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• ffmpeg can't stream to remote client

  4 septembre 2014, par KFL

  I’m building a simple ffmpeg command line on my laptop to stream from its camera. The command line reads (verbose output at the botton) :

  host1> ffmpeg -v verbose \
  
                -f dshow \
  
                -i video="Camera":audio="Microphone" \
  
                -r 30 -g 0 -vcodec h264 -acodec libmp3lame \
  
                -tune zerolatency \
  
                -preset ultrafast \
  
                -f mpegts udp://12.34.56.78:12345

  Firstly, it works locally. I.e., I can view the output by using ffplay on the same host :

  host1> ffplay -hide_banner -v udp://12.34.56.78:12345

  Now what is NOT working is when I do this from another machine in the same network. It shows a nan progress :

  host2> ffplay -hide_banner -v udp://12.34.56.78:12345
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   

  I used ncat to dump the raw content. But there’s no output :

  host2>\ncat\ncat -v -u 12.34.56.78 12345
  
  Ncat: Version 5.59BETA1 ( http://nmap.org/ncat )
  
  Ncat: Connected to 12.34.56.78:12345.
  
  (...and nothing happen...)

  Note that I can exclude firewall issues as I used ncat to communicate with each other across the wire using the same port and protocol (UDP). This works and they can chat to each other :

  host1> ncat -l -u -p 12345
  
  host2> ncat -u 12.34.56.78 12345

  Any hint ?

  I’m using Windows x64 with FFMPEG 64bit installed from here. Below is the Output of my ffmpeg command :

  C:\ffmpeg\bin>ffmpeg -v verbose -f dshow -i video="Integrated Camera":audio="Microphone (Realtek High Definition Audio)" -r 30 -g 0 -vcodec h264 -acodec libmp3lame -tune zerolatency -preset ultrafast -f mpegts udp://12.34.56.78:12345
  
  ffmpeg version N-66012-g97b8809 Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Sep  1 2014 00:21:15 with gcc 4.8.3 (GCC)
  
    configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug -enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib
  
    libavutil      54.  7.100 / 54.  7.100
  
    libavcodec     56.  1.100 / 56.  1.100
  
    libavformat    56.  3.100 / 56.  3.100
  
    libavdevice    56.  0.100 / 56.  0.100
  
    libavfilter     5.  0.103 /  5.  0.103
  
    libswscale      3.  0.100 /  3.  0.100
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  0.100 / 53.  0.100
  
  Guessed Channel Layout for  Input Stream #0.1 : stereo
  
  Input #0, dshow, from 'video=Integrated Camera:audio=Microphone (Realtek High Definition Audio)':
  
    Duration: N/A, start: 171840.657000, bitrate: N/A
  
      Stream #0:0: Video: rawvideo, bgr24, 640x480, 30 fps, 30 tbr, 10000k tbn, 30 tbc
  
      Stream #0:1: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s
  
  Matched encoder 'libx264' for codec 'h264'.
  
  [graph 0 input from stream 0:0 @ 0000000000470aa0] w:640 h:480 pixfmt:bgr24 tb:1/10000000 fr:10000000/333333 sar:0/1 sws_param:flags=2
  
  [auto-inserted scaler 0 @ 0000000004326d00] w:iw h:ih flags:'0x4' interl:0
  
  [format @ 0000000004325a00] auto-inserting filter 'auto-inserted scaler 0' between the filter 'Parsed_null_0' and the filter 'format'
  
  [auto-inserted scaler 0 @ 0000000004326d00] w:640 h:480 fmt:bgr24 sar:0/1 -> w:640 h:480 fmt:yuv444p sar:0/1 flags:0x4
  
  No pixel format specified, yuv444p for H.264 encoding chosen.
  
  Use -pix_fmt yuv420p for compatibility with outdated media players.
  
