RTP sending PS stream zero copy scheme

The code is still there gitee On , Gradual modification
https://gitee.com/guanzhi0319/rtp

Data reach send RTP

After data collection , code h264, perhaps h265, send out ps Stream or h264 Raw stream to the specified server .

void LiveRTPUdpEncoder::OnCaptureVideoBuffer(uint8_t *data, int len, unsigned int timestamp, bool isKeyframe)
{
    
#define D(x) *(data+x)
	if (SimpleThread::IsStop()) return;

#if 0
	char buffer[32];
	sprintf_s(buffer, "QB:%02x %02x %02x %02x %02x %02x",
		D(0), D(1), D(2), D(3), D(4), D(5));
	OutputDebugStringA(buffer);
	return;
#endif
    uint8_t* odata = NULL;
    int olen = 0;
    uint8_t n = D(4) & 0x1f;
    int isIFrame = n == 0x07;
    switch (v_stream_type)
    {
    
    case en_rtp:
        odata = data;
        olen = len;
        break;
    case en_ps_over_rtp:
        uint8_t* odata;
        int olen;
        v_ps.packagingPs(isIFrame, timestamp, data, len, &odata, &olen);
        break;
    }

	//timestamp = GetTimestamp();
    {
    
        Lock lock(this);
               if(v_stream_type == en_ps_over_rtp)
        {
    
            uint8_t* ps = odata + RTP_UDP_HEAD_LEN;
            olen = olen - RTP_UDP_HEAD_LEN;
            v_rtp.send_video_ps(ps, olen, convertToRTPTimestamp(), isKeyframe);
            delete[]odata;

        }
        else if (v_stream_type == en_rtp)
        {
    
            v_rtp.send_video(odata, olen, convertToRTPTimestamp(), isKeyframe);
        }

        if (!gVideoBegin)
            gVideoBegin = true;
    }
}

send out RTP ps

stay ps Leave blank in front of the stream 12 byte , Every time 1400 When bytes are sent , Change the front 12 byte RTP package , Note that the packet has been modified , Do not reuse , And the transmission must be synchronized to be correct , The benefit is to avoid data copying .
There are two ways to avoid data copying ,
1、 One is the synchronous buffer , But send asynchronously
2、 The synchronous
3、 Use the process , The synchronous .
This is the client side , There is only one way to send , Used UDP, Just send it synchronously . apply ps In order not to copy data in memory , The front is left blank 12 Bytes as RTP UDP The head of . So when sending every other packet , Push forward the data sent 12 byte , Avoid copying , But this changes the data in the buffer , If you want to reuse , Every time you cover it, put 12 Byte assignment , sent , Restore the bytes .

int c_rtp::send_video_ps(uint8_t* data, int len, uint32_t timestamp, bool iskeyframe)
{
    
	//data  There is 12  Byte is empty , structure  ps  Leave blank when streaming 
	uint8_t* x = data;
	int num = (len - 1) / 1400 + 1;
	for (int i = 0; i < num; i++)
	{
    
		uint8_t* pos = x - 12;
		memset(pos, 0, 12);
		RTP_FIXED_HEADER* rtp_hdr;  //RTP Head 
		rtp_hdr = (RTP_FIXED_HEADER*)data;
		// Set up RTP HEADER,
		rtp_hdr->payload = H264;  // Load type number ,
		rtp_hdr->version = 2;  // Version number , This version is fixed to 2
		rtp_hdr->marker = 0;   // Sign a , Its value is specified by specific agreement .
		rtp_hdr->ssrc = little2big(1111);
		rtp_hdr->timestamp = little2big(timestamp);
		if (i < num-1)
			v_sock.SendTo(pos, 12 + 1400, v_sockaddr);
		else
			v_sock.SendTo(pos, 12 + len -1400*i,v_sockaddr);
		x += 1400;
	}

}

there v_sock It is simply synchronous sending socket. Readers are interested in , You can change to asynchronous mode by yourself .