AAC编码实战

作者: coder_feng | 来源:发表于2021-05-20 21:03 被阅读0次

0.编码流程图

音频编码.png

1前言

本文将会使用上篇文章手动编译的ffmpeg对输入的PCM数据进行AAC编码，但是fdk-aac 对输入的PCM数据是有要求的，如果输入的参数不符合预期，就会报错

2参数要求

参数要求

2.1采样格式

必须是16位整数PCM

2.2 采样率

支持的采样率有(Hz)

8000、11025、12000、16000、22050、24000、32000
44100、48000、64000、88200、96000

3.命令行编码

3.1基本使用

# pcm -> aac
ffmpeg -ar 44100 -ac 2 -f s16le -i in.pcm -c:a libfdk_aac out.aac
 
# wav -> aac
# 为了简化指令，本文后面会尽量使用in.wav取代in.pcm
ffmpeg -i in.wav -c:a libfdk_aac out.aac

ar 指定比特率
ac 指定声道数
-f 音频样本格式

这三个参数必须要和in.pcm 对应上

-c:a 设置音频编码(c表示codec编解码器,a表示audio音频)，等价写法-codec:a -acodec

songlin@feng-sl  ~/audio/aac   master ±✚  ffmpeg -ar 44100 -ac 2 -f f32le -i 44100_f32le_2.pcm -c:1 libfdk_aac out.aac
ffmpeg version 4.3.2 Copyright (c) 2000-2021 the FFmpeg developers
  built with Apple clang version 12.0.0 (clang-1200.0.32.29)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.3.2_4 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
[f32le @ 0x7fabca010600] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, f32le, from '44100_f32le_2.pcm':
  Duration: 00:00:17.36, bitrate: 2822 kb/s
    Stream #0:0: Audio: pcm_f32le, 44100 Hz, stereo, flt, 2822 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_f32le (native) -> aac (native))
Press [q] to stop, [?] for help
Output #0, adts, to 'out.aac':
  Metadata:
    encoder         : Lavf58.45.100
    Stream #0:0: Audio: aac (LC), 44100 Hz, stereo, fltp, 128 kb/s
    Metadata:
      encoder         : Lavc58.91.100 aac
size=     277kB time=00:00:17.36 bitrate= 130.8kbits/s speed=85.7x
video:0kB audio:272kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.880452%
[aac @ 0x7fabcb81cc00] Qavg: 252.198
 songlin@feng-sl  ~/audio/aac   master ±✚  ls
44100_f32le_2.pcm out.aac

pcm->aac大小.png

PCM：6.1M，AAC：284k

3.2查看ac文件规格

✘ songlin@feng-sl  ~/audio/aac   master ±✚  ffprobe out.aac
ffprobe version 4.3.2 Copyright (c) 2007-2021 the FFmpeg developers
  built with Apple clang version 12.0.0 (clang-1200.0.32.29)
  configuration: --prefix=/usr/local/ffmpeg --enable-shared --disable-static --enable-gpl --enable-nonfree --enable-libfdk-aac --enable-libx264 --enable-libx265
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
[aac @ 0x7fbaa5808200] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'out.aac':
  Duration: 00:00:16.74, bitrate: 135 kb/s
    Stream #0:0: Audio: aac (LC), 44100 Hz, stereo, fltp, 135 kb/s

