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• ffmpeg is adding an extra new line into a box

  4 août 2017, par newbie123

  I’m overlaying weather data over my webcam stream. I put a background box filter, but the padding between the text and the box is uneven. There is "a new line" of extended box bellow the text.

  

  Why is that ? There are no empty lines in a text file I’m providing to ffmpeg. Here is my code for the ffmpeg :

  ffmpeg \
  
  -f lavfi -i anullsrc \
  
  -rtsp_transport tcp \
  
  -i "$SOURCE" \
  
  -vcodec libx264 -pix_fmt yuv420p -preset ultrafast -g 20 -b:v 1000k \
  
  -vf "drawtext="fontfile=${FONT}":textfile=${textfile}:x=5:y=55:reload=1: \
  
  fontcolor=white:fontsize=${FONTSIZE}:box=1:boxborderw=5:boxcolor=black@0.5" \
  
  -threads $THREADS -bufsize 512k \
  
  -f flv "$YOUTUBE_URL/$KEY"

  EDIT :

  I found out what the problem was. I was using printf "%s\n" "$variable1" "$variable2" to create a text file. Printf %s\n automatically prints each variable into a new line. The solution was to print the last variable without the new line. Code example :

  #!/bin/sh
  
  LC_CTYPE=en_US.utf8
  
  
  
  # Get APRS weather data from aprs.fi
  
  
  
  wxstation="S55MA-10"
  
  name="Juršče, Pivka"
  
  
  
  
  
  # Basic weather data
  
  temp="$(wget -q https://aprs.fi/weather/a/${wxstation} -O - | grep Temperature | egrep '[-+]?([0-9]*\.[0-9]+|[0-9]+)' -o)"
  
  humidity="$(wget -q https://aprs.fi/weather/a/${wxstation} -O - | grep Humidity | egrep '[-+]?([0-9]*\.[0-9]+|[0-9]+)' -o)"
  
  wind="$(wget -q https://aprs.fi/weather/a/${wxstation} -O - | grep Wind | egrep '[-+]?([0-9]*\.[0-9]+|[0-9]+)' -o | sed -n -e 2p)"
  
  rain="$(wget -q https://aprs.fi/weather/a/${wxstation} -O - | grep Rain | egrep '[-+]?([0-9]*\.[0-9]+|[0-9]+)' -o | sed -n -e 1p)"
  
  
  
  # Telemetry
  
  radioactivity="$(wget -q https://aprs.fi/telemetry/a/${wxstation} -O - | grep Radioactivity | egrep '[-+]?([0-9]*\.[0-9]+|[0-9]+)' -o | sed -n -e 5p)"
  
  
  
  printf "%s\n" "$name" "Temperature: ${temp}\°C" "Humidity: ${humidity}\%" "Wind: ${wind} m/s" "Rain: ${rain} mm/h"
  
  printf "Radioactivity: ${radioactivity} uSv/h"

• Screeching white sound coming while playing audio as a raw stream

  27 avril 2020, par Sri Nithya Sharabheshwarananda

  I. Background

  



  

  1. I am trying to make an application which helps to match subtitles to the audio waveform very accurately at the waveform level, at the word level or even at the character level.

  


  2. The audio is expected to be Sanskrit chants (Yoga, rituals etc.) which are extremely long compound words [ example - aṅganyā-sokta-mātaro-bījam is traditionally one word broken only to assist reading ]

  


  3. The input transcripts / subtitles might be roughly in sync at the sentence/verse level but surely would not be in sync at the word level.

  


  4. The application should be able to figure out points of silence in the audio waveform, so that it can guess the start and end points of each word (or even letter/consonant/vowel in a word), such that the audio-chanting and visual-subtitle at the word level (or even at letter/consonant/vowel level) perfectly match, and the corresponding UI just highlights or animates the exact word (or even letter) in the subtitle line which is being chanted at that moment, and also show that word (or even the letter/consonant/vowel) in bigger font. This app's purpose is to assist learning Sanskrit chanting.

  


  5. It is not expected to be a 100% automated process, nor 100% manual but a mix where the application should assist the human as much as possible.

  


  



  II. Following is the first code I wrote for this purpose, wherein

  



  

  1. First I open a mp3 (or any audio format) file,

  


  2. Seek to some arbitrary point in the timeline of the audio file // as of now playing from zero offset

  


  3. Get the audio data in raw format for 2 purposes - (1) playing it and (2) drawing the waveform.

  


  4. Playing the raw audio data using standard java audio libraries

  


  



  III. The problem I am facing is, between every cycle there is screeching sound.

  



  

  • Probably I need to close the line between cycles ? Sounds simple, I can try.

  


  • But I am also wondering if this overall approach itself is correct ? Any tip, guide, suggestion, link would be really helpful.

  


  • Also I just hard coded the sample-rate etc ( 44100Hz etc. ), are these good to set as default presets or it should depend on the input format ?

