精简aac相关代码

This commit is contained in:
xiongziliang 2020-05-11 23:25:12 +08:00
parent 625d7e30c0
commit 70e9a20352
5 changed files with 120 additions and 150 deletions

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@ -12,62 +12,55 @@
namespace mediakit{
void writeAdtsHeader(const AACFrame &hed, uint8_t *pcAdts) {
pcAdts[0] = (hed.syncword >> 4 & 0xFF); //8bit
pcAdts[1] = (hed.syncword << 4 & 0xF0); //4 bit
pcAdts[1] |= (hed.id << 3 & 0x08); //1 bit
pcAdts[1] |= (hed.layer << 1 & 0x06); //2bit
pcAdts[1] |= (hed.protection_absent & 0x01); //1 bit
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
};
pcAdts[2] = (hed.profile << 6 & 0xC0); // 2 bit
pcAdts[2] |= (hed.sf_index << 2 & 0x3C); //4bit
pcAdts[2] |= (hed.private_bit << 1 & 0x02); //1 bit
pcAdts[2] |= (hed.channel_configuration >> 2 & 0x03); //1 bit
pcAdts[3] = (hed.channel_configuration << 6 & 0xC0); // 2 bit
pcAdts[3] |= (hed.original << 5 & 0x20); //1 bit
pcAdts[3] |= (hed.home << 4 & 0x10); //1 bit
pcAdts[3] |= (hed.copyright_identification_bit << 3 & 0x08); //1 bit
pcAdts[3] |= (hed.copyright_identification_start << 2 & 0x04); //1 bit
pcAdts[3] |= (hed.aac_frame_length >> 11 & 0x03); //2 bit
pcAdts[4] = (hed.aac_frame_length >> 3 & 0xFF); //8 bit
pcAdts[5] = (hed.aac_frame_length << 5 & 0xE0); //3 bit
pcAdts[5] |= (hed.adts_buffer_fullness >> 6 & 0x1F); //5 bit
pcAdts[6] = (hed.adts_buffer_fullness << 2 & 0xFC); //6 bit
pcAdts[6] |= (hed.no_raw_data_blocks_in_frame & 0x03); //2 bit
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
}
string makeAdtsConfig(const uint8_t *pcAdts){
if (!(pcAdts[0] == 0xFF && (pcAdts[1] & 0xF0) == 0xF0)) {
return "";
}
// Get and check the 'profile':
unsigned char profile = (pcAdts[2] & 0xC0) >> 6; // 2 bits
if (profile == 3) {
return "";
}
// Get and check the 'sampling_frequency_index':
unsigned char sampling_frequency_index = (pcAdts[2] & 0x3C) >> 2; // 4 bits
if (samplingFrequencyTable[sampling_frequency_index] == 0) {
return "";
}
// Get and check the 'channel_configuration':
unsigned char channel_configuration = ((pcAdts[2] & 0x01) << 2)
| ((pcAdts[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);
}
void makeAdtsHeader(const string &strAudioCfg,AACFrame &adts) {
uint8_t cfg1 = strAudioCfg[0];
uint8_t cfg2 = strAudioCfg[1];
static void parseAacConfig(const string &config, AdtsHeader &adts) {
uint8_t cfg1 = config[0];
uint8_t cfg2 = config[1];
int audioObjectType;
int sampling_frequency_index;
@ -93,9 +86,44 @@ void makeAdtsHeader(const string &strAudioCfg,AACFrame &adts) {
adts.adts_buffer_fullness = 2047;
adts.no_raw_data_blocks_in_frame = 0;
}
void getAACInfo(const AACFrame &adts,int &iSampleRate,int &iChannel){
iSampleRate = samplingFrequencyTable[adts.sf_index];
iChannel = adts.channel_configuration;
string makeAacConfig(const uint8_t *hex){
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);
}
void dumpAacConfig(const string &config, int length, uint8_t *out){
AdtsHeader header;
parseAacConfig(config, header);
header.aac_frame_length = length;
dumpAdtsHeader(header, out);
}
void parseAacConfig(const string &config, int &samplerate, int &channels){
AdtsHeader header;
parseAacConfig(config, header);
samplerate = samplingFrequencyTable[header.sf_index];
channels = header.channel_configuration;
}
Sdp::Ptr AACTrack::getSdp() {
@ -106,6 +134,4 @@ Sdp::Ptr AACTrack::getSdp() {
return std::make_shared<AACSdp>(getAacCfg(),getAudioSampleRate(), getAudioChannel());
}
}//namespace mediakit
}//namespace mediakit

