/*
* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
*
* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
*
* Use of this source code is governed by MIT license that can be found in the
* LICENSE file in the root of the source tree. All contributing project authors
* may be found in the AUTHORS file in the root of the source tree.
*/
#include "AAC.h"
#ifdef ENABLE_MP4
#include "mpeg4-aac.h"
#endif
namespace mediakit{
#ifndef ENABLE_MP4
unsigned const samplingFrequencyTable[16] = { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350, 0, 0, 0 };
class AdtsHeader{
public:
unsigned int syncword = 0; //12 bslbf 同步字The bit string ‘1111 1111 1111’，说明一个ADTS帧的开始
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
};
static void dumpAdtsHeader(const AdtsHeader &hed, uint8_t *out) {
out[0] = (hed.syncword >> 4 & 0xFF); //8bit
out[1] = (hed.syncword << 4 & 0xF0); //4 bit
out[1] |= (hed.id << 3 & 0x08); //1 bit
out[1] |= (hed.layer << 1 & 0x06); //2bit
out[1] |= (hed.protection_absent & 0x01); //1 bit
out[2] = (hed.profile << 6 & 0xC0); // 2 bit
out[2] |= (hed.sf_index << 2 & 0x3C); //4bit
out[2] |= (hed.private_bit << 1 & 0x02); //1 bit
out[2] |= (hed.channel_configuration >> 2 & 0x03); //1 bit
out[3] = (hed.channel_configuration << 6 & 0xC0); // 2 bit
out[3] |= (hed.original << 5 & 0x20); //1 bit
out[3] |= (hed.home << 4 & 0x10); //1 bit
out[3] |= (hed.copyright_identification_bit << 3 & 0x08); //1 bit
out[3] |= (hed.copyright_identification_start << 2 & 0x04); //1 bit
out[3] |= (hed.aac_frame_length >> 11 & 0x03); //2 bit
out[4] = (hed.aac_frame_length >> 3 & 0xFF); //8 bit
out[5] = (hed.aac_frame_length << 5 & 0xE0); //3 bit
out[5] |= (hed.adts_buffer_fullness >> 6 & 0x1F); //5 bit
out[6] = (hed.adts_buffer_fullness << 2 & 0xFC); //6 bit
out[6] |= (hed.no_raw_data_blocks_in_frame & 0x03); //2 bit
}
static void parseAacConfig(const string &config, AdtsHeader &adts) {
uint8_t cfg1 = config[0];
uint8_t cfg2 = config[1];
int audioObjectType;
int sampling_frequency_index;
int channel_configuration;
audioObjectType = cfg1 >> 3;
sampling_frequency_index = ((cfg1 & 0x07) << 1) | (cfg2 >> 7);
channel_configuration = (cfg2 & 0x7F) >> 3;
adts.syncword = 0x0FFF;
adts.id = 0;
adts.layer = 0;
adts.protection_absent = 1;
adts.profile = audioObjectType - 1;
adts.sf_index = sampling_frequency_index;
adts.private_bit = 0;
adts.channel_configuration = channel_configuration;
adts.original = 0;
adts.home = 0;
adts.copyright_identification_bit = 0;
adts.copyright_identification_start = 0;
adts.aac_frame_length = 7;
adts.adts_buffer_fullness = 2047;
adts.no_raw_data_blocks_in_frame = 0;
}
#endif// ENABLE_MP4
int getAacFrameLength(const uint8_t *data, size_t bytes) {
uint16_t len;
if (bytes < 7) return -1;
if (0xFF != data[0] || 0xF0 != (data[1] & 0xF0)) {
return -1;
}
len = ((uint16_t) (data[3] & 0x03) << 11) | ((uint16_t) data[4] << 3) | ((uint16_t) (data[5] >> 5) & 0x07);
return len;
}
string makeAacConfig(const uint8_t *hex, size_t length){
#ifndef ENABLE_MP4
if (!(hex[0] == 0xFF && (hex[1] & 0xF0) == 0xF0)) {
return "";
}
// Get and check the 'profile':
unsigned char profile = (hex[2] & 0xC0) >> 6; // 2 bits
if (profile == 3) {
return "";
}
// Get and check the 'sampling_frequency_index':
unsigned char sampling_frequency_index = (hex[2] & 0x3C) >> 2; // 4 bits
if (samplingFrequencyTable[sampling_frequency_index] == 0) {
return "";
}
// Get and check the 'channel_configuration':
unsigned char channel_configuration = ((hex[2] & 0x01) << 2) | ((hex[3] & 0xC0) >> 6); // 3 bits
unsigned char audioSpecificConfig[2];
unsigned char const audioObjectType = profile + 1;
audioSpecificConfig[0] = (audioObjectType << 3) | (sampling_frequency_index >> 1);
audioSpecificConfig[1] = (sampling_frequency_index << 7) | (channel_configuration << 3);
return string((char *)audioSpecificConfig,2);
#else
struct mpeg4_aac_t aac = {0};
if (mpeg4_aac_adts_load(hex, length, &aac) > 0) {
char buf[32] = {0};
int len = mpeg4_aac_audio_specific_config_save(&aac, (uint8_t *) buf, sizeof(buf));
if (len > 0) {
return string(buf, len);
}
}
WarnL << "生成aac config失败, adts header:" << hexdump(hex, length);
return "";
#endif
}
int dumpAacConfig(const string &config, size_t length, uint8_t *out, size_t out_size) {
