[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
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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
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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
在这里插入图片描述

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 <stdint.h>
#include <stdio.h>
#include <stdlib.h>
#include <iostream>
using namespace std;
extern"C"
{
#include <libavcodec/avcodec.h>
#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <libavutil/frame.h>
#include <libavutil/samplefmt.h>
}const int sampling_frequencies[] = {96000,  // 0x088200,  // 0x164000,  // 0x248000,  // 0x344100,  // 0x432000,  // 0x524000,  // 0x622050,  // 0x716000,  // 0x812000,  // 0x911025,  // 0xa8000   // 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48000hzint 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8bitsp_adts_header[1] = 0xf0;         //syncword:0xfff                          低4bitsp_adts_header[1] |= (0 << 3);    //MPEG Version:0 for MPEG-4,1 for MPEG-2  1bitp_adts_header[1] |= (0 << 1);    //Layer:0                                 2bitsp_adts_header[1] |= 1;           //protection absent:1                     1bitp_adts_header[2] = (profile) << 6;            //profile:profile               2bitsp_adts_header[2] |= (sampling_frequency_index & 0x0f) << 2; //sampling frequency index:sampling_frequency_index  4bitsp_adts_header[2] |= (0 << 1);             //private bit:0                   1bitp_adts_header[2] |= (channels & 0x04) >> 2; //channel configuration:channels  高1bitp_adts_header[3] = (channels & 0x03) << 6; //channel configuration:channels 低2bitsp_adts_header[3] |= (0 << 5);               //original：0                1bitp_adts_header[3] |= (0 << 4);               //home：0                    1bitp_adts_header[3] |= (0 << 3);               //copyright id bit：0        1bitp_adts_header[3] |= (0 << 2);               //copyright id start：0      1bitp_adts_header[3] |= ((adtsLen & 0x1800) >> 11);           //frame length：value   高2bitsp_adts_header[4] = (uint8_t)((adtsLen & 0x7f8) >> 3);     //frame length:value    中间8bitsp_adts_header[5] = (uint8_t)((adtsLen & 0x7) << 5);       //frame length:value    低3bitsp_adts_header[5] |= 0x1f;                                 //buffer fullness:0x7ff 高5bitsp_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 <output file>\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

添加头文件需要加上 extern “C”

extern"C"
{
#include <libavcodec/avcodec.h>
#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <libavutil/frame.h>
#include <libavutil/samplefmt.h>
}

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);