ZLMediaKit/src/Extension/AAC.h

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/*
* MIT License
*
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* Copyright (c) 2016-2019 xiongziliang <771730766@qq.com>
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*
* This file is part of ZLMediaKit(https://github.com/xiongziliang/ZLMediaKit).
*
* 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.
*/
#ifndef ZLMEDIAKIT_AAC_H
#define ZLMEDIAKIT_AAC_H
#include "Frame.h"
#include "Track.h"
namespace mediakit{
class AACFrame;
unsigned const samplingFrequencyTable[16] = { 96000, 88200,
64000, 48000,
44100, 32000,
24000, 22050,
16000, 12000,
11025, 8000,
7350, 0, 0, 0 };
void makeAdtsHeader(const string &strAudioCfg,AACFrame &adts);
void writeAdtsHeader(const AACFrame &adts, uint8_t *pcAdts) ;
string makeAdtsConfig(const uint8_t *pcAdts);
void getAACInfo(const AACFrame &adts,int &iSampleRate,int &iChannel);
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/**
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* aac帧adts头
*/
class AACFrame : public Frame {
public:
typedef std::shared_ptr<AACFrame> Ptr;
char *data() const override{
return (char *)buffer;
}
uint32_t size() const override {
return aac_frame_length;
}
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uint32_t dts() const override {
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return timeStamp;
}
uint32_t prefixSize() const override{
return iPrefixSize;
}
TrackType getTrackType() const override{
return TrackAudio;
}
CodecId getCodecId() const override{
return CodecAAC;
}
bool keyFrame() const override {
return false;
}
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bool configFrame() const override{
return false;
}
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public:
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unsigned int syncword = 0; //12 bslbf 同步字The bit string 1111 1111 1111说明一个ADTS帧的开始
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unsigned int id; //1 bslbf MPEG 标示符, 设置为1
unsigned int layer; //2 uimsbf Indicates which layer is used. Set to 00
unsigned int protection_absent; //1 bslbf 表示是否误码校验
unsigned int profile; //2 uimsbf 表示使用哪个级别的AAC如01 Low Complexity(LC)--- AACLC
unsigned int sf_index; //4 uimsbf 表示使用的采样率下标
unsigned int private_bit; //1 bslbf
unsigned int channel_configuration; //3 uimsbf 表示声道数
unsigned int original; //1 bslbf
unsigned int home; //1 bslbf
//下面的为改变的参数即每一帧都不同
unsigned int copyright_identification_bit; //1 bslbf
unsigned int copyright_identification_start; //1 bslbf
unsigned int aac_frame_length; // 13 bslbf 一个ADTS帧的长度包括ADTS头和raw data block
unsigned int adts_buffer_fullness; //11 bslbf 0x7FF 说明是码率可变的码流
//no_raw_data_blocks_in_frame 表示ADTS帧中有number_of_raw_data_blocks_in_frame + 1个AAC原始帧.
//所以说number_of_raw_data_blocks_in_frame == 0
//表示说ADTS帧中有一个AAC数据块并不是说没有。(一个AAC原始帧包含一段时间内1024个采样及相关数据)
unsigned int no_raw_data_blocks_in_frame; //2 uimsfb
unsigned char buffer[2 * 1024 + 7];
uint16_t sequence;
uint32_t timeStamp;
uint32_t iPrefixSize = 7;
} ;
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class AACFrameNoCacheAble : public FrameNoCacheAble {
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public:
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typedef std::shared_ptr<AACFrameNoCacheAble> Ptr;
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AACFrameNoCacheAble(char *ptr,uint32_t size,uint32_t dts,int prefixeSize = 7){
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_ptr = ptr;
_size = size;
_dts = dts;
_prefixSize = prefixeSize;
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}
TrackType getTrackType() const override{
return TrackAudio;
}
CodecId getCodecId() const override{
return CodecAAC;
}
bool keyFrame() const override {
return false;
}
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bool configFrame() const override{
return false;
}
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} ;
/**
* aac音频通道
*/
class AACTrack : public AudioTrack{
public:
typedef std::shared_ptr<AACTrack> Ptr;
/**
* adts头信息
* inputFrame中获取adts头信息
*/
