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• Write, with ffmpeg, every 30 seconds wav file from ip stream to file.temp and change the name to timestamp.wav after

  18 juillet 2018, par Eliya

  I’m using ffmpeg to record IP stream and write every 30 seconds to wav file.

  Here is my bash script code :

  #!/bin/bash
  
  function start_ffmpeg_stream ()
  
  {
  
      address=$1 #This is IP stream address
  
      ffmpeg_option=$2 # "?overrun_nonfatal=1&fifo_size=250000"
  
      folder_name=$3 #folderName
  
      channel_number=$4 #i
  
      #local pid      
  
      ffmpeg  -loglevel 8 -thread_queue_size 1500 -i "$address$ffmpeg_option" -c copy\
  
          -vn -f segment -segment_time 30 -ar 8000 -acodec pcm_s32le -ac 1 -strftime 1 /"$folder_name"/"X$channel_number""_""%s.wav"&     
  
      pid=$!
  
      echo "$Start ffmpeg, pid is - $pid" 
  
      __="$pid"
  
  }
  
  ffmpegOptions="?overrun_nonfatal=1&fifo_size=250000"
  
  folderName="/wav_files"
  
  start_ffmpeg_stream "udp://224.10.0.1"  "$ffmpegOptions" "$folderName" "1"

  Now the wav file name is like "X000001_unix_time_stamp.wav".

  I want to write the file name in writeing time something like "X000001_unix_time_stamp.temp"

  And when the 30 seconds done and the FFmpeg finish to write this 30 seconds, I want that FFmpeg changes the name to "X000001_unix_time_stamp.wav"

  And keep writing next 30 seconds.

  The only change that i want is, when FFmpeg writing it write in temp name and after FFmpeg finish to write it change the name.

  It’s seem to case when i donloaded a file so until the donload not finish the file has a temp name and when it done the name change to const name.

• Getting ffserver && ffmpeg to work of Debian with Logitech C270 [closed]

  14 mars 2013, par Joseph Baldwin Roberts

  I'm having real trouble getting my webcam to work and I wonder if anyone can help. Im running Raspian on a raspberry pi.

  When I run lsusb I get :

  pi@raspberrycar ~ $ lsusb
  
  Bus 001 Device 002: ID 0424:9512 Standard Microsystems Corp.
  
  Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub
  
  Bus 001 Device 003: ID 0424:ec00 Standard Microsystems Corp.
  
  Bus 001 Device 004: ID 05e3:0608 Genesys Logic, Inc. USB-2.0 4-Port HUB
  
  Bus 001 Device 005: ID 0bda:8176 Realtek Semiconductor Corp. RTL8188CUS 802.11n WLAN Adapter
  
  Bus 001 Device 006: ID 046d:0825 Logitech, Inc. Webcam C270

  When I run v412-ctl —all I get :

  pi@raspberrycar ~ $ v4l2-ctl --all
  
  Driver Info (not using libv4l2):
  
          Driver name   : uvcvideo
  
          Card type     : UVC Camera (046d:0825)
  
          Bus info      : usb-bcm2708_usb-1.2.4
  
          Driver version: 3.6.11
  
          Capabilities  : 0x04000001
  
                  Video Capture
  
                  Streaming
  
  Format Video Capture:
  
          Width/Height  : 640/480
  
          Pixel Format  : 'YUYV'
  
          Field         : None
  
          Bytes per Line: 1280
  
          Size Image    : 614400
  
          Colorspace    : SRGB
  
  Crop Capability Video Capture:
  
          Bounds      : Left 0, Top 0, Width 640, Height 480
  
          Default     : Left 0, Top 0, Width 640, Height 480
  
          Pixel Aspect: 1/1
  
  Video input : 0 (Camera 1: ok)
  
  Streaming Parameters Video Capture:
  
          Capabilities     : timeperframe
  
          Frames per second: 5.000 (5/1)
  
          Read buffers     : 0

  When I run v412-ctl —list-formats I get :

  pi@raspberrycar ~ $ v4l2-ctl --list-formats
  
  ioctl: VIDIOC_ENUM_FMT
  
          Index       : 0
  
          Type        : Video Capture
  
          Pixel Format: 'YUYV'
  
          Name        : YUV 4:2:2 (YUYV)
  
  
  
          Index       : 1
  
          Type        : Video Capture
  
          Pixel Format: 'MJPG' (compressed)
  
          Name        : MJPEG

  My ffserver settings file is /etc/ffserver.conf

  Port 80
  
  BindAddress 0.0.0.0
  
  MaxClients 10
  
  MaxBandwidth 50000
  
  NoDaemon
  
  
  
  <feed>
  
  file /tmp/webcam.ffm
  
  FileMaxSize 10M
  
  </feed>
  
  
  
  <stream>
  
  Feed webcam.ffm
  
  Format mpeg
  
  VideoSize 640x480
  
  VideoFrameRate 10
  
  VideoBitRate 2000
  
  VideoQMin 1
  
  VideoQMax 10
  
  </stream>
  
  
  
  <stream>
  
  Format status
  
  
  
  ACL allow 192.168.2.0 192.168.2.255
  
  </stream>

  My ffmpeg run string is /usr/sbin/webcam.sh

  ffserver -f /etc/ffserver.conf & ffmpeg -v verbose -r 5 -s 640x480 -f video4linux2 -i /dev/video0 http://localhost/webcam.ffm

