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• ffmpeg audio is stuttering when converting from h264 to mpeg2

  9 février 2019, par user11038211

  I want to convert from h264 stream source (streamlink) to mpeg2 video and send it to a DVB-T modulator (Hides UT100C). So far, I have managed to play the video smoothly, but audio is jerky. It sounds like parts are missing. I think that some parameters are not set perfectly :

  $streamlink -O https://zattoo.com/watch/orf-2 best | ffmpeg -re -i pipe:0 -vcodec mpeg2video -s 720x576 -r 30 -b:v 10M -b:a 64k -acodec mp2 -mpegts_original_network_id 1 -mpegts_transport_stream_id 1 -mpegts_service_id 1 -mpegts_pmt_start_pid 1000 -mpegts_start_pid 1001 -metadata service_provider="HOME" -metadata service_name="IPTV" -muxrate 10M -f mpegts -y tspipe | sudo ./tsrfsend tspipe 0 578000 8000 4 1/2 1/4 8 0 0
  
  
  
  [cli][info] Found matching plugin zattoo for URL https://zattoo.com/watch/orf-2
  
  [cli][info] Available streams: 600k_alt (worst), 600k, 900k, 1500k (best)
  
  [cli][info] Opening stream: 1500k (hls)
  
  Input #0, mpegts, from 'pipe:0':
  
    Duration: N/A, start: 45236.000000, bitrate: N/A
  
    Program 1
  
      Stream #0:0[0x100]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(progressive), 768x432 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc
  
      Stream #0:1[0x101]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 128 kb/s
  
  
  
  Open /dev/usb-it950x0 ok
  
  g_ITEAPI_TxDeviceInit ok
  
  1 Devices
  
  g_ITEAPI_GetDrvInfo ok
  
  DriverInfo.DriverVerion  = v16.11.10.1
  
  DriverInfo.APIVerion     = 1.3.20160929.0
  
  DriverInfo.FWVerionLink  = 255.39.2.0
  
  DriverInfo.FWVerionOFDM  = 255.9.11.0
  
  DriverInfo.Company       = ITEtech
  
  DriverInfo.SupportHWInfo = Eagle DVBT
  
  DriverInfo.ProductID     = 0x9507
  
  Frequency = 578000 KHz
  
  Bandwidth = 8000 MHz
  
  Constellation: QPSK
  
  Code Rate: 1/2
  
  Interval: 1/4
  
  Transmission Mode: 8K
  
  Output #0, mpegts, to 'tspipe':
  
    Metadata:
  
      service_provider: HOME
  
      service_name    : IPTV
  
      encoder         : Lavf57.56.101
  
      Stream #0:0: Video: mpeg2video (Main), yuv420p, 720x576 [SAR 64:45 DAR 16:9], q=2-31, 10000 kb/s, 30 fps, 90k tbn, 30 tbc
  
      Metadata:
  
        encoder         : Lavc57.64.101 mpeg2video
  
      Side data:
  
        cpb: bitrate max/min/avg: 0/0/10000000 buffer size: 0 vbv_delay: -1
  
      Stream #0:1: Audio: mp2, 48000 Hz, stereo, s16, 64 kb/s
  
      Metadata:
  
        encoder         : Lavc57.64.101 mp2
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (h264 (native) -> mpeg2video (native))
  
    Stream #0:1 -> #0:1 (aac (native) -> mp2 (native))
  
  Channel Capacity is 4976172 bps
  
  MinGain: -52, MaxGain: 5
  
  frame=    3 fps=0.0 q=2.0 size=      63kB time=00:00:00.38 bitrate=1337.3kbits/s

  Tsrfsend is the application that sends the generated ts stream over a named pipe to the USB modulator.

  I also tried various audio bitrates such as 64, 192, 384k but all is very similar.

• ffmpeg unicast vs multicast bandwidth usage

  14 février 2019, par user2884023

  Assuming there are no errors on this code below to (convert mpeg2.ts to h264) :

  $ ffmpeg
  
  -i playlist1.txt
  
  -vcodec h264
  
  -acodec mp3
  
  -s 1280x720
  
  -b:v 4808k
  
  -b:a 192k
  
  -bufsize 8000k
  
  -maxrate 5000k
  
  -r 30
  
  -muxdelay 0.1
  
  -f mpegts udp_multicast "rtp://hostname[:port]/path"

  If I have qty - 10, h.264 decoders receiving the RTP stream, is the bandwidth consumption (@5mb/s maxrate from FFMPEG) 50mb/s uploaded from my server (unicast) or because of the multicast protocol actually only 5mb/s uploaded / consumed at my server to all 10 decoders which are all capable of receiving the multicast transmission via RTP ?

  Also if I am not erroring by attempting to send the RTP feed out of FFMPEG assuimg my server is acting as a (tx) so it can be picked up directly as input by the h.264 decoders acting as receivers.

  Question : 5mb/s or 50mb/s being uploaded on my server end to 10 decoders with assigned FFMPEG command ?

• very low latency streaming with ffmpeg using a webcam [closed]

  14 août, par userDtrm

  I'm trying to configure ffmpeg to do a real-time video streaming using a webcam. The ffmpeg encoder command I use is as follows.

  



  ffmpeg -f v4l2 -input_format yuyv422 -s 640x480 -i /dev/video0 -c:v libx264 -profile:v baseline -trellis 0 -subq 1 -level 32 -preset superfast -tune zerolatency -me_method epzs -crf 30 -threads 0 -bufsize 1 -refs 4 -coder 0 -b_strategy 0 -bf 0 -sc_threshold 0 -x264-params vbv-maxrate=2000:slice-max-size=1500:keyint=30:min-keyint=10: -pix_fmt yuv420p -an -f mpegts udp://192.168.1.8:5001


  



  The ffplay command used to display the video feed is,

  



  ffplay -analyzeduration 1 -fflags -nobuffer -i udp://192.168.1.8:5001


  



  However, I'm experiencing a latency of 0.5 - 1.0s latency in the video stream. Is there a way to reduce this to a number less than 100ms. Also, when I replace the v4l2 camera capture with a screen capture using x11grab, the stream is almost real-time and I experience no noticeable delays. Moreover, changing the encoder from x264 to mpeg2 had no effect on the latency. In addition, the statistics from the ffmpeg shows that the encoder is performing at a 30fps rate, which I believe indicates that the encoding is real-time. This leaves me with only one reason for the experienced delay.

  



  

  • Is there a significant delay in buffers when using v4l2 during video capturing in a webcam ?

  


  • I don't think the transmission delay is in effect in this case as I see no latencies when screen capture is used under the same conditions.

  


  • Can this latency be further reduced ?. Can someone think of a different encoder configuration to be used instead of the one that I've been using ?