[ffmpeg] aac 音频编码

aac 介绍

aac 简单说就是音频的一种压缩编码器，相同音质下压缩比 mp3好，目前比较常用。

aac 编码支持的格式

aac 支持的 sample_fmts: 8

aac 支持的 samplerates: 96000 88200 64000 48000 44100 32000 24000 22050 16000 12000 11025 8000 7350

通过 AVCodec 中的 supported_xx 字段来获取

具体代码

static int check_sample_fmt(const AVCodec* codec, enum AVSampleFormat sample_fmt)
{
    const enum AVSampleFormat* p = codec->sample_fmts;
    cout << "sample_fmts: ";
    while (*p != AV_SAMPLE_FMT_NONE)
    {
        cout << *p << " ";
        p++;
    }
	cout << endl;
    p = codec->sample_fmts;
    while (*p != AV_SAMPLE_FMT_NONE) {
        if (*p == sample_fmt)
            return 1;
        p++;
    }
    return 0;
}

也可以用命令行获取支持格式，以及可设置的额外参数

具体实现

编码步骤

// 1. 通过名字或者 id 找到编码器(相当于找到了那个能力结构体指针)；获取的结构体会有些编码器的简单介绍，以及编码器支持的能力
// 2. 通过编码器创建上下文，相当于创建上下文实例，并将 codec 指针保存在上下文中，并根据编码器能力初始化一些参数
 // 3. 根据用户需要，以及编码器支持的能力，将编码参数设置到编码器上下文中
 // 4. 根据编码器上下文初始化编码器
 // 5. 创建 avframe 并把编码器上下文中的参数赋值给他
 // 6. avframe 根据参数，算出每次编码需要的内部大小，并分配
 // 7. 将编码数据传给 avframe
 // 8. 将 avframe 传给 avcodec_send_frame
 // 9. 通过 avcodec_receive_packet 获取 avpacket 数据

具体代码

目前直接拿了 fffmpeg demo，后面有空按照步骤规整一下。

/*
 * Copyright (c) 2001 Fabrice Bellard
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */

 /**
  * @file
  * audio encoding with libavcodec API example.
  *
  * @example encode_audio.c
  */

#include 
#include 
#include 
#include 
using namespace std;
extern"C"
{
#include 
#include 
#include 
#include 
#include 
}


const int sampling_frequencies[] = {
	96000,  // 0x0
	88200,  // 0x1
	64000,  // 0x2
	48000,  // 0x3
	44100,  // 0x4
	32000,  // 0x5
	24000,  // 0x6
	22050,  // 0x7
	16000,  // 0x8
	12000,  // 0x9
	11025,  // 0xa
	8000   // 0xb
	// 0xc d e f是保留的
};

int adts_header(char* const p_adts_header, const int data_length,
	const int profile, const int samplerate,
	const int channels)
{

	int sampling_frequency_index = 3; // 默认使用48000hz
	int adtsLen = data_length + 7;

	int frequencies_size = sizeof(sampling_frequencies) / sizeof(sampling_frequencies[0]);
	int i = 0;
	for (i = 0; i < frequencies_size; i++)
	{
		if (sampling_frequencies[i] == samplerate)
		{
			sampling_frequency_index = i;
			break;
		}
	}
	if (i >= frequencies_size)
	{
		printf("unsupport samplerate:%d\n", samplerate);
		return -1;
	}

	p_adts_header[0] = 0xff;         //syncword:0xfff                          高8bits
	p_adts_header[1] = 0xf0;         //syncword:0xfff                          低4bits
	p_adts_header[1] |= (0 << 3);    //MPEG Version:0 for MPEG-4,1 for MPEG-2  1bit
	p_adts_header[1] |= (0 << 1);    //Layer:0                                 2bits
	p_adts_header[1] |= 1;           //protection absent:1                     1bit

	p_adts_header[2] = (profile) << 6;            //profile:profile               2bits
	p_adts_header[2] |= (sampling_frequency_index & 0x0f) << 2; //sampling frequency index:sampling_frequency_index  4bits
	p_adts_header[2] |= (0 << 1);             //private bit:0                   1bit
	p_adts_header[2] |= (channels & 0x04) >> 2; //channel configuration:channels  高1bit

	p_adts_header[3] = (channels & 0x03) << 6; //channel configuration:channels 低2bits
	p_adts_header[3] |= (0 << 5);               //original：0                1bit
	p_adts_header[3] |= (0 << 4);               //home：0                    1bit
	p_adts_header[3] |= (0 << 3);               //copyright id bit：0        1bit
	p_adts_header[3] |= (0 << 2);               //copyright id start：0      1bit
	p_adts_header[3] |= ((adtsLen & 0x1800) >> 11);           //frame length：value   高2bits

	p_adts_header[4] = (uint8_t)((adtsLen & 0x7f8) >> 3);     //frame length:value    中间8bits
	p_adts_header[5] = (uint8_t)((adtsLen & 0x7) << 5);       //frame length:value    低3bits
	p_adts_header[5] |= 0x1f;                                 //buffer fullness:0x7ff 高5bits
	p_adts_header[6] = 0xfc;      //       //buffer fullness:0x7ff 低6bits
	// number_of_raw_data_blocks_in_frame：
	//    表示ADTS帧中有number_of_raw_data_blocks_in_frame + 1个AAC原始帧。

	return 0;
}

/* select layout with the highest channel count */
static int select_channel_layout(const AVCodec* codec, AVChannelLayout* dst)
{
    const AVChannelLayout* p, * best_ch_layout;
    int best_nb_channels = 0;

    if (!codec->ch_layouts)
    {
        AVChannelLayout layout = AV_CHANNEL_LAYOUT_STEREO;
        return av_channel_layout_copy(dst, &layout);

