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• Gstreamer, x264enc "Redistribute latency..." error

  20 août 2021, par Jason

  I'm trying to setup a video pipeline with very limited bandwidth. I was able to do it with two raspberry pis using the below lines. The first is for the camera pi and the second is to watch stream :

  


  gst-launch-1.0 rpicamsrc preview=false !  'video/x-h264, width=800, height=600, framerate=30/1' ! h264parse ! rtph264pay config-interval=1 pt=96 ! gdppay ! udpsink host=YOUR_PC_IP port=5000
gst-launch-1.0 udpsrc port=5000 ! gdpdepay ! rtph264depay ! h264parse ! avdec_h264 ! autovideosink sync=false


  


  It works but I go over my bandwidth limit if there is movement. I'm not sure if there is a way to limit bandwidth by setting a parameter here :

  


  'video/x-h264, width=800, height=600, framerate=30/1'


  


  From what I can find online, I have to use something like x264enc. I've followed tutorials but I can't get x264enc to work. it always outputs "Redistribute latency..." on both machines when run and it stays there.

  


  I've tried using x264enc like follows :

  


  gst-launch-1.0 rpicamsrc preview=false !  'video/x-raw, width=800, height=600, framerate=30/1' ! x264enc ! h264parse ! rtph264pay config-interval=1 pt=96 ! gdppay ! udpsink host=YOUR_PC_IP port=5000
gst-launch-1.0 rpicamsrc preview=false !  'video/x-raw, width=800, height=600, framerate=30/1' ! x264enc tune=zerolatency ! h264parse ! rtph264pay config-interval=1 pt=96 ! gdppay ! udpsink host=YOUR_PC_IP port=5000
gst-launch-1.0 rpicamsrc preview=false ! x264enc ! h264parse ! rtph264pay config-interval=1 pt=96 ! gdppay ! udpsink host=YOUR_PC_IP port=5000
gst-launch-1.0 rpicamsrc preview=false ! x264enc tune=zerolatency ! h264parse ! rtph264pay config-interval=1 pt=96 ! gdppay ! udpsink host=YOUR_PC_IP port=5000


  


  Based on tutorials, I would think some of those should work. Other threads say that tune=zerolatency fixes my problem. At least the ones with same output of "Redistribute latency..." I don't know what I'm doing wrong.

  


  Any help would be appreciated. Thanks !

  


• How to debug "Error in pull function & Input buffer exhausted" ?

  15 septembre 2021, par user16909319

  Introduction :

  


  I'm working on a Discord Music Bot for personal use only. Basically, when I play a 2min song with the bot, it plays the song for 1min 30secs, then skips it and plays the next one in the queue. The error is shown below :

  


  


  Error in Pull Function
  
IO error : Error number -10054 occurred
  
[mov,mp4,m4a,3gp,3g2,mj2 @ 0000018f86a6f4c0] Packet corrupt (stream = 0, dts = 11154432).
  
Input buffer exhausted before END element found
  
Invalid Data found when processing Input
  
[mov,mp4,m4a,3gp,3g2,mj2 @ 0000018f86a6f4c0] stream 0, offset 0x3e805c : partial file

  


  


  The code block which I think is causing the problem :

  


  Search song method

  


      async def search_song(self, amount, song, get_url=False):
        info = await self.bot.loop.run_in_executor(None, lambda: youtube_dl.YoutubeDL(ytdl_format_options).extract_info(
            f"ytsearch{amount}:{song}", download=False, ie_key="YoutubeSearch"))
        if len(info["entries"]) == 0: return None

        return [entry["webpage_url"] for entry in info["entries"]] if get_url else info


  


  Play_song method

  


      async def play_song(self, ctx, song):
        url = pafy.new(song).getbestaudio().url
        ctx.voice_client.play(discord.PCMVolumeTransformer(discord.FFmpegPCMAudio(url, executable="C:/ffmpeg/bin/ffmpeg.exe")),
                              after=lambda error: self.bot.loop.create_task(self.check_queue(ctx)))
        ctx.voice_client.source.volume = 0.5


  


  Formatting Options I provided :

  


  ytdl_format_options = {
    'format': 'bestaudio/best',
    'outtmpl': '%(extractor)s-%(id)s-%(title)s.%(ext)s',
    'restrictfilenames': True,
    'noplaylist': True,
    'nocheckcertificate': True,
    'ignoreerrors': True,
    'logtostderr': False,
    'quiet': True,
    'no_warnings': True,
    'default_search': 'auto',
    'source_address': '0.0.0.0'
}


  


  Solutions that I've tried :

  


  

  • Running it on both Replit and locally.

  


  • Redownloading FFmpeg

  


  • Ensuring FFmpeg, pafy, and youtube_dl are all up to date.

  


  


  Things to Note :

  


  

  • Playing a 2mins song, it stops after 1min 30 seconds and displays the error above. (75% of the song)

  


  • Playing a 1hr song, it still continues after 10 minutes.

  


  


  I do not have much experience in this yet so I'm not entirely sure where in my code is actually causing this issue or other ways which I can use to test and fix the issue.

  


• Static ffmpeg library windows

  29 novembre 2022, par peter

  Im trying to use the ffmpeg libraries (libavutil, libswscale, ...) in a bigger project by statically linking them, im not interested in the binaries. Now the issue comes down to getting them statically built. I tried searching for pre-compiled builds but without success.

  


  Following the docs, i compiled them on wsl2 for win64 with

  


  ./configure --arch=x86_64 --target-os=mingw32 --cross-prefix=i686-w64-mingw32- --disable-shared --enable-static
make


  


  After that i copied the libavformat/libavformat.a as libavformat.lib (which should be fine since they are cross-compiled ar archives ?) into my project and linked them with CMake.

  


  When compiling smth like

  


  AVFormatContext* pFormatContext = avformat_alloc_context();


  


  i get a linker error saying

  


  lld-link : error : undefined symbol: avformat_alloc_context


  


  When looking at the libs with dumpbin -linkermember libavformat.lib, i can see

  


  >dumpbin -linkermember libavformat.lib
...
  135CD92 _avformat_alloc_context
...


  


  notice the underscore.

  


  Can i not link the cross-compiled *.a files directly into a windows executable ?

  


  Honestly at this point any ideas are welcome.

  


  Thanks