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RTP解析音视频帧
RTP解析H264、AAC负载
RTSP中音视频是通过RTP传输的，本文记录从RTP解析出H264、AAC的过程。
协议介绍可参考 https://blog.csdn.net/lostyears/article/details/51374997
拿到RTP数据后，先去除12字节RTP头部，然后进行下面处理。
解析H264
数据较大的H264包，需要进行RTP分片发送。
实现代码：
/*
* @par pBufIn 待解析RTP(不包含12字节头)
* nLenIn 载荷长度
* pBufOut 装载H264的buf(外部传入，分配空间不小于nLenIn)
* nLenOut 一帧H264数据长度，函数返回true时有效
*
* @return true 一帧结束
* false 分片未结束
*/
bool UnpackRtpH264(const UInt8 *pBufIn, const Int32 nLenIn, UInt8 *pBufOut, Int32& nLenOut)
{
bool bFinished = true;
do
{
nLenOut = 0;
const Int32 eFrameType = pBufIn[0] & 0x1F;
if (eFrameType >= 1 && eFrameType <= 23) //单一NAL单元
{
pBufOut[0] = 0x00;//添加H264四字节头部
pBufOut[1] = 0x00;
pBufOut[2] = 0x00;
pBufOut[3] = 0x01;
memcpy(pBufOut + 4, pBufIn, nLenIn);
nLenOut = nLenIn + 4;
}
else //分片NAL单元，由多个RTP包拼接成完整的NAL单元
{
bFinished = false;
if (pBufIn[1] & 0x80) // 分片Nal单元开始位
{
m_nH264FrmeSize = 0;
pBufOut[0] = 0x00;
pBufOut[1] = 0x00;
pBufOut[2] = 0x00;
pBufOut[3] = 0x01;
pBufOut[4] = ((pBufIn[0] & 0xe0)|(pBufIn[1] & 0x1f));//取pBufIn[0]的前3位 与 pBufIn[1]的后5位
memcpy(pBufOut + 5, pBufIn + 2, nLenIn - 2); //跳过分片RTP的前两字节
m_nH264FrmeSize = nLenIn + 5 - 2;
}
else //后续的Nal单元载荷
{
Assert(m_nH264FrmeSize + nLenIn - 2 <= MAX_FRAME_SISE);
memcpy(pBufOut + m_nH264FrmeSize, pBufIn + 2, nLenIn -2);//跳过分片RTP的前两字节
m_nH264FrmeSize += nLenIn -2;
if (pBufIn[1] & 0x40) // 分片Nal单元结束位
{
nLenOut = m_nH264FrmeSize;
m_nH264FrmeSize = 0;
bFinished = true;
}
}
}
}while (0);
return bFinished;
}
解析AAC
这里要注意，可能是一个RTP包含多个AAC帧，之前按网上找的RTP后直接负载1帧AAC，大部分场景没问题，后面有个输入源解析AAC后没声音，最后发现是一个RTP包含了多个AAC负载。解析协议规范最好还是花时间研读协议规范文档，网上找的博客介绍可能不够全面，导致部分场景失效。
实现代码如下：
/*
* @par pBufIn 待解析RTP(不包含12字节头)
* nLenIn 载荷长度
* pBufOut 装载AAC的buf(外部传入，分配空间不小于nLenIn)
* nLenOut 一帧AAC数据长度，函数返回true时有效
*
* 注：可能一个RTP包中包含多个AAC帧，是通过AU_HEADER_LENGTH(除以8得帧个数)来判断
*
* @return true 一帧结束
* false 分片未结束
*/
bool UnpackRtpAAC(const UInt8 * pBufIn, const Int32 nLenIn, UInt8* pBufOut, Int32& nLenOut)
{
bool bFinished = true;
do
{
nLenOut = 0;
Int32 nAuHeaderOffset = 0;//查询头部的偏移，每次2字节
const UInt16 AU_HEADER_LENGTH = (((pBufIn[nAuHeaderOffset] << 8) | pBufIn[nAuHeaderOffset + 1]) >> 4);//首2字节表示Au-Header的长度，单位bit，所以除以16得到Au-Header字节数
nAuHeaderOffset += 2;
Assert(nLenIn > (2 + AU_HEADER_LENGTH*2));
vector<UInt32 > vecAacFrameLen[AU_HEADER_LENGTH];
for (int i = 0; i < AU_HEADER_LENGTH; ++i)
{
const UInt16 AU_HEADER = ((pBufIn[nAuHeaderOffset] << 8) | pBufIn[nAuHeaderOffset + 1]);//之后的2字节是AU_HEADER
UInt32 nAac = (AU_HEADER >> 3);//其中高13位表示一帧AAC负载的字节长度，低3位无用
vecAacFrameLen->push_back(nAac);
nAuHeaderOffset += 2;
}
const UInt8 *pAacPayload = pBufIn + nAuHeaderOffset;//真正AAC负载开始处
UInt32 nAacPayloadOffset = 0;
for (int j = 0; j < AU_HEADER_LENGTH; ++j)
{