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@ -13,71 +13,26 @@
#include "Frame.h"
#include "Track.h"
#define ADTS_HEADER_LEN 7
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);
string makeAacConfig(const uint8_t *hex);
void dumpAacConfig(const string &config, int length, uint8_t *out);
void parseAacConfig(const string &config, int &samplerate, int &channels);
/**
* aac帧adts头
*/
class AACFrame : public Frame {
class AACFrame : public FrameImp {
public:
typedef std::shared_ptr<AACFrame> Ptr;
char *data() const override{
return (char *)buffer;
AACFrame(){
_codecid = CodecAAC;
}
uint32_t size() const override {
return aac_frame_length;
}
uint32_t dts() const override {
return _dts;
}
uint32_t prefixSize() const override{
return _prefix_size;
}
CodecId getCodecId() const override{
return CodecAAC;
}
bool keyFrame() const override {
return false;
}
bool configFrame() const override{
return false;
}
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
unsigned char buffer[2 * 1024 + 7];
uint32_t _dts;
uint32_t _prefix_size = 7;
};
class AACFrameNoCacheAble : public FrameFromPtr {
@ -138,7 +93,7 @@ public:
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);
_cfg = makeAacConfig((uint8_t *) adts_header);
onReady();
}
@ -150,7 +105,7 @@ public:
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());
_cfg = makeAacConfig((uint8_t *) aac_frame_with_adts->data());
onReady();
}
@ -205,7 +160,7 @@ public:
//未获取到aac_cfg信息
if (frame->prefixSize() >= 7) {
//7个字节的adts头
_cfg = makeAdtsConfig((uint8_t *)(frame->data()));
_cfg = makeAacConfig((uint8_t *) (frame->data()));
onReady();
} else {
WarnL << "无法获取adts头!";
@ -218,12 +173,10 @@ private:
* 2aac配置
*/
void onReady(){
if(_cfg.size() < 2){
if (_cfg.size() < 2) {
return;
}
AACFrame aacFrame;
makeAdtsHeader(_cfg,aacFrame);
getAACInfo(aacFrame,_sampleRate,_channel);
parseAacConfig(_cfg, _sampleRate, _channel);
}
Track::Ptr clone() override {

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@ -20,8 +20,9 @@ AACRtmpDecoder::AACRtmpDecoder() {
AACFrame::Ptr AACRtmpDecoder::obtainFrame() {
//从缓存池重新申请对象,防止覆盖已经写入环形缓存的对象
auto frame = ResourcePoolHelper<AACFrame>::obtainObj();
frame->aac_frame_length = 7;
frame->_prefix_size = 7;
frame->_prefix_size = ADTS_HEADER_LEN;
//预留7个字节的空位以便后续覆盖
frame->_buffer.assign(ADTS_HEADER_LEN,(char)0);
return frame;
}
@ -41,7 +42,7 @@ static string getAacCfg(const RtmpPacket &thiz) {
return ret;
}
bool AACRtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt, bool key_pos) {
bool AACRtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt, bool) {
if (pkt->isCfgFrame()) {
_aac_cfg = getAacCfg(*pkt);
return false;
@ -52,26 +53,18 @@ bool AACRtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt, bool key_pos) {
return false;
}
void AACRtmpDecoder::onGetAAC(const char* pcData, int iLen, uint32_t ui32TimeStamp) {
if(iLen + 7 > sizeof(_adts->buffer)){
WarnL << "Illegal adts data, exceeding the length limit.";
return;
}
//写adts结构头
makeAdtsHeader(_aac_cfg,*_adts);
//拷贝aac负载
memcpy(_adts->buffer + 7, pcData, iLen);
_adts->aac_frame_length = 7 + iLen;
_adts->_dts = ui32TimeStamp;
//adts结构头转成头7个字节
writeAdtsHeader(*_adts, _adts->buffer);
void AACRtmpDecoder::onGetAAC(const char* data, int len, uint32_t stamp) {
_adts->_dts = stamp;
//先追加数据
_adts->_buffer.append(data, len);
//覆盖adts头
dumpAacConfig(_aac_cfg, _adts->size(), (uint8_t *) _adts->data());
//写入环形缓存
RtmpCodec::inputFrame(_adts);
_adts = obtainFrame();
}
/////////////////////////////////////////////////////////////////////////////////////
AACRtmpEncoder::AACRtmpEncoder(const Track::Ptr &track) {
@ -93,7 +86,7 @@ void AACRtmpEncoder::inputFrame(const Frame::Ptr &frame) {
if (_aac_cfg.empty()) {
if (frame->prefixSize() >= 7) {
//包含adts头,从adts头获取aac配置信息
_aac_cfg = makeAdtsConfig((uint8_t *)(frame->data()));
_aac_cfg = makeAacConfig((uint8_t *) (frame->data()));
}
makeConfigPacket();
}

