Go语言调用ffmpeg-api实现音频重采样

目录

• 前言
• 环境部署
• 代码
• 总结

前言

最近对golang处理音视频很感兴趣，对golang音视频常用库goav进行了一番研究。自己写了一个wav转采样率的功能。给大家分享一下，中间遇到了不少坑，解决的过程中还是蛮有意思的。

环境部署

代码运行在Ubuntu环境上，需要使用到goav，goav是对ffmpeg源码的golang封装。

goav地址：https://github.com/giorgisio/goav

goav安装如下

sudo apt-get -y install autoconf automake build-essential libass-dev libfreetype6-dev libsdl1.2-dev libtheora-dev libtool libva-dev libvdpau-dev libvorbis-dev libxcb1-dev libxcb-shm0-dev libxcb-xfixes0-dev pkg-config texi2html zlib1g-dev
 
sudo apt install -y libavdevice-dev libavfilter-dev libswscale-dev libavcodec-dev libavformat-dev libswresample-dev libavutil-dev
 
sudo apt-get install yasm
 
export FFMPEG_ROOT=$HOME/ffmpeg
export CGO_LDFLAGS="-L$FFMPEG_ROOT/lib/ -lavcodec -lavformat -lavutil -lswscale -lswresample -lavdevice -lavfilter"
export CGO_CFLAGS="-I$FFMPEG_ROOT/include"
export LD_LIBRARY_PATH=$HOME/ffmpeg/lib
``` 
 
``` 
go get github.com/xueqing/goav

代码

先看代码

package main //#include<stdlib.h> import "C" import ( "flag" "fmt" "github.com/google/logger" "github.com/xueqing/ffmpeg-demo/logutil" "github.com/xueqing/goav/libswresample" "github.com/youpy/go-wav" "io" "os" "reflect" "unsafe" ) func main() { var ( inputUrl string = "./data/1.wav" inNumChannels int64 = 1 inSampleRate int = 16000 //inBitsPerSample uint16 = 16 outNumChannels int64 = 1 outSampleRate int = 48000 outBitsPerSample uint16 = 16 swr *libswresample.SwrContext = libswresample.SwrAlloc() ) flag.Parse() logutil.Init(true, false, "resample.log") defer logutil.Close() swr.SwrAllocSetOpts(outNumChannels, libswresample.AvSampleFormat(1), outSampleRate, inNumChannels, libswresample.AvSampleFormat(1), inSampleRate, 0, 0) swr.SwrInit() defer swr.SwrClose() _inputFile, err := os.Open(inputUrl) if err != nil { logger.Errorf("open input file error(%v)", err) return } defer _inputFile.Close() _reader := wav.NewReader(_inputFile) format, err := _reader.Format() if err != nil { logger.Errorf("input file format error(%v)", err) return } fmt.Printf("input file format info -> AudioFormat:%v,NumChannels:%v,SampleRate:%v,ByteRate:%v,BlockAlign:%v,BitsPerSample:%v", int(format.AudioFormat), format.NumChannels, format.SampleRate, format.ByteRate, format.BlockAlign, format.BitsPerSample) _tempFile, err := os.CreateTemp("", "*.wav") if err != nil { logger.Errorf("create temp file error(%v)", err) return } logger.Infof("Create tempFile %v", _tempFile.Name()) defer func() { _tempFile.Close() }() _samples := []wav.Sample{} n := 4096 for { spls, err := _reader.ReadSamples(uint32(n)) if err == io.EOF { break } _samples = append(_samples, spls...) } _result := ResampleByFFmpegApi2(swr, _samples) _writer := wav.NewWriter(_tempFile, uint32(len(_result)), uint16(outNumChannels), uint32(outSampleRate), outBitsPerSample) err4 := _writer.WriteSamples(_result) if err4 != nil { logger.Errorf("write file error(%v)", err4) err = err4 return } } func ResampleByFFmpegApi2(swr *libswresample.SwrContext, samples []wav.Sample) []wav.Sample { var ( _inArr uint8 _outArr uint8 _inptr []uint16 _outptr []uint16 ) _inArr = (uint8)(C.malloc(C.sizeof_int)) defer C.free(unsafe.Pointer(_inArr)) _inptr = make([]uint16, len(samples)) _outArr = (uint8)(C.malloc(C.sizeof_int)) defer C.free(unsafe.Pointer(_outArr)) _outptr = make([]uint16, len(samples)*3) //fmt.Println(unsafe.Sizeof(uint16(0))) for i, v := range samples { _inptr[i] = uint16(v.Values[0]) } *_inArr = (*uint8)(unsafe.Pointer((*reflect.SliceHeader)(unsafe.Pointer(&_inptr)).Data)) *_outArr = (*uint8)(unsafe.Pointer((*reflect.SliceHeader)(unsafe.Pointer(&_outptr)).Data)) ret := swr.SwrConvert(_outArr, len(samples)*3, _inArr, len(samples)) if ret > 0 { fmt.Println(ret) } _result := make([]wav.Sample, ret) for i := 0; i < ret; i++ { _result[i] = wav.Sample{[2]int{int(_outptr[i]), 0}} } return _result }

