret < 0 { // 读取音频帧
            break
        }

        packet.PointsInTwo(&ptsDelta) // 变速

        packet.SetStreamIndex(outputStream.Index()) // 设置音频帧的流索引
        if err := avformat.AvInterleavedWriteFrame(formatContextOut, packet); err != nil { // 写入音频帧
            fmt.Println("Error writing frame:", err)
            break
        }

        avformat.AvPacketUnref(packet)
    }

    if err := avformat.AvWriteTrailer(formatContextOut); err != nil { // 写入文件尾
        fmt.Println("Error writing trailer:", err)
        return
    }

    fmt.Println("Audio speed changed successfully!")
}

This code is similar to the previous audio synthesis example. It first opens the audio file, obtains the audio stream information, and creates a new AVFormatContext object. Then, create a new output stream and set the corresponding codec parameters.

Then, we enter a loop, read the audio frame, and use the AvRescaleDelta function to perform variable speed processing of the timestamp. Then, the variable-speed audio frames are written to the output stream. Finally, the end of the file is written to complete the audio speed change.

Summary

Through the introduction of this article, we have learned how to use Golang and FFmpeg to achieve audio synthesis and speed change. Through the encapsulation of the GoFFmpeg library, we can easily use the FFmpeg tool in Golang to process audio. I hope this article will be helpful to you and you can successfully implement the functions of audio synthesis and speed change.

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