Sur d’autres sites (7081)

• The Future of the VP8 Bitstream

  18 juin 2010, par noreply@blogger.com (John Luther) — vp8

  Recently we’ve seen software products such as VLC, FFmpeg, Logitech Vid, Flumotion and Tixeo adopting and using WebM and VP8 (the video codec in WebM) in exciting new ways.

  In addition to software developers, many hardware vendors have committed to shipping VP8-accelerated products based on our current bitstream in 2011 . Devices that use hardware acceleration for video are a very small percentage of overall web traffic today, but they are a rapidly growing segment of the market and our project must be mindful of these vendors’ needs. Given the longer lead times for changes in chipsets, hardware companies implementing the codec today need to be confident that it will be stable and supported as VP8 content proliferates.

  Like every codec, WebM is not immune to change ; the difference in our project is that the improvements are publicly visible, and compatibility and implementation issues can be worked through in an open forum.

  So, to maintain codec stability while also allowing for quality and performance improvements in VP8, we have added an experimental branch to the VP8 source tree. The WebM community can use this unstable branch to propose changes to VP8 that will produce the best video codec possible, but without the constraints of a frozen bitstream. At some point in the future, when the experimental branch proves significantly better than the stable branch, we will create a new version of the codec.

  Teams dedicated to improving WebM are actively investigating and evaluating new techniques, and are committed to do so for the long term. We encourage the WebM community to keep contributing as well. To learn more about the experimental branch and get involved, see our repository layout page.

  Jim Bankoski is Codec Engineering Manager at Google.

  

• FFMPEG - infinite file video stream to RTMP server

  5 août 2022, par samoht9277

  I'm trying to create an infinite video stream that I can push to a local RTMP server.

  


  I believe it's worth mentioning that I'm not very good with ffmpeg, I started using it some weeks ago.

  


  Here is a similat question, but it doesn't fully answer what I need.

  


  What I tried

  


  At my first attempt, I made a bash script that spawned a new ffmpeg instance for each video I was trying to stream. I noticed that that the few seconds that takes to spawn the new instance interrupts the stream.

  


  After I noticed this problem, I decided to google some kind of queue system, and I found that ffmpeg has a concat function, that reads a file with a custom syntax and queues it.

  


  So I made a script that lists the files in the directory where I store all my videos, modifies the output with sed, shuffles it (so the episodes are random), and then it redirects the output to a playlist.txt

  


  find /mnt/episode/ -print | sed '/**.mp4/!d' | sed "s/^/file '/; s/$/'/" | shuf > playlist.txt


  


  playlist.txt

  


  example :

  


  ...
file '/mnt/episode/9/1.mp4'
file '/mnt/episode/8/4.mp4'
file '/mnt/episode/5/11.mp4'
file '/mnt/episode/7/15.mp4'
...


  


  This creates 2 problems :

  


  

  1. For some reason, while using the playlist/queue method, when it finishes a video... it takes like 30 seconds to switch to the next video.

  


  


  ffmpeg -f concat -safe 0 -i playlist.txt -f v4l2 /dev/video0


  


  (simplified command, using /dev/video0 device for testing purposes)

  


  ffmpeg does output something after the first video, but i'm not quite sure what it means.

  


  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x55c8abc1b900] st: 1 edit list: 1 Missing key frame while searching for timestamp: 0
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x55c8abc1b900] st: 1 edit list 1 Cannot find an index entry before timestamp: 0.


  


  

  2. Once the playlist finishes, the stream is over. I could run the script that generates the playlist again, but that would mean that the stream cuts, which I don't want.

  


  


  ---

  


  What I want

  


  I need a way to pick a random file, stream it normally via RTMP and once it finished, automatically pick another one without interrupting the streaming, and keep doing this until I decide to turn it off.

  


  This is the directory format :

  


  episodes
│
├── 1
│   ├── 1.mp4
│   ├── 2.mp4
│   ├── 3.mp4
│   ...
├── 2
│   ├── 1.mp4
│   ├── 2.mp4
│   ├── 3.mp4
│   ...
├── 3
│   ├── 1.mp4
│   ├── 2.mp4
│   ├── 3.mp4
│   ...
├── 4
│   ├── 1.mp4
│   ├── 2.mp4
│   ├── 3.mp4
│   ...
├── 5
│   ├── 1.mp4
│   ├── 2.mp4
│   ├── 3.mp4
│   ...
├── 6
│   ├── 1.mp4
│   ├── 2.mp4
│   ├── 3.mp4
│   ...
├── 7
│   ├── 1.mp4
│   ├── 2.mp4
│   ├── 3.mp4
│   ...
├── 8
│   ├── 1.mp4
│   ├── 2.mp4
│   ├── 3.mp4
│   ...
└── 9
    ├── 1.mp4
    ├── 2.mp4
    ├── 3.mp4
    ...


  


  Gladly appreciate some help here. I tried to make the problem as clear as possible.

  


  Thanks <3

  &#xA;

• FFMPEG background audio cutting out where clips are joined ?

  11 avril 2020, par legit_hacker_2000

  I am concatenating a few audio files together, then adding a background track to them. Concatenating the audio files together works however when I add the background music it seems that at every frame where an old clip ends and a new one starts, the music cuts out for a frame or two.

  &#xA;&#xA;

  Concat

  &#xA;&#xA;

  ffmpeg-y -fconcat -safe 0  -i ffmpeg_files.txt output/no_bg_out.mp4


  &#xA;&#xA;

  Background Track

  &#xA;&#xA;

  ffmpeg -y -i output/no_bg_out.mp4 -i media/bg1.mp3 -filter_complex[1:a]volume=0.1apad[A];[0:a][A]amerge[out] -c:v copy -map 0:v -map [out] -shortest out.mp4


  &#xA;&#xA;

  Console Output

  &#xA;&#xA;

  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'output/no_bg_out.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf58.42.100
  Duration: 00:02:11.04, start: 0.000000, bitrate: 221 kb/s
    Stream #0:0(eng): Video: h264 (High 4:4:4 Predictive) (avc1 / 0x31637661), yuv444p, 1920x1080, 145 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 24000 Hz, mono, fltp, 71 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
Input #1, mp3, from 'media/bg1.mp3':
  Metadata:
    encoder         : Lavf58.42.100
  Duration: 00:34:32.69, start: 0.023021, bitrate: 128 kb/s
    Stream #1:0: Audio: mp3, 48000 Hz, stereo, fltp, 128 kb/s
    Metadata:
      encoder         : Lavc58.77
Stream mapping:
  Stream #0:1 (aac) -> amerge:in0
  Stream #1:0 (mp3float) -> volume
  Stream #0:0 -> #0:0 (copy)
  amerge -> Stream #0:1 (aac)
Press [q] to stop, [?] for help
[Parsed_amerge_2 @ 000001b99fe46d00] No channel layout for input 1
Output #0, mp4, to 'out.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf58.42.100
    Stream #0:0(eng): Video: h264 (High 4:4:4 Predictive) (avc1 / 0x31637661), yuv444p, 1920x1080, q=2-31, 145 kb/s, 25 fps, 25 tbr, 12800 tbn, 12800 tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1: Audio: aac (LC) (mp4a / 0x6134706D), 24000 Hz, 3.0, fltp, 197 kb/s (default)
    Metadata:
      encoder         : Lavc58.77.101 aac
frame= 3276 fps=706 q=-1.0 Lsize=    5620kB time=00:02:10.92 bitrate= 351.6kbits/s speed=28.2x
video:2328kB audio:3211kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.447797%


  &#xA;