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• Web camera Logitech and Linux

  8 novembre 2019, par Nick Saw

  I have Logitech C310 camera with the declared characteristics of 720p 30fps.

  If you connect the camera to windows, the recording is fully consistent with the stated 720p 30fps - the picture is clear.

  The challenge is to connect the same camera to OrangePI (server Armbian) and to save video files on it.

  The camera appears as /dev/video0.

  sudo ffmpeg -f v4l2 -s 1280x720 -i /dev/video0 output.wmv

  As a result, I get a crumbly picture with a frequency of 5 fps.

  Maybe I’m using ffmpeg incorrectly ? Please help me who has experience with Web cameras on Linux ...
  Thanks in advance.

  USB-camera configuration :

  v4l2-ctl --all --device=/dev/video0
  
  
  
  
  
  Driver Info (not using libv4l2):
  
          Driver name   : uvcvideo
  
          Card type     : UVC Camera (046d:081b)
  
          Bus info      : usb-1c1c000.usb-1
  
          Driver version: 4.14.18
  
          Capabilities  : 0x84200001
  
                  Video Capture
  
                  Streaming
  
                  Extended Pix Format
  
                  Device Capabilities
  
          Device Caps   : 0x04200001
  
                  Video Capture
  
                  Streaming
  
                  Extended Pix Format
  
  Priority: 2
  
  Video input : 0 (Camera 1: ok)
  
  Format Video Capture:
  
          Width/Height      : 1280/720
  
          Pixel Format      : 'YUYV'
  
          Field             : None
  
          Bytes per Line    : 2560
  
          Size Image        : 1843200
  
          Colorspace        : sRGB
  
          Transfer Function : Default
  
          YCbCr/HSV Encoding: Default
  
          Quantization      : Default
  
          Flags             :
  
  Crop Capability Video Capture:
  
          Bounds      : Left 0, Top 0, Width 1280, Height 720
  
          Default     : Left 0, Top 0, Width 1280, Height 720
  
          Pixel Aspect: 1/1
  
  Selection: crop_default, Left 0, Top 0, Width 1280, Height 720
  
  Selection: crop_bounds, Left 0, Top 0, Width 1280, Height 720
  
  Streaming Parameters Video Capture:
  
          Capabilities     : timeperframe
  
          Frames per second: 5.000 (5/1)
  
          Read buffers     : 0
  
                       brightness (int)    : min=0 max=255 step=1 default=128 value=128
  
                         contrast (int)    : min=0 max=255 step=1 default=32 value=32
  
                       saturation (int)    : min=0 max=255 step=1 default=32 value=32
  
   white_balance_temperature_auto (bool)   : default=1 value=1
  
                             gain (int)    : min=0 max=255 step=1 default=64 value=192
  
             power_line_frequency (menu)   : min=0 max=2 default=2 value=2
  
        white_balance_temperature (int)    : min=0 max=10000 step=10 default=4000 value=4610 flags=inactive
  
                        sharpness (int)    : min=0 max=255 step=1 default=24 value=24
  
           backlight_compensation (int)    : min=0 max=1 step=1 default=0 value=0
  
                    exposure_auto (menu)   : min=0 max=3 default=3 value=3
  
                exposure_absolute (int)    : min=1 max=10000 step=1 default=166 value=249 flags=inactive
  
           exposure_auto_priority (bool)   : default=0 value=1
  
                        led1_mode (menu)   : min=0 max=3 default=3 value=3
  
                   led1_frequency (int)    : min=0 max=131 step=1 default=0 value=0

• How to Skip frames while decoding H264 stream ?

  16 septembre 2018, par TTGroup

  I’m using FFMPEG to decode H264 (or H265) RTSP Stream.

  My system have 2 software : Server and Client

  Server: Read frames from RTSP stream --> Forward frames to Client    
  
  Client: Receive frames from Server --> Decode --> Render

  I have implemented and it worked ok, but there is a case make my system work not good. That is when internet from Server - Client is slow, frames can not transfer real-time to Client.

