Recherche avancée

Sur d’autres sites (10894)

• ffmpeg merge separate .webm audio and video files using pts_time possible ?

  4 août 2021, par Zach

  I have a number of audio and video files with different start times and end times. They're all generated from piped input streams (node.js) with .webm format, so the audio files have gaps where there is no audio in the piped stream.

  


  I'm trying to :

  


  

  1. Merge the audio files together with wall-clock correct start/end times

  


  2. Merge the video files using hstack.

  


  3. Combine the merged audio and merged video into 1 final video with all video/audio.

  


  


  Right now I'm still stuck on step 1 - Merge the audio

  


  My command that generates separate audio files is :

  


          '-protocol_whitelist',
        'pipe,udp,rtp',
        '-fflags',
        '+genpts',
        '-f',
        'sdp',
        '-use_wallclock_as_timestamps',
        'true',
        '-i',
        'pipe:0'
        '-copyts',
        '-map',
        '0:a:0',
        '-strict'
        '-2',
        '-c:a',
        'copy'


  


  I'd love to combine them somehow using the timestamps of each packet fill the empty space with silence. Right now, I'm putting them together with offsets using the time that I initiate the ffmpeg process from node.js, but these times are incorrect, as it takes a moment for ffmpeg to start up.

  


  Any assistance or a push in the right direction for time sensitive merging of audio/video .webm files with ffmpeg would be outstanding.

  


  Thanks !

  


  PS, here's what I'm currently doing and running into a problems with :

  


     '-i',
  './recordings/audio_1.webm',
  '-i',
  './recordings/audio_2.webm',
  '-filter_complex',
  '[1]adelay=6384|6384[b];[0][b]amix=2',
  './recordings/merged_audio.webm'


  


  The delays are inaccurate (because they're based on an estimate of when the first packet starts) and doesn't account for gaps in the audio files :(

  


• Low latency video player on android

  20 mai 2021, par Louis Blenner

  I'd like to be able to stream the video from my webcam to an Android app with a latency below 500ms, on my local network.

  


  To capture and send the video over the network, I use ffmpeg.

  


  ffmpeg -f v4l2 -i /dev/video0 -preset ultrafast -tune zerolatency -vcodec libx264 -an -vf format=yuv420p -f mpegts  udp://192.168.1.155:5000


  


  This command takes the webcam as an input, convert it and send it to a device using the mpegts protocol.
  
This is not a requirement, if another technique could work, I could change the way I send the video.

  


  I am able to read the video on another PC from the local network with a latency below 500 ms, using commands like

  


  gst-launch-1.0 -v udpsrc port=5000 ! video/mpegts ! tsdemux ! h264parse ! avdec_h264 ! fpsdisplaysink sync=false


  


  or

  


  mpv udp://0.0.0.0:5000 --no-cache --untimed --no-demuxer-thread --video-sync=audio --vd-lavc-threads=1 


  


  So it is possible to have this range of latency.
  
I'd like to have the same thing on Android.

  


  Here are my tries to do that.

  


  Exoplayer

  


  After looking at the different players available on Android studio, it seems like Exoplayer is the go-to choice.
  
I tried different options indicated in the live-streaming documentation, but I always end up with a stream taking seconds to start and with a latency of seconds.
  
I tried to add a Button to seek to the default position of the windows, but it results in a loading of several seconds.

  


  DefaultExtractorsFactory extractorsFactory =
                new DefaultExtractorsFactory()
                        .setTsExtractorFlags(DefaultTsPayloadReaderFactory.FLAG_IGNORE_AAC_STREAM);

        player = new SimpleExoPlayer.Builder(this)
                .setMediaSourceFactory(
                        new DefaultMediaSourceFactory(this, extractorsFactory))
                .setLoadControl(new DefaultLoadControl.Builder()
                        .setBufferDurationsMs(DefaultLoadControl.DEFAULT_MIN_BUFFER_MS, DefaultLoadControl.DEFAULT_MAX_BUFFER_MS, 200, 200)
                        .build())
                .build();
        MyPlayerView playerView = findViewById(R.id.player_view);
        // Bind the player to the view.
        playerView.setPlayer(player);
        // Build the media item.
        MediaItem mediaItem = new MediaItem.Builder()
                .setUri(Uri.parse("udp://0.0.0.0:5000"))
                .setLiveMaxOffsetMs(500)
                .setLiveTargetOffsetMs(0)
                .setLiveMinOffsetMs(0)
                .build();
        // Set the media item to be played.
        player.setMediaItem(mediaItem);
        // Prepare the player.
        player.setPlayWhenReady(true);
        player.prepare();
        //player.seekToDefaultPosition();


  


  This issue is about the same issue and the conclusion was that Exoplayer was not fit for this use case.

  


  


  I'll be honest, ultra low-latency like this isn't ExoPlayer's main use-case

  


  


  Vlc

  


  Another try was to use the Vlc library.
  
But I was unable to have the same low latency stream as with the two previous players with Vlc.
  
I tried changing the preferences of Vlc to stream as fast as possible as described here

  


  Input/Codecs -> x264 preset: ultrafast - zerolatency
Input/Codecs -> Access Module: UDP input
Input/Codecs -> Clock Jitter: 500
Audio: disable audio


  


  I also tried reducing the different buffers.
  
However, I still have a latency of more than 1 seconds with that.

  


  Gstreamer

  


  Another try was to create a react-native project to use the different players available here.
  
One player that seemed promising was react-native-gstreamer because it uses gstreamer which is able to stream with low latency (gst-launch command).
  
But the library is now outdated.

  


  Question

  


  There were other tries, but none were successful.
  
Is there a problem with one of my approaches ?
  
And if not, Is there a player on Android (that I missed) which is able to achieve low latency stream like gstream or mpv on linux ?

  


• Is it posible to add an HTML page as an overlay in a RTMP stream using Nginx and FFMPEG ?

  4 août 2021, par tanowalla

  I have already installed Nginx on a EC2 instance (Ubuntu 18.04) for livestreaming to Youtube using RTMP. I want to setup a HTML page with a world clock and transparent background.

  


  I installed Nginx using this tutorial.

  


  https://www.scaleway.com/en/docs/setup-rtmp-streaming-server/

  


  Would it be possible to load the HTML page using NGINX and FFMPEG so i can overlay it over the RTMP livestream incoming from a camera and push it to Youtube ?

  


  I hope someone gets the idea, i apologize for my english, it is not my native language and have really no idea if this is even possible.

  


  Thank you very much in advance.