    }

    p = codec->ch_layouts;
    while (p->nb_channels) {
        int nb_channels = p->nb_channels;

        if (nb_channels > best_nb_channels) {
            best_ch_layout = p;
            best_nb_channels = nb_channels;
        }
        p++;
    }
    return av_channel_layout_copy(dst, best_ch_layout);
}

static void encode(AVCodecContext* ctx, AVFrame* frame, AVPacket* pkt,
    FILE* output)
{
    int ret;

    /* send the frame for encoding */
    ret = avcodec_send_frame(ctx, frame);
    if (ret < 0) {
        fprintf(stderr, "Error sending the frame to the encoder\n");
        exit(1);
    }

    /* read all the available output packets (in general there may be any
     * number of them */
    while (ret >= 0) {
        ret = avcodec_receive_packet(ctx, pkt);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
            return;
        else if (ret < 0) {
            fprintf(stderr, "Error encoding audio frame\n");
            exit(1);
        }
        char adts_header_buf[7] = { 0 };
        adts_header(adts_header_buf, pkt->size, ctx->profile, ctx->sample_rate, ctx->ch_layout.nb_channels);
        fwrite(adts_header_buf, 1, 7, output);

        fwrite(pkt->data, 1, pkt->size, output);
        av_packet_unref(pkt);
    }
}

int main(int argc, char** argv)
{
    // 1. 通过名字或者 id 找到编码器(相当于找到了那个能力结构体指针)；获取的结构体会有些编码器的简单介绍，以及编码器支持的能力
    // 2. 通过编码器创建上下文，相当于创建上下文实例，并将 codec 指针保存在上下文中，并根据编码器能力初始化一些参数
    // 3. 根据用户需要，以及编码器支持的能力，将编码参数设置到编码器上下文中
    // 4. 根据编码器上下文初始化编码器
    // 5. 创建 avframe 并把编码器上下文中的参数赋值给他
    // 6. avframe 根据参数，算出每次编码需要的内部大小，并分配
    // 7. 将编码数据传给 avframe
    // 8. 将 avframe 传给 avcodec_send_frame
    // 9. 通过 avcodec_receive_packet 获取 avpacket 数据

    const char* filename;
    const AVCodec* codec;
    AVCodecContext* c = NULL;
    AVFrame* frame;
    AVPacket* pkt;
    int i, j, k, ret;
    FILE* f;
    float* samples;
    float t, tincr;

    if (argc <= 1) {
        fprintf(stderr, "Usage: %s 
\n", argv[0]);
        return 0;
    }
    filename = argv[1];
    codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }
    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate audio codec context\n");
        exit(1);
    }

    c->bit_rate = 64000;
    c->sample_fmt = AV_SAMPLE_FMT_FLTP;
    c->sample_rate = 48000;
    ret = select_channel_layout(codec, &c->ch_layout);
    if (ret < 0)
        exit(1);

    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }

    f = fopen(filename, "wb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }

    /* packet for holding encoded output */
    pkt = av_packet_alloc();
    if (!pkt) {
        fprintf(stderr, "could not allocate the packet\n");
        exit(1);
    }

    /* frame containing input raw audio */
    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate audio frame\n");
        exit(1);
    }

    frame->nb_samples = c->frame_size;
    frame->format = c->sample_fmt;
    ret = av_channel_layout_copy(&frame->ch_layout, &c->ch_layout);
    if (ret < 0)
        exit(1);

    /* allocate the data buffers */
    ret = av_frame_get_buffer(frame, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not allocate audio data buffers\n");
        exit(1);
    }

    /* encode a single tone sound */
    t = 0;
    tincr = 2 * M_PI * 440.0 / c->sample_rate;
    for (i = 0; i < 200; i++) {
        /* make sure the frame is writable -- makes a copy if the encoder
         * kept a reference internally */
        ret = av_frame_make_writable(frame);
        if (ret < 0)
            exit(1);
        for (k = 0; k < c->ch_layout.nb_channels; k++)
        {
			samples = (float*)frame->data[k];
			for (j = 0; j < c->frame_size; j++) {
				samples[j] = sin(t) * 10000;
				t += tincr;
			}
        }
       
        encode(c, frame, pkt, f);
    }

    encode(c, NULL, pkt, f);
    fclose(f);
    av_frame_free(&frame);
    av_packet_free(&pkt);
    avcodec_free_context(&c);

    return 0;
}

备注

ffmpeg demo 在 c++ 环境不能直接编译通过

1. 添加头文件需要加上 extern “C”

extern"C"
{
#include 
#include 
#include 
#include 
#include 
}

2. 另一个报错不清楚，ffmpeg是怎么编译通过的，c++这边会报错

av_channel_layout_copy(dst, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);

需要改成
AVChannelLayout layout = AV_CHANNEL_LAYOUT_STEREO;
av_channel_layout_copy(dst, &layout);