3.3 常用参数

从上面可以看到aac文件的比特率是135kb/s,那么这个值我们能修改么？答案是肯定的，可以通过一些输出参数来达到这个目的

-b:a设置输出比特率,-b:a 48k

 ✘ songlin@feng-sl  ~/audio/aac   master ±✚  ffmpeg -ar 44100 -ac 2 -f f32le -i 44100_f32le_2.pcm -c:a libfdk_aac -b:a 48k 48kout.aac
ffmpeg version 4.3.2 Copyright (c) 2000-2021 the FFmpeg developers
  built with Apple clang version 12.0.0 (clang-1200.0.32.29)
  configuration: --prefix=/usr/local/ffmpeg --enable-shared --disable-static --enable-gpl --enable-nonfree --enable-libfdk-aac --enable-libx264 --enable-libx265
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
[f32le @ 0x7fd90480c400] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, f32le, from '44100_f32le_2.pcm':
  Duration: 00:00:17.36, bitrate: 2822 kb/s
    Stream #0:0: Audio: pcm_f32le, 44100 Hz, stereo, flt, 2822 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_f32le (native) -> aac (libfdk_aac))
Press [q] to stop, [?] for help
Output #0, adts, to '48kout.aac':
  Metadata:
    encoder         : Lavf58.45.100
    Stream #0:0: Audio: aac (libfdk_aac), 44100 Hz, stereo, s16, 48 kb/s
    Metadata:
      encoder         : Lavc58.91.100 libfdk_aac
size=     103kB time=00:00:17.36 bitrate=  48.4kbits/s speed= 103x
video:0kB audio:103kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000000%
 songlin@feng-sl  ~/audio/aac   master ±✚  o
 songlin@feng-sl  ~/audio/aac   master ±✚  ffprobe 48kout.aac
ffprobe version 4.3.2 Copyright (c) 2007-2021 the FFmpeg developers
  built with Apple clang version 12.0.0 (clang-1200.0.32.29)
  configuration: --prefix=/usr/local/ffmpeg --enable-shared --disable-static --enable-gpl --enable-nonfree --enable-libfdk-aac --enable-libx264 --enable-libx265
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
[aac @ 0x7fc72b008200] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from '48kout.aac':
  Duration: 00:00:17.42, bitrate: 48 kb/s
    Stream #0:0: Audio: aac (LC), 44100 Hz, stereo, fltp, 48 kb/s

48kbacc.png

-profile:a 设置输出规格( Value is one of LC, HE-AAC, HE-AACv2, LD, or ELD. Default is LC.)

取值有：
aac_low：Low Complexity AAC (LC)，默认值
aac_he：High Efficiency AAC (HE-AAC)
aac_he_v2：High Efficiency AAC version 2 (HE-AACv2)
aac_ld：Low Delay AAC (LD)
aac_eld：Enhanced Low Delay AAC (ELD)

示例

 ✘ songlin@feng-sl  ~/audio/aac   master ±✚  ffmpeg -ar 44100 -ac 2 -f f32le -i 44100_f32le_2.pcm -c:a libfdk_aac -profile:a aac_he_v2  -b:a 48k 48k_aac_he_v2_out.aac
ffmpeg version 4.3.2 Copyright (c) 2000-2021 the FFmpeg developers
  built with Apple clang version 12.0.0 (clang-1200.0.32.29)
  configuration: --prefix=/usr/local/ffmpeg --enable-shared --disable-static --enable-gpl --enable-nonfree --enable-libfdk-aac --enable-libx264 --enable-libx265
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
[f32le @ 0x7fd022010a00] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, f32le, from '44100_f32le_2.pcm':
  Duration: 00:00:17.36, bitrate: 2822 kb/s
    Stream #0:0: Audio: pcm_f32le, 44100 Hz, stereo, flt, 2822 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_f32le (native) -> aac (libfdk_aac))
Press [q] to stop, [?] for help
Output #0, adts, to '48k_aac_he_v2_out.aac':
  Metadata:
    encoder         : Lavf58.45.100
    Stream #0:0: Audio: aac (libfdk_aac) (HE-AACv2), 44100 Hz, stereo, s16, 48 kb/s
    Metadata:
      encoder         : Lavc58.91.100 libfdk_aac
size=     103kB time=00:00:17.39 bitrate=  48.4kbits/s speed=76.2x
video:0kB audio:103kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000000%
 songlin@feng-sl  ~/audio/aac   master ±✚  ffprobe 48k_aac_he_v2_out.aac
ffprobe version 4.3.2 Copyright (c) 2007-2021 the FFmpeg developers
  built with Apple clang version 12.0.0 (clang-1200.0.32.29)
  configuration: --prefix=/usr/local/ffmpeg --enable-shared --disable-static --enable-gpl --enable-nonfree --enable-libfdk-aac --enable-libx264 --enable-libx265
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
[aac @ 0x7fcd70810000] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from '48k_aac_he_v2_out.aac':
  Duration: 00:00:18.25, bitrate: 46 kb/s
    Stream #0:0: Audio: aac (HE-AACv2), 44100 Hz, stereo, fltp, 46 kb/s

一旦设置了输出规格，会自动设置一个合适的输出比特率(我设置的是48k结果输出了46k)

-vbr
vbr模式

如果开启了VBR模式，-b:a选项将会被忽略，但-profile:a选项仍然有效
取值范围是0 ~ 5
0：默认值，关闭VBR模式，开启CBR模式（Constant Bit Rate，固定比特率）
1：质量最低（但是音质仍旧很棒）
5：质量最高