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@ -38,7 +38,7 @@ public:
}
protected:
void onGetAAC(const char* pcData, int iLen, uint32_t ui32TimeStamp);
void onGetAAC(const char* data, int len, uint32_t stamp);
AACFrame::Ptr obtainFrame();
protected:
AACFrame::Ptr _adts;

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@ -9,7 +9,7 @@
*/
#include "AACRtp.h"
#define ADTS_HEADER_LEN 7
#define AAC_MAX_FRAME_SIZE (2 * 1024)
namespace mediakit{
@ -77,11 +77,9 @@ AACRtpDecoder::AACRtpDecoder() {
AACFrame::Ptr AACRtpDecoder::obtainFrame() {
//从缓存池重新申请对象,防止覆盖已经写入环形缓存的对象
auto frame = ResourcePoolHelper<AACFrame>::obtainObj();
frame->aac_frame_length = ADTS_HEADER_LEN;
frame->_prefix_size = ADTS_HEADER_LEN;
if(frame->syncword == 0 && !_aac_cfg.empty()) {
makeAdtsHeader(_aac_cfg,*frame);
}
//预留7个字节的空位以便后续覆盖
frame->_buffer.assign(ADTS_HEADER_LEN,(char)0);
return frame;
}
@ -96,19 +94,18 @@ bool AACRtpDecoder::inputRtp(const RtpPacket::Ptr &rtppack, bool key_pos) {
//忽略Au-Header区
ptr += 2 + au_header_count * 2;
static const uint32_t max_size = sizeof(AACFrame::buffer) - ADTS_HEADER_LEN;
static const uint32_t max_size = AAC_MAX_FRAME_SIZE - ADTS_HEADER_LEN;
while (ptr < end) {
auto size = (uint32_t) (end - ptr);
if(size > max_size){
size = max_size;
}
if (_adts->aac_frame_length + size > sizeof(AACFrame::buffer)) {
if (_adts->size() + size > AAC_MAX_FRAME_SIZE) {
//数据太多了,先清空
flushData();
}
//追加aac数据
memcpy(_adts->buffer + _adts->aac_frame_length, ptr, size);
_adts->aac_frame_length += size;
_adts->_buffer.append((char *)ptr, size);
_adts->_dts = rtppack->timeStamp;
ptr += size;
}
@ -120,13 +117,14 @@ bool AACRtpDecoder::inputRtp(const RtpPacket::Ptr &rtppack, bool key_pos) {
return false;
}
void AACRtpDecoder::flushData() {
if(_adts->aac_frame_length == ADTS_HEADER_LEN){
if (_adts->size() == ADTS_HEADER_LEN) {
//没有有效数据
return;
}
writeAdtsHeader(*_adts, _adts->buffer);
//覆盖adts头
dumpAacConfig(_aac_cfg, _adts->size(), (uint8_t *) _adts->data());
RtpCodec::inputFrame(_adts);
_adts = obtainFrame();
}