Go语言调用ffmpeg-api实现音频重采样
mylove136 · 154浏览 · 发布于2022-12-07
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，希望大家能喜欢
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
input file format info -> AudioFormat:1,NumChannels:1,SampleRate:16000,ByteRate:32000, BlockAlign:2,BitsPerSample:16INFO : 2022/12/06 17:14:49.937547 csdn_wav_util.go:62: Create tempFile /tmp/2402235346.wav 11603961
最终音频
(base) xxx@hu:/tmp$ ffmpeg -i 2402235346.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 '2402235346.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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