VBR	kbps/channel	Audio Object Type
1	20-32	LC、HE、HEv2
2	32-40	LC、HE、HEv2
3	48-56	LC、HE、HEv2
4	64-72	LC
5	96-112	LC

示例

** -b: a 被忽略例子**
 songlin@feng-sl  ~/audio/aac   master ±✚  ffmpeg -ar 44100 -ac 2 -f f32le -i 44100_f32le_2.pcm -c:a libfdk_aac -vbr 1  -b:a 48k vb1out.aac
ffmpeg version 4.3.2 Copyright (c) 2000-2021 the FFmpeg developers
  built with Apple clang version 12.0.0 (clang-1200.0.32.29)
  configuration: --prefix=/usr/local/ffmpeg --enable-shared --disable-static --enable-gpl --enable-nonfree --enable-libfdk-aac --enable-libx264 --enable-libx265
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
[f32le @ 0x7fa4d0013400] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, f32le, from '44100_f32le_2.pcm':
  Duration: 00:00:17.36, bitrate: 2822 kb/s
    Stream #0:0: Audio: pcm_f32le, 44100 Hz, stereo, flt, 2822 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_f32le (native) -> aac (libfdk_aac))
Press [q] to stop, [?] for help
[libfdk_aac @ 0x7fa4d1009a00] Note, the VBR setting is unsupported and only works with some parameter combinations
Output #0, adts, to 'vb1out.aac':
  Metadata:
    encoder         : Lavf58.45.100
    Stream #0:0: Audio: aac (libfdk_aac), 44100 Hz, stereo, s16, 48 kb/s
    Metadata:
      encoder         : Lavc58.91.100 libfdk_aac
size=     149kB time=00:00:17.36 bitrate=  70.4kbits/s speed=89.8x
video:0kB audio:149kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000000%
 songlin@feng-sl  ~/audio/aac   master ±✚  ffprobe vb1out.aac
ffprobe version 4.3.2 Copyright (c) 2007-2021 the FFmpeg developers
  built with Apple clang version 12.0.0 (clang-1200.0.32.29)
  configuration: --prefix=/usr/local/ffmpeg --enable-shared --disable-static --enable-gpl --enable-nonfree --enable-libfdk-aac --enable-libx264 --enable-libx265
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
[aac @ 0x7fc07100ac00] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'vb1out.aac':
  Duration: 00:00:12.49, bitrate: 97 kb/s
    Stream #0:0: Audio: aac (LC), 44100 Hz, stereo, fltp, 97 kb/s
 

**profile 依然生效**
✘ songlin@feng-sl  ~/audio/aac   master ±✚  ffmpeg -ar 44100 -ac 2 -f f32le -i 44100_f32le_2.pcm -c:a libfdk_aac -profile:a aac_he_v2  -vbr 1 vb1_profile_out.aac
ffmpeg version 4.3.2 Copyright (c) 2000-2021 the FFmpeg developers
  built with Apple clang version 12.0.0 (clang-1200.0.32.29)
  configuration: --prefix=/usr/local/ffmpeg --enable-shared --disable-static --enable-gpl --enable-nonfree --enable-libfdk-aac --enable-libx264 --enable-libx265
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
[f32le @ 0x7fa64980ca00] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, f32le, from '44100_f32le_2.pcm':
  Duration: 00:00:17.36, bitrate: 2822 kb/s
    Stream #0:0: Audio: pcm_f32le, 44100 Hz, stereo, flt, 2822 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_f32le (native) -> aac (libfdk_aac))
Press [q] to stop, [?] for help
[libfdk_aac @ 0x7fa64980dc00] Note, the VBR setting is unsupported and only works with some parameter combinations
Output #0, adts, to 'vb1_profile_out.aac':
  Metadata:
    encoder         : Lavf58.45.100
    Stream #0:0: Audio: aac (libfdk_aac) (HE-AACv2), 44100 Hz, stereo, s16
    Metadata:
      encoder         : Lavc58.91.100 libfdk_aac
size=      70kB time=00:00:17.39 bitrate=  32.8kbits/s speed=  88x
video:0kB audio:70kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000000%
 songlin@feng-sl  ~/audio/aac   master ±✚  ffprobe vb1_profile_out.aac
ffprobe version 4.3.2 Copyright (c) 2007-2021 the FFmpeg developers
  built with Apple clang version 12.0.0 (clang-1200.0.32.29)
  configuration: --prefix=/usr/local/ffmpeg --enable-shared --disable-static --enable-gpl --enable-nonfree --enable-libfdk-aac --enable-libx264 --enable-libx265
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
[aac @ 0x7f8d3b80ca00] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'vb1_profile_out.aac':
  Duration: 00:00:14.58, bitrate: 39 kb/s
    Stream #0:0: Audio: aac (HE-AACv2), 44100 Hz, stereo, fltp, 39 kb/s