  [graph 1 input from stream 0:1 @ 0000000000460c20] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3
  
  [audio format for output stream 0:1 @ 00000000004601a0] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:1'
  
  [auto-inserted resampler 0 @ 00000000004604a0] ch:2 chl:stereo fmt:s16 r:44100Hz -> ch:2 chl:stereo fmt:s16p r:44100Hz
  
  [libx264 @ 000000000081bb20] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
  
  [libx264 @ 000000000081bb20] profile High 4:4:4 Intra, level 3.0, 4:4:4 8-bit
  
  [mpegts @ 000000000081abe0] muxrate VBR, pcr every 3 pkts, sdt every 200, pat/pmt every 40 pkts
  
  Output #0, mpegts, to 'udp://12.34.56.78:12345':
  
    Metadata:
  
      encoder         : Lavf56.3.100
  
      Stream #0:0: Video: h264 (libx264), yuv444p, 640x480, q=-1--1, 30 fps, 90k tbn, 30 tbc
  
      Metadata:
  
        encoder         : Lavc56.1.100 libx264
  
      Stream #0:1: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p
  
      Metadata:
  
        encoder         : Lavc56.1.100 libmp3lame
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
  
    Stream #0:1 -> #0:1 (pcm_s16le (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  *** 1 dup!
  
  frame=  241 fps= 31 q=28.0 Lsize=    3439kB time=00:00:08.03 bitrate=3506.4kbits/s dup=1 drop=0
  
  video:3035kB audio:125kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 8.791966%
  
  Input file #0 (video=Integrated Camera:audio=Microphone (Realtek High Definition Audio)):
  
    Input stream #0:0 (video): 240 packets read (221184000 bytes); 240 frames decoded;
  
    Input stream #0:1 (audio): 16 packets read (1411200 bytes); 16 frames decoded (352800 samples);
  
    Total: 256 packets (222595200 bytes) demuxed
  
  Output file #0 (udp://12.34.56.78:12345):
  
    Output stream #0:0 (video): 241 frames encoded; 241 packets muxed (3108187 bytes);
  
    Output stream #0:1 (audio): 306 frames encoded (352512 samples); 307 packets muxed (128313 bytes);
  
    Total: 548 packets (3236500 bytes) muxed
  
  [libx264 @ 000000000081bb20] frame I:241   Avg QP:27.97  size: 12897
  
  [libx264 @ 000000000081bb20] mb I  I16..4: 100.0%  0.0%  0.0%
  
  [libx264 @ 000000000081bb20] coded y,u,v intra: 26.3% 0.5% 0.0%
  
  [libx264 @ 000000000081bb20] i16 v,h,dc,p: 19% 28% 21% 31%
  
  [libx264 @ 000000000081bb20] kb/s:3095.29
  
  [dshow @ 0000000000467720] real-time buffer[Integrated Camera] too full (90% of size: 3041280)! frame dropped!
  
  Received signal 2: terminating. (I pressed CTRL-C)

• Troubleshooting ffmpeg/ffplay client RTSP RTP UDP * multicast * issue

  6 novembre 2020, par MAXdB

  I'm having problem with using udp_multicast transport method using ffmpeg or ffplay as a client to a webcam.

  


  TCP transport works :

  


  ffplay -rtsp_transport tcp rtsp://192.168.1.100/videoinput_1/mjpeg_3/media.stm


  


  UDP transport works :

  


  ffplay -rtsp_transport udp rtsp://192.168.1.100/videoinput_1/mjpeg_3/media.stm


  


  Multicast transport does not work :

  


  ffplay -rtsp_transport udp_multicast rtsp://192.168.1.100/videoinput_1/mjpeg_3/media.stm


  


  The error message when udp_multicast is chosen reads :

  


  [rtsp @ 0x7fd6a8000b80] Could not find codec parameters for stream 0 (Video: mjpeg, none(bt470bg/unknown/unknown)): unspecified size


  


  Run with -v debug : Observe that the UDP multicast information appears in the SDP even though the chosen transport is unicast for this run. The SDP content is unchanged for unicast or multicast.

  


  [tcp @ 0x7f648c002f40] Starting connection attempt to 192.168.1.100 port 554
[tcp @ 0x7f648c002f40] Successfully connected to 192.168.1.100 port 554
[rtsp @ 0x7f648c000b80] SDP:
v=0
o=- 621355968671884050 621355968671884050 IN IP4 192.168.1.100
s=/videoinput_1:0/mjpeg_3/media.stm
c=IN IP4 0.0.0.0
m=video 40004 RTP/AVP 26
c=IN IP4 237.0.0.3/1
a=control:trackID=1
a=range:npt=0-
a=framerate:25.0