const UInt32 nAac = vecAacFrameLen->at(j);
//生成ADTS头
SAacParam param(nAac, m_AudioInfo.nSample, m_AudioInfo.nChannel);
CADTS adts;
adts.Init(param);
//写入ADTS头
memcpy(pBufOut + nLenOut, adts.GetBuf(), adts.GetBufSize());
nLenOut += adts.GetBufSize();
//写入AAC负载
memcpy(pBufOut + nLenOut, pAacPayload + nAacPayloadOffset, nAac);
nLenOut += nAac;
nAacPayloadOffset += nAac;
}
Assert((nLenIn - nAuHeaderOffset) == nAacPayloadOffset);
} while (0);
return bFinished;
}
封装AAC的ADTS头部
CADTS.h
#ifndef max
#define max(a, b) (((a) > (b)) ? (a) : (b))
#endif
#ifndef min
#define min(a, b) (((a) < (b)) ? (a) : (b))
#endif
#define BYTE_NUMBIT 8 /* bits in byte (char) */
#define N_ADTS_SIZE 7
/*
* 定义是哪个级别的AAC
*/
enum eAACProfile
{
E_AAC_PROFILE_MAIN_PROFILE = 0,
E_AAC_PROFILE_LC,
E_AAC_PROFILE_SSR,
E_AAC_PROFILE_PROFILE_RESERVED,
};
enum eAACSample
{
E_AAC_SAMPLE_96000_HZ = 0,
E_AAC_SAMPLE_88200_HZ,
E_AAC_SAMPLE_64000_HZ,
E_AAC_SAMPLE_48000_HZ,
E_AAC_SAMPLE_44100_HZ,
E_AAC_SAMPLE_32000_HZ,
E_AAC_SAMPLE_24000_HZ,
E_AAC_SAMPLE_22050_HZ,
E_AAC_SAMPLE_16000_HZ,
E_AAC_SAMPLE_12000_HZ,
E_AAC_SAMPLE_11025_HZ,
E_AAC_SAMPLE_8000_HZ,
E_AAC_SAMPLE_7350_HZ,
E_AAC_SAMPLE_RESERVED,
};
enum eAACChannel
{
E_AAC_CHANNEL_SPECIFC_CONFIG = 0,
E_AAC_CHANNEL_MONO,
E_AAC_CHANNEL_STEREO,
E_AAC_CHANNEL_TRIPLE_TRACK,
E_AAC_CHANNEL_4,
E_AAC_CHANNEL_5,
E_AAC_CHANNEL_6,
E_AAC_CHANNEL_8,
E_AAC_CHANNEL_RESERVED,
};
enum eMpegId
{
E_MPEG4 = 0,
E_MPEG_2
};
struct SAacParam
{
SAacParam(UInt32 playod, Int32 sample, Int32 channel = 1, eAACProfile profile = E_AAC_PROFILE_LC, eMpegId id = E_MPEG4)
:eId(id), eProfile(profile), nChannel(channel), nSample(sample), nPlayod(playod)
{
};
eMpegId eId;
eAACProfile eProfile;
Int32 nChannel;
Int32 nSample;
UInt32 nPlayod;//aac负载大小(不包含ADTS头)
};
class CADTS
{
public:
CADTS();
public:
/*
* 初始化函数完成ADTS头的填充
*/
void Init(const SAacParam& aacHead);
/*
* 获取ADTS头地址
*/
UInt8* GetBuf();
/*
* 获取ADTS头长度(字节)
*/
UInt32 GetBufSize() const ;
private:
int PutBit(UInt32 data, int numBit);
int WriteByte(UInt32 data, int numBit);
/*
* 采样率下标
*/
static eAACSample GetSampleIndex(const UInt32 nSample);
/*
* 声道下标
*/
static eAACChannel GetChannelIndex(const UInt32 nChannel);
private:
UInt8 m_pBuf[N_ADTS_SIZE]; //buffer的头指针
const UInt32 m_nBit; //总位数
UInt32 m_curBit; //当前位数
};
CADTS.cpp
CADTS::CADTS():m_pBuf(),m_nBit(BYTE_NUMBIT*N_ADTS_SIZE),m_curBit(0)
{
}
void CADTS::Init(const SAacParam &aacHead)
{
/* Fixed ADTS header */
PutBit(0xFFFF, 12);// 12 bit Syncword
PutBit(aacHead.eId, 1); //ID == 0 for MPEG4 AAC, 1 for MPEG2 AAC
PutBit(0, 2); //layer == 0
PutBit(1, 1); //protection absent
PutBit(aacHead.eProfile, 2); //profile
PutBit(CADTS::GetSampleIndex(aacHead.nSample), 4); //sampling rate
PutBit(0, 1); //private bit