3.3 文件格式

音频基础
从文章中可以知道AAC编码的文件扩展名主要有3种：aac、m4a、mp4。
示例

# m4a
ffmpeg -i in.wav -c:a libfdk_aac out.m4a
 
# mp4
ffmpeg -i in.wav -c:a libfdk_aac out.mp4

4.编程

ffmpegs.h

#ifndef FFMPEGS_H
#define FFMPEGS_H

extern "C" {
#include <libavformat/avformat.h>
}

typedef struct {
    const char *filename;
    int sampleRate;
    AVSampleFormat sampleFmt;
    int chLayout;
} AudioEncodeSpec;

class FFmpegs
{
public:
    FFmpegs();

    static void aacEncode(AudioEncodeSpec &in,
                          const char *outFilename);
};

#endif // FFMPEGS_H

ffmpegs.cpp

#include "ffmpegs.h"
#include <QDebug>
#include <QFile>

extern "C" {
#include <libavcodec/avcodec.h>
#include <libavutil/avutil.h>
}

#define ERROR_BUF(ret) \
    char errbuf[1024]; \
    av_strerror(ret, errbuf, sizeof (errbuf));

FFmpegs::FFmpegs() {

}

// 检查采样格式
static int check_sample_fmt(const AVCodec *codec,
                            enum AVSampleFormat sample_fmt) {
    const enum AVSampleFormat *p = codec->sample_fmts;

    while (*p != AV_SAMPLE_FMT_NONE) {
//        qDebug() << av_get_sample_fmt_name(*p);
        if (*p == sample_fmt) return 1;
        p++;
    }
    return 0;
}

// 音频编码
// 返回负数：中途出现了错误
// 返回0：编码操作正常完成
static int encode(AVCodecContext *ctx,
                  AVFrame *frame,
                  AVPacket *pkt,
                  QFile &outFile) {
    // 发送数据到编码器
    int ret = avcodec_send_frame(ctx, frame);
    if (ret < 0) {
        ERROR_BUF(ret);
        qDebug() << "avcodec_send_frame error" << errbuf;
        return ret;
    }

    // 不断从编码器中取出编码后的数据
    // while (ret >= 0)
    while (true) {
        ret = avcodec_receive_packet(ctx, pkt);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
            // 继续读取数据到frame，然后送到编码器
            return 0;
        } else if (ret < 0) { // 其他错误
            return ret;
        }

        // 成功从编码器拿到编码后的数据
        // 将编码后的数据写入文件
        outFile.write((char *) pkt->data, pkt->size);

        // 释放pkt内部的资源
        av_packet_unref(pkt);
    }
}

void FFmpegs::aacEncode(AudioEncodeSpec &in,
                        const char *outFilename) {
    // 文件
    QFile inFile(in.filename);
    QFile outFile(outFilename);

    // 返回结果
    int ret = 0;

    // 编码器
    AVCodec *codec = nullptr;

    // 编码上下文
    AVCodecContext *ctx = nullptr;

    // 存放编码前的数据（pcm）
    AVFrame *frame = nullptr;

    // 存放编码后的数据（aac）
    AVPacket *pkt = nullptr;

    // 获取编码器
//    codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
    codec = avcodec_find_encoder_by_name("libfdk_aac");
    if (!codec) {
        qDebug() << "encoder not found";
        return;
    }

    // libfdk_aac对输入数据的要求：采样格式必须是16位整数
    // 检查输入数据的采样格式
    if (!check_sample_fmt(codec, in.sampleFmt)) {
        qDebug() << "unsupported sample format"
                 << av_get_sample_fmt_name(in.sampleFmt);
        return;
    }

    // 创建编码上下文
    ctx = avcodec_alloc_context3(codec);
    if (!ctx) {
        qDebug() << "avcodec_alloc_context3 error";
        return;
    }

    // 设置PCM参数
    ctx->sample_rate = in.sampleRate;
    ctx->sample_fmt = in.sampleFmt;
    ctx->channel_layout = in.chLayout;
    // 比特率
    ctx->bit_rate = 32000;
    // 规格
    ctx->profile = FF_PROFILE_AAC_HE_V2;