代码说明：

1、代码不是个工具方法，如果看懂逻辑的话，可以自行修改成工具方法。

2、里面会用到ffmpeg里面swresample库，对音频数据进行冲采样。

3、可以细看一下，如果你想作实时处理也是可以改的。

4、其中SwrAllocSetOpts方法中有个参数libswresample.AvSampleFormat(1)，为什么取1，这里主要是选择采样表示方式的枚举，参考底层源码枚举，我发在下面。我这边因为音频是s16的，所以选择1。

enum AVSampleFormat { AV_SAMPLE_FMT_NONE = -1, AV_SAMPLE_FMT_U8, ///< unsigned 8 bits AV_SAMPLE_FMT_S16, ///< signed 16 bits AV_SAMPLE_FMT_S32, ///< signed 32 bits AV_SAMPLE_FMT_FLT, ///< float AV_SAMPLE_FMT_DBL, ///< double AV_SAMPLE_FMT_U8P, ///< unsigned 8 bits, planar AV_SAMPLE_FMT_S16P, ///< signed 16 bits, planar AV_SAMPLE_FMT_S32P, ///< signed 32 bits, planar AV_SAMPLE_FMT_FLTP, ///< float, planar AV_SAMPLE_FMT_DBLP, ///< double, planar AV_SAMPLE_FMT_S64, ///< signed 64 bits AV_SAMPLE_FMT_S64P, ///< signed 64 bits, planar AV_SAMPLE_FMT_NB ///< Number of sample formats. DO NOT USE if linking dynamically };

音频准备，输入音频为16k采样率音频。

(base) xxx@hu:~/GolandProjects/MediaRelay/data$ ffmpeg -i 1.wav
ffmpeg version 4.2.7-0ubuntu0.1 Copyright (c) 2000-2022 the FFmpeg developers
built with gcc 9 (Ubuntu 9.4.0-1ubuntu1~20.04.1)
configuration: --prefix=/usr --extra-version=0ubuntu0.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-nvenc --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
libavutil 56. 31.100 / 56. 31.100
libavcodec 58. 54.100 / 58. 54.100
libavformat 58. 29.100 / 58. 29.100
libavdevice 58. 8.100 / 58. 8.100
libavfilter 7. 57.100 / 7. 57.100
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 5.100 / 5. 5.100
libswresample 3. 5.100 / 3. 5.100
libpostproc 55. 5.100 / 55. 5.100
Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, wav, from '1.wav':
Metadata:
date : 2020-09-28
encoder : Lavf58.45.100
Duration: 00:04:01.75, bitrate: 256 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, mono, s16, 256 kb/s

执行情况

最终音频

(base) xxx@hu:/tmp$ ffmpeg -i .wav
ffmpeg version 4.2.7-0ubuntu0.1 Copyright (c) 2000-2022 the FFmpeg developers
built with gcc 9 (Ubuntu 9.4.0-1ubuntu1~20.04.1)
configuration: --prefix=/usr --extra-version=0ubuntu0.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-nvenc --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
libavutil 56. 31.100 / 56. 31.100
libavcodec 58. 54.100 / 58. 54.100
libavformat 58. 29.100 / 58. 29.100
libavdevice 58. 8.100 / 58. 8.100
libavfilter 7. 57.100 / 7. 57.100
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 5.100 / 5. 5.100
libswresample 3. 5.100 / 3. 5.100
libpostproc 55. 5.100 / 55. 5.100
Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, wav, from '.wav':
Duration: 00:04:01.75, bitrate: 768 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16, 768 kb/s

总结

其实在写代码过程中，有个让我特别头疼的东西，就是怎么把数组转为uint。如果大家有兴趣可以研究一下ResampleByFFmpegApi2方法的转换逻辑，会学到不少东西。

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