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• (ffmpeg) How to sync dshow inputs, dropping frames, and -rtbufsize [closed]

  29 juillet 2021, par Zach Fleeman

  I wrote a quick batch script to capture anything from my Elgato HD60 Pro capture card, and while it works in some ways, I don't really understand how certain parameters are affecting my capture.

  


  Whenever I run this command without the -rtbufsize 2048M -thread_queue_size 5096 params, I drop a ton of frames. I only added those params with those values because I found them on another stackoverflow thread. I wouldn't mind actually knowing what these do, and how I can fine-tune them for my script.

  


  ffmpeg.exe -y -rtbufsize 2048M -thread_queue_size 5096 -fflags +igndts ^
-f dshow -i video="Game Capture HD60 Pro":audio="Game Capture HD60 Pro Audio" ^
-filter:v "crop=1410:1080:255:0, scale=706x540" ^
-c:v libx264 -preset veryfast -b:v 1500k -pix_fmt yuv420p ^
-c:a aac ^
-f tee -map 0:v -map 0:a "%mydate%_%mytime%_capture.mp4|[f=flv]rtmp://xxx.xxx.xxx.xxx/live"


  


  In Open Broadcaster Software, my Elgato is a near-instant video feed, but this captures/streams things at a 3-ish second delay, which is okay until I work on this second command. I'm using gdigrab to capture the window from LiveSplit for my speedrunning, but I can't get the video streams to be synced up. I tried adding and modifying another -rtbufsize before the gdigrab input, but again, I'm not sure if this is what I need to do to delay the LiveSplit grab. It seems to always be 2 to 3 seconds ahead of my capture card. How can I get these inputs to be synced and react at the same time ? i.e., I start the timer in LiveSplit at the same time that I hit a button on my super nintendo.

  


  ffmpeg.exe -y -rtbufsize 750M -thread_queue_size 5096 ^
-f dshow  -i video="Game Capture HD60 Pro":audio="Game Capture HD60 Pro Audio" ^
-rtbufsize 2000M -thread_queue_size 5096 ^
-f gdigrab -r 60 -i title=LiveSplit ^
-filter_complex "[0:v][0:v]overlay=255:0 [game];[game][1:v]overlay=0:40 [v]" ^
-c:v libx264 -preset veryfast -b:v 1500k -pix_fmt yuv420p ^
-c:a aac ^
-f tee -map "[v]" -map 0:a "%mydate%_%mytime%_capture.mp4|[f=flv]rtmp://192.168.1.7/live"


  


  tl ;dr
Where should I put -rtbufsize ? What value should it be ? And how about -thread_queue_size ? Are these things that I have to specify once or multiple times for each input ? How can I get my different input sources to sync up ?

  


  p.s., I'm cropping and overlaying my Elgato inputs because my capture card does 1920x1080, but my video is most likely a 4:3-ish SNES/NES game.

  


• Chiptune Database and API

  14 septembre 2012, par Multimedia Mike — General

  So I set out to create a website that allows people to easily listen to video game music directly through their web browser. I succeeded in that goal. However, I must admit that the project has limited appeal since the web player is delivered via Chrome’s Native Client technology, somewhat limiting its audience. I’m not certain if anyone really expects NaCl to take off in any serious way, but I still have a few other projects in mind.

  I recently realized that, as a side effect of this project, I accidentally created something of significant value to fans of old video games and associated music– a searchable database of chiptune music and metadata. To my knowledge, no one else has endeavored to create such a thing. I figured that I might as well make the database easily accessible with an API and see where it leads.

  To that end, I created 2 API entry points. First, there is the search API located at http://gamemusic.multimedia.cx/api/search/. This can be exercised by ending the URL with a URL-encoded search string, e.g. : http://gamemusic.multimedia.cx/api/search/super+mario. This returns JSON data containing an array of results in decreasing order of relevance. Each result has a game title, database ID, media URL, system type, and an SHA-1 hash. This is the same API that the site’s own search page uses.

  The database ID can be plugged into http://gamemusic.multimedia.cx/api/metadata/ to retrieve the song’s metadata in JSON format. E.g., the ID for Super Mario Bros. 3 on the NES is 161 : http://gamemusic.multimedia.cx/api/metadata/161.

  I recently read an article about sins against true RESTful API principles which led me to believe I’m almost certainly doing this web API stuff wrong. I don’t think it’s a huge deal, though, since I don’t think anyone actually listens to chiptunes any more. But if there are offline chiptune music players that are still in service and actively maintained, perhaps the authors would like to implement this API. It would require some type of HTTP networking library, a JSON parser, the embedded XZ decoder, and some new code to parse through my .gamemusic and .psfarchive formats.

  This database could be a significant value-add to chiptune playback software, and could help people experience classic game music much more easily.

• ffmpeg scale, how to crop correctly

  23 mars 2015, par sathia

  I’m using this command to encode videos

  $transcode = FFMPEG_BINARY.' -loglevel panic -y -i "'.$files['original'].'" -vf scale='.VIDEO_SIZE_X.':'.VIDEO_SIZE_Y.' -vcodec libx264 -profile main -preset slow -r 25 -b '.VIDEO_BITRATE.' -maxrate '.VIDEO_BITRATE.' -bufsize 1000k -threads '.VIDEO_THREADS.' -acodec aac -ar 44100 -f mp4 -strict -2 '.$files['mp4'];
  
  
  
  where: VIDEO_SIZE_X = 640 and VIDEO_SIZE_Y = 480, VIDEO_BITRATE = 900k

  it all seems to work fine, but the problem I’m having is that the video is not resized to the desired size which is 640x480

  

  ^ This is the output from vlc

  It looks like there’s some reference to the desired size, but the video is not scaled/cropped,
  what’s the correct way of scaling videos in order to have the desired size ? I don’t mind having black stripes above and below or bands at the sides.

  so, here’s a bit of debugging as requested :

  /usr/bin/ffmpeg -loglevel panic -y -i "in.wmv" -vf scale=640:480 -vcodec libx264 -profile main -preset slow -r 25 -b 900k -maxrate 900k -bufsize 1000k -threads 8 -acodec aac -ar 44100 -f mp4 -strict -2 out.mp4

  original video :

  [wmv3 @ 0x13245c0] Extra data: 8 bits left, value: 20
  
  Input #0, asf, from 'in.wmv':
  
    Metadata:
  
      WMFSDKVersion   : 12.0.9600.16384
  
      WMFSDKNeeded    : 0.0.0.0000
  
      IsVBR           : 1
  
      VBR Peak        : 313
  
      Buffer Average  : 397
  
    Duration: 00:06:09.13, start: 0.000000, bitrate: 2111 kb/s
  
      Stream #0:0: Audio: wmav2 (a[1][0][0] / 0x0161), 48000 Hz, 2 channels, s16, 96 kb/s
  
      Stream #0:1: Video: wmv3 (Main) (WMV3 / 0x33564D57), yuv420p, 860x484, 2000 kb/s, SAR 1:1 DAR 215:121, 29.97 tbr, 1k tbn, 1k tbc
  
  [wmv3 @ 0x13245c0] Extra data: 8 bits left, value: 20

  while the result is :

  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'out.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : Lavf54.29.104
  
    Duration: 00:06:08.76, start: 0.000000, bitrate: 975 kb/s
  
      Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 640x480 [SAR 645:484 DAR 215:121], 842 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc
  
      Metadata:
  
        handler_name    : VideoHandler
  
      Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 125 kb/s
  
      Metadata:
  
        handler_name    : SoundHandler

  cheers