#ifndef ENABLE_MP4
AdtsHeader header;
parseAacConfig(config, header);
header.aac_frame_length = (decltype(header.aac_frame_length))(ADTS_HEADER_LEN + length);
dumpAdtsHeader(header, out);
return ADTS_HEADER_LEN;
#else
struct mpeg4_aac_t aac = {0};
int ret = mpeg4_aac_audio_specific_config_load((uint8_t *) config.data(), config.size(), &aac);
if (ret > 0) {
ret = mpeg4_aac_adts_save(&aac, length, out, out_size);
}
if (ret < 0) {
WarnL << "生成adts头失败:" << ret << ", aac config:" << hexdump(config.data(), config.size());
}
return ret;
#endif
}
bool parseAacConfig(const string &config, int &samplerate, int &channels){
#ifndef ENABLE_MP4
AdtsHeader header;
parseAacConfig(config, header);
samplerate = samplingFrequencyTable[header.sf_index];
channels = header.channel_configuration;
return true;
#else
struct mpeg4_aac_t aac = {0};
int ret = mpeg4_aac_audio_specific_config_load((uint8_t *) config.data(), config.size(), &aac);
if (ret > 0) {
samplerate = aac.sampling_frequency;
channels = aac.channels;
return true;
}
WarnL << "获取aac采样率、声道数失败:" << hexdump(config.data(), config.size());
return false;
#endif
}
////////////////////////////////////////////////////////////////////////////////////////////////////
/**
* aac类型SDP
*/
class AACSdp : public Sdp {
public:
/**
* 构造函数
* @param aac_cfg aac两个字节的配置描述
* @param sample_rate 音频采样率
* @param payload_type rtp payload type 默认98
* @param bitrate 比特率
*/
AACSdp(const string &aac_cfg,
int sample_rate,
int channels,
int bitrate = 128,
int payload_type = 98) : Sdp(sample_rate,payload_type){
_printer << "m=audio 0 RTP/AVP " << payload_type << "\r\n";
if (bitrate) {
_printer << "b=AS:" << bitrate << "\r\n";
}
_printer << "a=rtpmap:" << payload_type << " " << getCodecName() << "/" << sample_rate << "/" << channels << "\r\n";
string configStr;
char buf[4] = {0};
for(auto &ch : aac_cfg){
snprintf(buf, sizeof(buf), "%02X", (uint8_t)ch);
configStr.append(buf);
}
_printer << "a=fmtp:" << payload_type << " streamtype=5;profile-level-id=1;mode=AAC-hbr;"
<< "sizelength=13;indexlength=3;indexdeltalength=3;config=" << configStr << "\r\n";
_printer << "a=control:trackID=" << (int)TrackAudio << "\r\n";
}
string getSdp() const override {
return _printer;
}
CodecId getCodecId() const override {
return CodecAAC;
}
private:
_StrPrinter _printer;
};
////////////////////////////////////////////////////////////////////////////////////////////////////
AACTrack::AACTrack(const string &aac_cfg) {
if (aac_cfg.size() < 2) {
throw std::invalid_argument("adts配置必须最少2个字节");
}
_cfg = aac_cfg;
onReady();
}
const string &AACTrack::getAacCfg() const {
return _cfg;
}
CodecId AACTrack::getCodecId() const {
return CodecAAC;
}
bool AACTrack::ready() {
return !_cfg.empty();
}
int AACTrack::getAudioSampleRate() const {
return _sampleRate;
}
int AACTrack::getAudioSampleBit() const {
return _sampleBit;
}
int AACTrack::getAudioChannel() const {
return _channel;
}
void AACTrack::inputFrame(const Frame::Ptr &frame) {
if (frame->prefixSize()) {
//有adts头，尝试分帧
auto ptr = frame->data();
auto end = frame->data() + frame->size();
while (ptr < end) {
auto frame_len = getAacFrameLength((uint8_t *) ptr, end - ptr);
if (frame_len < ADTS_HEADER_LEN) {
break;
}
auto sub_frame = std::make_shared<FrameInternal<FrameFromPtr> >(frame, (char *) ptr, frame_len, ADTS_HEADER_LEN);
ptr += frame_len;
sub_frame->setCodecId(CodecAAC);
inputFrame_l(sub_frame);
}
} else {
inputFrame_l(frame);
}
}
void AACTrack::inputFrame_l(const Frame::Ptr &frame) {
if (_cfg.empty()) {
//未获取到aac_cfg信息
if (frame->prefixSize()) {
//根据7个字节的adts头生成aac config
_cfg = makeAacConfig((uint8_t *) (frame->data()), frame->prefixSize());
onReady();
} else {
WarnL << "无法获取adts头!";
}
}
if (frame->size() > frame->prefixSize()) {
//除adts头外，有实际负载
AudioTrack::inputFrame(frame);
}
}
void AACTrack::onReady() {
if (_cfg.size() < 2) {
return;
}
parseAacConfig(_cfg, _sampleRate, _channel);
}
Track::Ptr AACTrack::clone() {
return std::make_shared<std::remove_reference<decltype(*this)>::type>(*this);
}
Sdp::Ptr AACTrack::getSdp() {
if(!ready()){
WarnL << getCodecName() << " Track未准备好";
return nullptr;
}
return std::make_shared<AACSdp>(getAacCfg(), getAudioSampleRate(), getAudioChannel(), getBitRate() / 1024);
}
}//namespace mediakit