AACTrack(){}
/**
* aac类型的媒体
* @param aac_cfg aac两个字节的配置信息
*/
AACTrack(const string &aac_cfg){
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if(aac_cfg.size() < 2){
throw std::invalid_argument("adts配置必须最少2个字节");
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}
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_cfg = aac_cfg.substr(0,2);
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onReady();
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}
/**
* aac类型的媒体
* @param adts_header adts头7
* @param adts_header_len adts头长度7
*/
AACTrack(const char *adts_header,int adts_header_len = 7){
if(adts_header_len < 7){
throw std::invalid_argument("adts头必须不少于7个字节");
}
_cfg = makeAdtsConfig((uint8_t*)adts_header);
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onReady();
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}
/**
* aac类型的媒体
* @param aac_frame_with_adts adts头的aac帧
*/
AACTrack(const Frame::Ptr &aac_frame_with_adts){
if(aac_frame_with_adts->getCodecId() != CodecAAC || aac_frame_with_adts->prefixSize() < 7){
throw std::invalid_argument("必须输入带adts头的aac帧");
}
_cfg = makeAdtsConfig((uint8_t*)aac_frame_with_adts->data());
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onReady();
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}
/**
* aac两个字节的配置
* @return
*/
const string &getAacCfg() const{
return _cfg;
}
/**
*
* @return
*/
CodecId getCodecId() const override{
return CodecAAC;
}
/**
* aac_cfg前是无效的Track
* @return
*/
bool ready() override {
return !_cfg.empty();
}
/**
*
* @return
*/
int getAudioSampleRate() const override{
return _sampleRate;
}
/**
* 168
* @return
*/
int getAudioSampleBit() const override{
return _sampleBit;
}
/**
*
* @return
*/
int getAudioChannel() const override{
return _channel;
}
/**
* ,aac_cfg
* @param frame
*/
void inputFrame(const Frame::Ptr &frame) override{
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if(_cfg.empty()){
//未获取到aac_cfg信息
if(frame->prefixSize() >= 7) {
//7个字节的adts头
_cfg = makeAdtsConfig(reinterpret_cast<const uint8_t *>(frame->data()));
onReady();
}else{
WarnL << "无法获取adts头!";
}
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}
AudioTrack::inputFrame(frame);
}
private:
/**
* 2aac配置
*/
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void onReady(){
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AACFrame aacFrame;
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makeAdtsHeader(_cfg,aacFrame);
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getAACInfo(aacFrame,_sampleRate,_channel);
}
Track::Ptr clone() override {
return std::make_shared<std::remove_reference<decltype(*this)>::type >(*this);
}
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//生成sdp
Sdp::Ptr getSdp() override ;
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private:
string _cfg;
int _sampleRate = 0;
int _sampleBit = 16;
int _channel = 0;
};
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/**
* aac类型SDP
*/
class AACSdp : public Sdp {
public:
/**
*
* @param aac_cfg aac两个字节的配置描述
* @param sample_rate
* @param playload_type rtp playload type 98
* @param bitrate
*/
AACSdp(const string &aac_cfg,
int sample_rate,
int playload_type = 98,
int bitrate = 128) : Sdp(sample_rate,playload_type){
_printer << "m=audio 0 RTP/AVP " << playload_type << "\r\n";
_printer << "b=AS:" << bitrate << "\r\n";
_printer << "a=rtpmap:" << playload_type << " MPEG4-GENERIC/" << sample_rate << "\r\n";
char configStr[32] = {0};
snprintf(configStr, sizeof(configStr), "%02X%02X", (uint8_t)aac_cfg[0], (uint8_t)aac_cfg[1]);
_printer << "a=fmtp:" << playload_type << " streamtype=5;profile-level-id=1;mode=AAC-hbr;"
<< "sizelength=13;indexlength=3;indexdeltalength=3;config="
<< configStr << "\r\n";
_printer << "a=control:trackID=" << getTrackType() << "\r\n";
}
string getSdp() const override {
return _printer;
}
TrackType getTrackType() const override {
return TrackAudio;
}
CodecId getCodecId() const override {
return CodecAAC;
}
private:
_StrPrinter _printer;
};
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}//namespace mediakit
#endif //ZLMEDIAKIT_AAC_H