  When I run this is the output :

  pi@raspberrycar ~ $ sudo /usr/sbin/webcam.sh
  
  ffmpeg version 1.0.5ffserver version 1.0.5 Copyright (c) 2000-2012 the FFmpeg developers
  
    built on Mar 14 2013 18:37:40 with gcc 4.6 (Debian 4.6.3-14+rpi1)
  
   Copyright (c) 2000-2012 the FFmpeg developers
  
    built on Mar 14 2013 18:37:40 with gcc 4.6 (Debian 4.6.3-14+rpi1)
  
    configuration:
  
    libavutil      51. 73.101 / 51. 73.101
  
    configuration:
  
    libavutil      51. 73.101 / 51. 73.101
  
    libavcodec     54. 59.100 / 54. 59.100
  
    libavformat    54. 29.104 / 54. 29.104
  
    libavdevice    54.  2.101 / 54.  2.101
  
    libavfilter     3. 17.100 /  3. 17.100
  
    libswscale      2.  1.101 /  2.  1.101
  
    libavcodec     54. 59.100 / 54. 59.100
  
    libavformat    54. 29.104 / 54. 29.104
  
    libavdevice    54.  2.101 / 54.  2.101
  
    libavfilter     3. 17.100 /  3. 17.100
  
    libswscale      2.  1.101 /  2.  1.101
  
    libswresample   0. 15.100 /  0. 15.100
  
    libswresample   0. 15.100 /  0. 15.100
  
  [video4linux2,v4l2 @ 0xfa5620] [3]Capabilities: 4000001

  I can see the stream on the status page and my webcam light is on but it never loads. Does anyone know what I am doing wrong ?

  Thanks in advance

  Joe

• How to Use FFmpeg to Fetch an Audio From Local Network and Decode it to PCM ?

  26 mai 2020, par Yousef Alaqra

  Currently, I have a node js server which is connected to a specific IP address on the local network (the source of the audio), to receive the audio using VBAN protocol. VBAN protocol, basically uses UDP to send audio over the local network.

  



  Node js implementation :

  



  http.listen(3000, () => {
  console.log("Server running on port 3000");
});

let PORT = 6980;
let HOST = "192.168.1.244";

io.on("connection", (socket) => {
  console.log("a user connected");
  socket.on("disconnect", () => {
    console.log("user disconnected");
  });
});

io.on("connection", () => {

  let dgram = require("dgram");
  let server = dgram.createSocket("udp4");

  server.on("listening", () => {
    let address = server.address();
    console.log("server host", address.address);
    console.log("server port", address.port);
  });

  server.on("message", function (message, remote) {
    let audioData = vban.ProcessPacket(message);
    io.emit("audio", audioData); // // console.log(`Received packet: ${remote.address}:${remote.port}`)
  });
  server.bind({
    address: "192.168.1.230",
    port: PORT,
    exclusive: false,
  });
});


  



  once the server receives a package from the local network, it processes the package, then, using socket.io it emits the processed data to the client.

  



  An example of the processed audio data that's being emitted from the socket, and received in the angular :

  



       audio {&#xA;      format: {&#xA;        channels: 2,&#xA;        sampleRate: 44100,&#xA;        interleaved: true,&#xA;        float: false,&#xA;        signed: true,&#xA;        bitDepth: 16,&#xA;        byteOrder: &#x27;LE&#x27;&#xA;      },&#xA;      sampleRate: 44100,&#xA;      buffer: <buffer 2e="2e" 00="00" ce="ce" ff="ff" 3d="3d" bd="bd" 44="44" b6="b6" 48="48" c3="c3" 32="32" d3="d3" 31="31" d4="d4" 30="30" dd="dd" 38="38" 34="34" e5="e5" 1d="1d" c6="c6" 25="25" 974="974" more="more" bytes="bytes">,&#xA;      channels: 2,&#xA;}&#xA;</buffer>

  



  In the client-side (Angular), after receiving a package using socket.io.clinet, AudioConetext is used to decode the audio and play it :

  



     playAudio(audioData) {
    let audioCtx = new AudioContext();
    let count = 0;
    let offset = 0;
    let msInput = 1000;
    let msToBuffer = Math.max(50, msInput);
    let bufferX = 0;
    let audioBuffer;
    let prevFormat = {};
    let source;

    if (!audioBuffer || Object.keys(audioData.format).some((key) => prevFormat[key] !== audioData.format[key])) {
      prevFormat = audioData.format;
      bufferX = Math.ceil(((msToBuffer / 1000) * audioData.sampleRate) / audioData.samples);
      if (bufferX < 3) {
        bufferX = 3;
      }
      audioBuffer = audioCtx.createBuffer(audioData.channels, audioData.samples * bufferX, audioData.sampleRate);
      if (source) {
        source.disconnect();
      }
      source = audioCtx.createBufferSource();
      console.log("source", source);
      source.connect(audioCtx.destination);
      source.loop = true;
      source.buffer = audioBuffer;
      source.start();
    }
  }


  



  Regardless that audio isn't playing in the client-side, and there is something wrong, this isn't the correct implementation.

  



  Brad, mentioned in the comments below, that I can implement this correctly and less complexity using FFmpeg child-process. And I'm very interested to know how to fetch the audio locally using FFmpeg.