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• FFMPEG output truncated when adding watermark

  17 juin 2020, par user1452030

  I'm using the following command to generate a watermarked 320Kbps MP3 preview of a wav file :

  



  ffmpeg -i /path/input.wav -y -filter_complex "amovie=/path/wm_padded.wav:loop=0,asetpts=N/SR/TB,adelay=3000|3000[beep];[0][beep]amix=duration=shortest,volume=2"  -b:a 320k  /path/preview.mp3


  



  (wm_padded.wav is the watermark file padded to 10 seconds and I'm using a Mac. The command was structured based on this post)

  



  While this works as expected at times, it produces a short, garbled preview some other times. Any help in debugging this issue would be greatly appreciated. I've verified the input file and it seems to be fine and here's the FFMPEG command output :

  



      $ ffmpeg -i /path/input.wav -y -filter_complex "amovie=/path/wm_padded.wav:loop=0,asetpts=N/SR/TB,adelay=3000|3000[beep];[0][beep]amix=duration=shortest,volume=2"  -b:a 320k -vsync 2 /path/test-Preview.mp3
ffmpeg version 4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.8)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.2.1_2 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags='-I/Library/Java/JavaVirtualMachines/adoptopenjdk-13.jdk/Contents/Home/include -I/Library/Java/JavaVirtualMachines/adoptopenjdk-13.jdk/Contents/Home/include/darwin -fno-stack-check' --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-libsoxr --enable-videotoolbox --disable-libjack --disable-indev=jack
  libavutil      56. 31.100 / 56. 31.100
  libavcodec     58. 54.100 / 58. 54.100
  libavformat    58. 29.100 / 58. 29.100
  libavdevice    58.  8.100 / 58.  8.100
  libavfilter     7. 57.100 /  7. 57.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
[wav @ 0x7ff2ba801a00] Discarding ID3 tags because more suitable tags were found.
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from '/path/input.wav':
  Metadata:
    comment         :  motion graphics, motion, textures
    time_reference  : 0
    coding_history  : 
  Duration: 00:00:01.75, bitrate: 1520 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
    Stream #0:1: Video: mjpeg (Baseline), yuvj420p(pc, bt470bg/unknown/unknown), 288x288 [SAR 72:72 DAR 1:1], 90k tbr, 90k tbn, 90k tbc (attached pic)
    Metadata:
      comment         : Cover (front)
[Parsed_amovie_0 @ 0x7ff2b9d0c040] Channel layout is not set in output stream 0, guessed channel layout is 'stereo'
Stream mapping:
  Stream #0:0 (pcm_s16le) -> amix:input0 (graph 0)
  volume (graph 0) -> Stream #0:0 (libmp3lame)
  Stream #0:1 -> #0:1 (mjpeg (native) -> png (native))
Press [q] to stop, [?] for help
[swscaler @ 0x108d00000] deprecated pixel format used, make sure you did set range correctly
[Parsed_amovie_0 @ 0x7ff2b9e09840] Channel layout is not set in output stream 0, guessed channel layout is 'stereo'
Output #0, mp3, to '/path/preview.mp3':
  Metadata:
    comment         :  motion graphics, motion, textures
    time_reference  : 0
    coding_history  : 
    TSSE            : Lavf58.29.100
    Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, fltp, 320 kb/s
    Metadata:
      encoder         : Lavc58.54.100 libmp3lame
    Stream #0:1: Video: png, rgb24(progressive), 288x288 [SAR 1:1 DAR 1:1], q=2-31, 200 kb/s, 90k fps, 90k tbn, 90k tbc (attached pic)
    Metadata:
      comment         : Cover (front)
      encoder         : Lavc58.54.100 png
frame=    1 fps=0.0 q=-0.0 Lsize=     206kB time=00:00:00.10 bitrate=15994.8kbits/s speed=3.62x    
video:200kB audio:5kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.598223%


  



  Thanks in advance !

  


• python for loop result in ffmpeg command [duplicate]

  6 août 2020, par madiha bout

  hi im trying to insert for loop results in a ffmpeg command i tried some code but all i get is first text with no sound and it glitches and freezes i tried many modification but no success i appreciate if you have any ideas that would be great, the imported text from api should display only the duration of audio file here is my code :

  


  
import requests
import json
import os
import requests
import http.client
import io
import subprocess


response = requests.get('http://api.quran.com:3000/api/v3/chapters/2/verses?text_type=image&language=ar&recitation=10')

json = json.loads(response.content)

data= json['verses']
#print(data)

for i in data:
    input_text = i['text_madani'] 
    input_duration = i['audio']['duration'] 
    input_mp3 = i['audio']['url']
    input_img = i['image']['url']
 #   print(input_duration)
    command = f'''ffmpeg -re -stream_loop -1 \
                    -i img2.jpeg \
                    -i {input_mp3} \
                    -vf 'pad=ceil(iw/2)*2:ceil(ih/2)*2,drawtext=enable='between(t, n, {input_duration})':text={input_text}:fontsize=90:x=50:y=50:fontcolor=black@0.8' \
                    -c:v libx264 -preset veryfast -b:v 3000k \
                    -maxrate 3000k -bufsize 6000k -pix_fmt yuv420p -g 50 \
                    -c:a aac -b:a 160k \
                    -ac 2 -ar 44100 \
                    -y -f flv rtmp://localhost/hls/test''' 
    print(command)
    os.system(command)


  


• How do I use FFmpeg to fetch an audio from a 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.