Failed to parse interval end specification ''
[rtp @ 0x7f648c008e00] No default whitelist set
[udp @ 0x7f648c009900] No default whitelist set
[udp @ 0x7f648c009900] end receive buffer size reported is 425984
[udp @ 0x7f648c019c80] No default whitelist set
[udp @ 0x7f648c019c80] end receive buffer size reported is 425984
[rtsp @ 0x7f648c000b80] setting jitter buffer size to 500
[rtsp @ 0x7f648c000b80] hello state=0
Failed to parse interval end specification ''
[mjpeg @ 0x7f648c0046c0] marker=d8 avail_size_in_buf=145103 
[mjpeg @ 0x7f648c0046c0] marker parser used 0 bytes (0 bits)
[mjpeg @ 0x7f648c0046c0] marker=e0 avail_size_in_buf=145101
[mjpeg @ 0x7f648c0046c0] marker parser used 16 bytes (128 bits)
[mjpeg @ 0x7f648c0046c0] marker=db avail_size_in_buf=145083
[mjpeg @ 0x7f648c0046c0] index=0
[mjpeg @ 0x7f648c0046c0] qscale[0]: 5
[mjpeg @ 0x7f648c0046c0] index=1
[mjpeg @ 0x7f648c0046c0] qscale[1]: 10
[mjpeg @ 0x7f648c0046c0] marker parser used 132 bytes (1056 bits)
[mjpeg @ 0x7f648c0046c0] marker=c4 avail_size_in_buf=144949
[mjpeg @ 0x7f648c0046c0] marker parser used 0 bytes (0 bits)
[mjpeg @ 0x7f648c0046c0] marker=c0 avail_size_in_buf=144529
[mjpeg @ 0x7f648c0046c0] Changing bps from 0 to 8
[mjpeg @ 0x7f648c0046c0] sof0: picture: 1920x1080
[mjpeg @ 0x7f648c0046c0] component 0 2:2 id: 0 quant:0
[mjpeg @ 0x7f648c0046c0] component 1 1:1 id: 1 quant:1
[mjpeg @ 0x7f648c0046c0] component 2 1:1 id: 2 quant:1
[mjpeg @ 0x7f648c0046c0] pix fmt id 22111100
[mjpeg @ 0x7f648c0046c0] Format yuvj420p chosen by get_format().
[mjpeg @ 0x7f648c0046c0] marker parser used 17 bytes (136 bits)
[mjpeg @ 0x7f648c0046c0] escaping removed 676 bytes
[mjpeg @ 0x7f648c0046c0] marker=da avail_size_in_buf=144510
[mjpeg @ 0x7f648c0046c0] marker parser used 143834 bytes (1150672 bits)
[mjpeg @ 0x7f648c0046c0] marker=d9 avail_size_in_buf=2
[mjpeg @ 0x7f648c0046c0] decode frame unused 2 bytes
[rtsp @ 0x7f648c000b80] All info found vq=    0KB sq=    0B f=0/0
[rtsp @ 0x7f648c000b80] rfps: 24.416667 0.018101
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 24.500000 0.013298
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 24.583333 0.009235
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 24.666667 0.005910
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 24.750000 0.003324
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 24.833333 0.001477
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 24.916667 0.000369
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 25.000000 0.000000
[rtsp @ 0x7f648c000b80] rfps: 25.083333 0.000370
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 25.166667 0.001478
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 25.250000 0.003326
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 25.333333 0.005912
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 25.416667 0.009238
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 25.500000 0.013302
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 25.583333 0.018105
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 50.000000 0.000000
[rtsp @ 0x7f648c000b80] Setting avg frame rate based on r frame rate
Input #0, rtsp, from 'rtsp://192.168.1.100/videoinput_1/mjpeg_3/media.stm':
  Metadata:
    title           : /videoinput_1:0/mjpeg_3/media.stm
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0, 21, 1/90000: Video: mjpeg (Baseline), 1 reference frame, yuvj420p(pc, bt470bg/unknown/unknown, center), 1920x1080 [SAR 1:1 DAR 16:9], 0/1, 25 fps, 25 tbr, 90k tbn, 90k tbc
[mjpeg @ 0x7f648c02ad80] marker=d8 avail_size_in_buf=145103


  


  Here is the same debug section when using udp_multicast. The SDP is identical as mentioned, and the block after the SDP containing [mjpeg] codec info is entirely missing (beginning with marker=d8)—the stream is never identified. This happens (to the eye) instantaneously, there's no indication of a timeout waiting unsuccessfully for an RTP packet, though this, too, could just be insufficient debug info in the driver. Also note that ffmpeg knows that the frames are MJPEG frames and the color primaries are PAL, it just doesn't know the size. Also curious, but not relevant to the problem, the unicast UDP transport destination port utilized for the stream does not appear in the ffmpeg debug dump shown above, meaning part of the RTSP/RTP driver is hiding important information under the kimono, that port number and how it knows that the frames will be MJPEG.