  In present, I deal with this issue by Skip some frames (not send to Client) when the Queue is reached limit of count. The following is my summary code

  //At Server Software (include 2 threads A and B)
  
  //Thread A: Read AVPacket and forward to Client
  
  while(true)
  
  {
  
      AVPacket packet;
  
      av_init_packet(&packet);
  
      packet.size = 0;
  
      packet.data = NULL;
  
      int ret = AVERROR(EAGAIN);
  
      while (AVERROR(EAGAIN) == ret)
  
          ret = av_read_frame(pFormatCtx, &packet);
  
      if(packet.size > 0)
  
      {
  
          if(mySendQueue.count < 120) //limit 120 packet in queue
  
              mySendQueue.Enqueue(packet); ////Thread B will read from this queue, to send packets to Client via TCP socket
  
          else
  
              ;//SkipThisFrame ***: No send
  
      }
  
  }
  
  //Thread B: Send To Client via TCP Socket
  
  While(true)
  
  {
  
      AVPacket packet;
  
      if(mySendQueue.Dequeue(packet))
  
      {
  
          SendPacketToClient(packet);
  
      }
  
  }
  
  
  
  //At Server Software : Receive AVPacket from Server --> Decode --> Render
  
  While(true)
  
  {
  
      AVPacket packet;
  
      AVFrame frame;
  
      ReadPacketFromServer(packet);
  
      if (av_decode_asyn(pCodecCtx, &frame, &frameFinished, &packet) == RS_OK)
  
      {
  
          if (frameFinished)
  
          {
  
              RenderFrame(frame);
  
          }
  
      }           
  
  }
  
  UINT32 __clrcall av_decode_asyn(AVCodecContext *pCodecCtx, AVFrame *frame, int *frameFinished, AVPacket *packet)
  
  {
  
      int ret = -1;
  
      *frameFinished = 0;
  
      if (packet) 
  
      {
  
          ret = avcodec_send_packet(pCodecCtx, packet);
  
          // In particular, we don't expect AVERROR(EAGAIN), because we read all
  
          // decoded frames with avcodec_receive_frame() until done.
  
          if (ret < 0 && ret != AVERROR_EOF)
  
              return RS_NOT_OK;
  
      }
  
  
  
      ret = avcodec_receive_frame(pCodecCtx, frame);
  
      if (ret < 0 && ret != AVERROR(EAGAIN))
  
      {
  
          return RS_NOT_OK;
  
      }
  
      if (ret >= 0)
  
          *frameFinished = 1;
  
  
  
      return RS_OK;
  
  }

  My question is focus in line of code SkipThisFrame ***, this algorithm skip frame continuously, so it maybe make the decoder on Client occur unexpectedly error or Crash ?

  And when skip frame like that, make Client Render frames is not normally ?

  And someone call show me the proper algorithm to skip frames in my case ?

  Thank you very much !

• RTMP Broadcast packet body structure for Twitch

  22 mai 2018, par Dobby

  I’m currently working on a project similar to OBS, where I’m capturing screen data, encoding it with the x264 library, and then broadcasting it to a twitch server.

  Currently, the servers are accepting the data, but no video is being played - it buffers for a moment, then returns an error code "2000 : network error"

  Like OBS Classic, I’m dividing each NAL provided by x264 by its type, and then making changes to each

  int frame_size = x264_encoder_encode(encoder, &nals, &num_nals, &pic_in, &pic_out);
  
  
  
      //sort the NAL's into their types and make necessary adjustments
  
  
  
      int timeOffset = int(pic_out.i_pts - pic_out.i_dts);
  
  
  
      timeOffset = htonl(timeOffset);//host to network translation, ensure the bytes are in the right format
  
      BYTE *timeOffsetAddr = ((BYTE*)&timeOffset) + 1;
  
  
  
      videoSection sect;
  
      bool foundFrame = false;
  
  
  
      uint8_t * spsPayload = NULL;
  
      int spsSize = 0;
  
  
  
      for (int i = 0; i/std::cout << "VideoEncoder: EncodedImages Size: " << encodedImages->size() << std::endl;
  
          x264_nal_t &nal = nals[i];
  
          //std::cout << "NAL is:" << nal.i_type << std::endl;
  
  
  
