RTP解析音视频帧

版权声明:本文为博主原创文章,未经博主允许不得转载。 https://blog.csdn.net/cugwuhan2014/article/details/84573269

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

在这里插入图片描述

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