    // 打开编码器
//    AVDictionary *options = nullptr;
//    av_dict_set(&options, "vbr", "5", 0);
//    ret = avcodec_open2(ctx, codec, &options);

//    avcodec_open2 error Invalid argument https://stackoverflow.com/questions/26205017/ffmpeg-avcodec-open2-returns-22-if-i-change-my-speaker-configuration/26205126
    ret = avcodec_open2(ctx, codec, nullptr);
    if (ret < 0) {
        ERROR_BUF(ret);
        qDebug() << "avcodec_open2 error" << errbuf;
        goto end;
    }

    // 创建AVFrame
    frame = av_frame_alloc();
    if (!frame) {
        qDebug() << "av_frame_alloc error";
        goto end;
    }

    // frame缓冲区中的样本帧数量（由ctx->frame_size决定）
    frame->nb_samples = ctx->frame_size;
    frame->format = ctx->sample_fmt;
    frame->channel_layout = ctx->channel_layout;

    // 利用nb_samples、format、channel_layout创建缓冲区
    ret = av_frame_get_buffer(frame, 0);
    if (ret) {
        ERROR_BUF(ret);
        qDebug() << "av_frame_get_buffer error" << errbuf;
        goto end;
    }

    // 创建AVPacket
    pkt = av_packet_alloc();
    if (!pkt) {
        qDebug() << "av_packet_alloc error";
        goto end;
    }

    // 打开文件
    if (!inFile.open(QFile::ReadOnly)) {
        qDebug() << "file open error" << in.filename;
        goto end;
    }
    if (!outFile.open(QFile::WriteOnly)) {
        qDebug() << "file open error" << outFilename;
        goto end;
    }

    // 读取数据到frame中
    while ((ret = inFile.read((char *) frame->data[0],
                              frame->linesize[0])) > 0) {
        // 从文件中读取的数据，不足以填满frame缓冲区
        if (ret < frame->linesize[0]) {
            int bytes = av_get_bytes_per_sample((AVSampleFormat) frame->format);
            int ch = av_get_channel_layout_nb_channels(frame->channel_layout);
            // 设置真正有效的样本帧数量
            // 防止编码器编码了一些冗余数据
            frame->nb_samples = ret / (bytes * ch);
        }

        // 进行编码
        if (encode(ctx, frame, pkt, outFile) < 0) {
            goto end;
        }
    }

    // 刷新缓冲区
    encode(ctx, nullptr, pkt, outFile);

end:
    // 关闭文件
    inFile.close();
    outFile.close();

    // 释放资源
    av_frame_free(&frame);
    av_packet_free(&pkt);
    avcodec_free_context(&ctx);

    qDebug() << "线程正常结束";
}

audiothread.h

#ifndef AUDIOTHREAD_H
#define AUDIOTHREAD_H

#include <QThread>

class AudioThread : public QThread {
    Q_OBJECT
private:
    void run();

public:
    explicit AudioThread(QObject *parent = nullptr);
    ~AudioThread();
signals:

};

#endif // AUDIOTHREAD_H

audiothread.cpp

#include "audiothread.h"
#include <QDebug>
#include "ffmpegs.h"


AudioThread::AudioThread(QObject *parent) : QThread(parent)
{
    //当监听到线程结束时候(finished)，就调用deleteLater回收内存
    connect(this,&AudioThread::finished,
            this,&AudioThread::deleteLater);
}

AudioThread::~AudioThread(){
    //断开所有的连接
    disconnect();
    //内存回收之前，正常结束线程
    requestInterruption();
    //安全退出
    quit();
    wait();
    qDebug() << this << "析构(内存被回收)";
}

void AudioThread::run(){
    AudioEncodeSpec in;
    in.filename = "/Users/songlin/audio/resample/44100_s16le_2.pcm";
    in.sampleRate = 44100;
    in.sampleFmt = AV_SAMPLE_FMT_S16;
    in.chLayout = AV_CH_LAYOUT_STEREO;
    FFmpegs::aacEncode(in,"/Users/songlin/audio/aac_encode/out.aac");
}

5.参考链接

参考链接

AAC编码实战

0.编码流程图

1前言

2参数要求

2.1采样格式

2.2 采样率

3.命令行编码

3.1基本使用

3.2查看ac文件规格

3.3 常用参数

3.3 文件格式

4.编程

5.参考链接

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