          //need to account for pps/sps, seems to always be the first frame sent
  
          if (nal.i_type == NAL_SPS) {
  
              spsSize = nal.i_payload;
  
              spsPayload = (uint8_t*)malloc(spsSize);
  
              memcpy(spsPayload, nal.p_payload, spsSize);
  
          } else if (nal.i_type == NAL_PPS){
  
              //pps always happens after sps
  
              if (spsPayload == NULL) {
  
                  std::cout << "VideoEncoder: critical error, sps not set" << std::endl;
  
              }
  
              uint8_t * payload = (uint8_t*)malloc(nal.i_payload + spsSize);
  
              memcpy(payload, spsPayload, spsSize);
  
              memcpy(payload, nal.p_payload + spsSize, nal.i_payload);
  
              sect = { nal.i_payload + spsSize, payload, nal.i_type };
  
              encodedImages->push(sect);
  
          } else if (nal.i_type == NAL_SEI || nal.i_type == NAL_FILLER) { 
  
              //these need some bytes at the start removed
  
              BYTE *skip = nal.p_payload;
  
              while (*(skip++) != 0x1);
  
              int skipBytes = (int)(skip - nal.p_payload);
  
  
  
              int newPayloadSize = (nal.i_payload - skipBytes);
  
  
  
              uint8_t * payload = (uint8_t*)malloc(newPayloadSize);
  
              memcpy(payload, nal.p_payload + skipBytes, newPayloadSize);
  
              sect = { newPayloadSize, payload, nal.i_type };
  
              encodedImages->push(sect);
  
  
  
          } else if (nal.i_type == NAL_SLICE_IDR || nal.i_type == NAL_SLICE) { 
  
              //these packets need an additional section at the start
  
              BYTE *skip = nal.p_payload;
  
              while (*(skip++) != 0x1);
  
              int skipBytes = (int)(skip - nal.p_payload);
  
  
  
              std::vector<byte> bodyData;
  
              if (!foundFrame) {
  
                  if (nal.i_type == NAL_SLICE_IDR) { bodyData.push_back(0x17); } else { bodyData.push_back(0x27); } //add a 17 or a 27 as appropriate
  
                  bodyData.push_back(1);
  
                  bodyData.push_back(*timeOffsetAddr);
  
  
  
                  foundFrame = true;
  
              }
  
  
  
              //put into the payload the bodyData followed by the nal payload
  
              uint8_t * bodyDataPayload = (uint8_t*)malloc(bodyData.size());
  
              memcpy(bodyDataPayload, bodyData.data(), bodyData.size() * sizeof(BYTE));
  
  
  
              int newPayloadSize = (nal.i_payload - skipBytes);
  
  
  
              uint8_t * payload = (uint8_t*)malloc(newPayloadSize + sizeof(bodyDataPayload));
  
              memcpy(payload, bodyDataPayload, sizeof(bodyDataPayload));
  
              memcpy(payload + sizeof(bodyDataPayload), nal.p_payload + skipBytes, newPayloadSize);
  
              int totalSize = newPayloadSize + sizeof(bodyDataPayload);
  
              sect = { totalSize, payload, nal.i_type };
  
              encodedImages->push(sect);
  
          } else {
  
              std::cout &lt;&lt; "VideoEncoder: Nal type did not match expected" &lt;&lt; std::endl;
  
              continue;
  
          }
  
      }
  
  </byte>

  The NAL payload data is then put into a struct, VideoSection, in a queue buffer

  //used to transfer encoded data
  
  struct videoSection {
  
      int frameSize;
  
      uint8_t* payload;
  
      int type;
  
  };

  After which it is picked up by the broadcaster, a few more changes are made, and then I call rtmp_send()

  videoSection sect = encodedImages->front();
  
  encodedImages->pop();
  
  
  
  //std::cout << "Broadcaster: Frame Size: " << sect.frameSize << std::endl;
  
  
  
  //two methods of sending RTMP data, _sendpacket and _write. Using sendpacket for greater control
  
  
  
  RTMPPacket * packet;
  
  
  
  unsigned char* buf = (unsigned char*)sect.payload;
  
