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 <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, // 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 <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);