PutBit(CADTS::GetChannelIndex(aacHead.nChannel), 3); //numChannels
PutBit(0, 1); //original/copy
PutBit(0, 1); // home
/* Variable ADTS header */
PutBit(0, 1); // copyr. id. bit
PutBit(0, 1); // copyr. id. start
PutBit(GetBufSize() + aacHead.nPlayod, 13); //ADTS帧的长度包括ADTS头和AAC原始流
PutBit(0x7FF, 11); // buffer fullness (0x7FF for VBR)
PutBit(0 ,2); //raw data blocks (0+1=1)
}
UInt8 *CADTS::GetBuf()
{
return m_pBuf;
}
UInt32 CADTS::GetBufSize() const
{
return m_nBit/BYTE_NUMBIT;
}
int CADTS::PutBit(UInt32 data, int numBit)
{
int num,maxNum,curNum;
unsigned long bits;
if (numBit == 0)
return 0;
/* write bits in packets according to buffer byte boundaries */
num = 0;
maxNum = BYTE_NUMBIT - m_curBit % BYTE_NUMBIT;
while (num < numBit) {
curNum = min(numBit-num,maxNum);
bits = data>>(numBit-num-curNum);
if (WriteByte(bits, curNum)) {
return 1;
}
num += curNum;
maxNum = BYTE_NUMBIT;
}
return 0;
}
int CADTS::WriteByte(UInt32 data, int numBit)
{
long numUsed,idx;
idx = (m_curBit / BYTE_NUMBIT) % N_ADTS_SIZE;
numUsed = m_curBit % BYTE_NUMBIT;
#ifndef DRM
if (numUsed == 0)
m_pBuf[idx] = 0;
#endif
m_pBuf[idx] |= (data & ((1<<numBit)-1)) << (BYTE_NUMBIT-numUsed-numBit);
m_curBit += numBit;
return 0;
}
eAACSample CADTS::GetSampleIndex(const UInt32 nSample)
{
eAACSample eSample = E_AAC_SAMPLE_RESERVED;
static std::map<UInt32 , eAACSample> mpSample;
if (mpSample.empty())
{
mpSample[96000] = E_AAC_SAMPLE_96000_HZ;
mpSample[88200] = E_AAC_SAMPLE_88200_HZ;
mpSample[64000] = E_AAC_SAMPLE_64000_HZ;
mpSample[48000] = E_AAC_SAMPLE_48000_HZ;
mpSample[44100] = E_AAC_SAMPLE_44100_HZ;
mpSample[32000] = E_AAC_SAMPLE_32000_HZ;
mpSample[24000] = E_AAC_SAMPLE_24000_HZ;
mpSample[22050] = E_AAC_SAMPLE_22050_HZ;
mpSample[16000] = E_AAC_SAMPLE_16000_HZ;
mpSample[12000] = E_AAC_SAMPLE_12000_HZ;
mpSample[11025] = E_AAC_SAMPLE_11025_HZ;
mpSample[8000] = E_AAC_SAMPLE_8000_HZ;
mpSample[7350] = E_AAC_SAMPLE_7350_HZ;
};
if (mpSample.find(nSample) != mpSample.end())
{
eSample = mpSample[nSample];
}
return eSample;
}
eAACChannel CADTS::GetChannelIndex(const UInt32 nChannel)
{
eAACChannel eChannel = E_AAC_CHANNEL_RESERVED;
static std::map<UInt32 , eAACChannel> mpChannel;
if (mpChannel.empty())
{
mpChannel[0] = E_AAC_CHANNEL_SPECIFC_CONFIG;
mpChannel[1] = E_AAC_CHANNEL_MONO;
mpChannel[2] = E_AAC_CHANNEL_STEREO;
mpChannel[3] = E_AAC_CHANNEL_TRIPLE_TRACK;
mpChannel[4] = E_AAC_CHANNEL_4;
mpChannel[5] = E_AAC_CHANNEL_5;
mpChannel[6] = E_AAC_CHANNEL_6;
mpChannel[8] = E_AAC_CHANNEL_8;
};
if (mpChannel.find(nChannel) != mpChannel.end())
{
eChannel = mpChannel[nChannel];
}
return eChannel;
}
采坑心得
1、协议解析优先考虑成熟的开源代码，例如ffmpeg，流媒体相关的协议里面基本都有实现；
2、如果找不到成熟开源代码做参考，搜索协议规范文档，不复杂的话照着文档一步步做吧，规范文档比较系统全面，比网上东拼西凑找的靠谱，最后花的时间可能比乱搜一通要少，而且自己解析印象更深刻。
附录：音频抓包分析