  
  
  int type = buf[0]&0x1f; //I believe &0x1f sets a 32bit limit
  
  int len = sect.frameSize;
  
  long timeOffset = GetTickCount() - rtmp_start_time;
  
  
  
  //assign space packet will need
  
  packet = (RTMPPacket *)malloc(sizeof(RTMPPacket)+RTMP_MAX_HEADER_SIZE + len + 9);
  
  memset(packet, 0, sizeof(RTMPPacket) + RTMP_MAX_HEADER_SIZE);
  
  
  
  packet->m_body = (char *)packet + sizeof(RTMPPacket) + RTMP_MAX_HEADER_SIZE;
  
  packet->m_nBodySize = len + 9;
  
  
  
  //std::cout << "Broadcaster: Packet Size: " << sizeof(RTMPPacket) + RTMP_MAX_HEADER_SIZE + len + 9 << std::endl;
  
  //std::cout << "Broadcaster: Packet Body Size: " << len + 9 << std::endl;
  
  
  
  //set body to point to the packetbody
  
  unsigned char *body = (unsigned char *)packet->m_body;
  
  memset(body, 0, len + 9);
  
  
  
  
  
  
  
  //NAL_SLICE_IDR represents keyframe
  
  //first element determines packet type
  
  body[0] = 0x27;//inter-frame h.264
  
  if (sect.type == NAL_SLICE_IDR) {
  
      body[0] = 0x17; //h.264 codec id
  
  }
  
  
  
  
  
  //-------------------------------------------------------------------------------
  
  //this section taken from https://stackoverflow.com/questions/25031759/using-x264-and-librtmp-to-send-live-camera-frame-but-the-flash-cant-show
  
  //in an effort to understand packet format. it does not resolve my previous issues formatting the data for twitch to play it
  
  
  
  //sets body to be NAL unit
  
  body[1] = 0x01;
  
  body[2] = 0x00;
  
  body[3] = 0x00;
  
  body[4] = 0x00;
  
  
  
  //>> is a shift right
  
  //shift len to the right, and AND it
  
  /*body[5] = (len >> 24) & 0xff;
  
  body[6] = (len >> 16) & 0xff;
  
  body[7] = (len >> 8) & 0xff;
  
  body[8] = (len) & 0xff;*/
  
  
  
  //end code sourced from https://stackoverflow.com/questions/25031759/using-x264-and-librtmp-to-send-live-camera-frame-but-the-flash-cant-show
  
  //-------------------------------------------------------------------------------
  
  
  
  //copy from buffer into rest of body
  
  memcpy(&body[9], buf, len);
  
  
  
  //DEBUG
  
  
  
  //save individual packet body to a file with name rtmp[packetnum]
  
  //determine why some packets do not have 0x27 or 0x17 at the start
  
  //still happening, makes no sense given the above code
  
  
  
  /*std::string fileLocation = "rtmp" + std::to_string(packCount++);
  
  std::cout << fileLocation << std::endl;
  
  const char * charConversion = fileLocation.c_str();
  
  
  
  FILE* saveFile = NULL;
  
  saveFile = fopen(charConversion, "w+b");//open as write and binary
  
  if (!fwrite(body, len + 9, 1, saveFile)) {
  
      std::cout << "VideoEncoder: Error while trying to write to file" << std::endl;
  
  }
  
  fclose(saveFile);*/
  
  
  
  //END DEBUG
  
  
  
  //other packet details
  
  packet->m_hasAbsTimestamp = 0;
  
  packet->m_packetType = RTMP_PACKET_TYPE_VIDEO;
  
  if (rtmp != NULL) {
  
      packet->m_nInfoField2 = rtmp->m_stream_id;
  
  }
  
  packet->m_nChannel = 0x04;
  
  packet->m_headerType = RTMP_PACKET_SIZE_LARGE;
  
  packet->m_nTimeStamp = timeOffset;
  
  
  
  //send the packet
  
  if (rtmp != NULL) {
  
      RTMP_SendPacket(rtmp, packet, TRUE);
  
  }

  I can see that Twitch is receiving the data in the inspector, at a steady 3kbps. so I’m sure something is wrong with how I’m adjusting the data before sending it. Can anyone advise me on what I’m doing wrong here ?