  


  [tcp @ 0x7effe0002f40] Starting connection attempt to 192.168.1.100 port 554
[tcp @ 0x7effe0002f40] Successfully connected to 192.168.1.100 port 554
[rtsp @ 0x7effe0000b80] SDP:aq=    0KB vq=    0KB sq=    0B f=0/0
v=0
o=- 621355968671884050 621355968671884050 IN IP4 192.168.1.100
s=/videoinput_1:0/mjpeg_3/media.stm
c=IN IP4 0.0.0.0
m=video 40004 RTP/AVP 26
c=IN IP4 237.0.0.3/1
a=control:trackID=1
a=range:npt=0-
a=framerate:25.0

Failed to parse interval end specification ''
[rtp @ 0x7effe0008e00] No default whitelist set
[udp @ 0x7effe0009900] No default whitelist set
[udp @ 0x7effe0009900] end receive buffer size reported is 425984
[udp @ 0x7effe0019c40] No default whitelist set
[udp @ 0x7effe0019c40] end receive buffer size reported is 425984
[rtsp @ 0x7effe0000b80] setting jitter buffer size to 500
[rtsp @ 0x7effe0000b80] hello state=0
Failed to parse interval end specification '' 
[rtsp @ 0x7effe0000b80] Could not find codec parameters for stream 0 (Video: mjpeg, 1 reference frame, none(bt470bg/unknown/unknown, center)): unspecified size
Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options
Input #0, rtsp, from 'rtsp://192.168.1.100/videoinput_1/mjpeg_3/media.stm':
  Metadata:
    title           : /videoinput_1:0/mjpeg_3/media.stm
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0, 0, 1/90000: Video: mjpeg, 1 reference frame, none(bt470bg/unknown/unknown, center), 90k tbr, 90k tbn, 90k tbc
    nan M-V:    nan fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0


  


  This is the TCPDUMP of the traffic. The information in both streams appears identical.

  


  19:21:30.703599 IP 192.168.1.100.64271 > 192.168.1.98.5239: UDP, length 60
19:21:30.703734 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 1400
19:21:30.703852 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 1400
19:21:30.704326 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 1400
19:21:30.704326 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 1400
19:21:30.704327 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 1400
19:21:30.704327 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 1400
19:21:30.704504 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 1400
19:21:30.704813 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 1400
19:21:30.704814 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 1400
19:21:30.704872 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 732
19:21:30.704873 IP 192.168.1.100.59869 > 237.0.0.3.40005: UDP, length 60
19:21:30.705513 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 1400
19:21:30.705513 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 1400
19:21:30.705513 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 1400
19:21:30.705513 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 1400
19:21:30.705594 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 1400
19:21:30.705774 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 1400
19:21:30.706236 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 1400
19:21:30.706236 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 1400
19:21:30.706236 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 1400
19:21:30.706236 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 732


  


  I hope this is a configuration problem, that I can fix this in my ffplay/ffmpeg line, and it's not a bug in ffmpeg. Thanks for any tips.

  


• How to prevent overwriting fifo while playing local file with ffmpeg ?

  21 juillet 2013, par frankish

  I use av_read_frame() to read and push frames to fifo. In another thread, I read from this fifo. So, reading and writing does not interfere that much.

  However, I found out that when playing from a local file, audio suddenly skips to somewhere else.. I tried to open recorded file with VLC player and it worked. So the file(recording) is not broken.

  I tried (just to see if it works) the following and it works but it has other problems.

  ...
  
  if(localFile){
  
      while(av_fifo_size(fifoPlayback)>2000000){
  
          pthread_mutex_unlock(&fifoPlaybackMutex);
  
          usleep(1000000);
  
          pthread_mutex_lock(&fifoPlaybackMutex);
  
          // Waiting before writing to fifo, it's not empty enough");
  
      }
  
  }
  
  
  
  av_fifo_generic_write(fifoPlayback,...)

  Since the above block at least does not skip, I think my problem is really overwriting on the fifo. But how should I correctly manage this situation ?