  


  



  IV. Here is the code

  



  import com.github.kokorin.jaffree.StreamType;
import com.github.kokorin.jaffree.ffmpeg.FFmpeg;
import com.github.kokorin.jaffree.ffmpeg.FFmpegProgress;
import com.github.kokorin.jaffree.ffmpeg.FFmpegResult;
import com.github.kokorin.jaffree.ffmpeg.NullOutput;
import com.github.kokorin.jaffree.ffmpeg.PipeOutput;
import com.github.kokorin.jaffree.ffmpeg.ProgressListener;
import com.github.kokorin.jaffree.ffprobe.Stream;
import com.github.kokorin.jaffree.ffmpeg.UrlInput;
import com.github.kokorin.jaffree.ffprobe.FFprobe;
import com.github.kokorin.jaffree.ffprobe.FFprobeResult;
import java.io.IOException;
import java.io.OutputStream;
import java.nio.file.Path;
import java.nio.file.Paths;
import java.util.concurrent.TimeUnit;
import java.util.concurrent.atomic.AtomicLong;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.SourceDataLine;


public class FFMpegToRaw {
    Path BIN = Paths.get("f:\\utilities\\ffmpeg-20190413-0ad0533-win64-static\\bin");
    String VIDEO_MP4 = "f:\\org\\TEMPLE\\DeviMahatmyamRecitationAudio\\03_01_Devi Kavacham.mp3";
    FFprobe ffprobe;
    FFmpeg ffmpeg;

    public void basicCheck() throws Exception {
        if (BIN != null) {
            ffprobe = FFprobe.atPath(BIN);
        } else {
            ffprobe = FFprobe.atPath();
        }
        FFprobeResult result = ffprobe
                .setShowStreams(true)
                .setInput(VIDEO_MP4)
                .execute();

        for (Stream stream : result.getStreams()) {
            System.out.println("Stream " + stream.getIndex()
                    + " type " + stream.getCodecType()
                    + " duration " + stream.getDuration(TimeUnit.SECONDS));
        }    
        if (BIN != null) {
            ffmpeg = FFmpeg.atPath(BIN);
        } else {
            ffmpeg = FFmpeg.atPath();
        }

        //Sometimes ffprobe can't show exact duration, use ffmpeg trancoding to NULL output to get it
        final AtomicLong durationMillis = new AtomicLong();
        FFmpegResult fFmpegResult = ffmpeg
                .addInput(
                        UrlInput.fromUrl(VIDEO_MP4)
                )
                .addOutput(new NullOutput())
                .setProgressListener(new ProgressListener() {
                    @Override
                    public void onProgress(FFmpegProgress progress) {
                        durationMillis.set(progress.getTimeMillis());
                    }
                })
                .execute();
        System.out.println("audio size - "+fFmpegResult.getAudioSize());
        System.out.println("Exact duration: " + durationMillis.get() + " milliseconds");
    }

    public void toRawAndPlay() throws Exception {
        ProgressListener listener = new ProgressListener() {
            @Override
            public void onProgress(FFmpegProgress progress) {
                System.out.println(progress.getFrame());
            }
        };

        // code derived from : https://stackoverflow.com/questions/32873596/play-raw-pcm-audio-received-in-udp-packets

        int sampleRate = 44100;//24000;//Hz
        int sampleSize = 16;//Bits
        int channels   = 1;
        boolean signed = true;
        boolean bigEnd = false;
        String format  = "s16be"; //"f32le"

        //https://trac.ffmpeg.org/wiki/audio types
        final AudioFormat af = new AudioFormat(sampleRate, sampleSize, channels, signed, bigEnd);
        final DataLine.Info info = new DataLine.Info(SourceDataLine.class, af);
        final SourceDataLine line = (SourceDataLine) AudioSystem.getLine(info);

        line.open(af, 4096); // format , buffer size
        line.start();

        OutputStream destination = new OutputStream() {
            @Override public void write(int b) throws IOException {
                throw new UnsupportedOperationException("Nobody uses thi.");
            }
            @Override public void write(byte[] b, int off, int len) throws IOException {
                String o = new String(b);
                boolean showString = false;
                System.out.println("New output ("+ len
                        + ", off="+off + ") -> "+(showString?o:"")); 
                // output wave form repeatedly

                if(len%2!=0) {
                    len -= 1;
                    System.out.println("");
                }
                line.write(b, off, len);
                System.out.println("done round");
            }
        };

        // src : http://blog.wudilabs.org/entry/c3d357ed/?lang=en-US
        FFmpegResult result = FFmpeg.atPath(BIN).
            addInput(UrlInput.fromPath(Paths.get(VIDEO_MP4))).
            addOutput(PipeOutput.pumpTo(destination).
                disableStream(StreamType.VIDEO). //.addArgument("-vn")
                setFrameRate(sampleRate).            //.addArguments("-ar", sampleRate)
                addArguments("-ac", "1").
                setFormat(format)              //.addArguments("-f", format)
            ).
            setProgressListener(listener).
            execute();

        // shut down audio
        line.drain();
        line.stop();
        line.close();

        System.out.println("result = "+result.toString());
    }

    public static void main(String[] args) throws Exception {
        FFMpegToRaw raw = new FFMpegToRaw();
        raw.basicCheck();
        raw.toRawAndPlay();
    }
}



  



  Thank You

  


• how to override_ffserver parameters with a ffmpeg command line ?

  1er février 2017, par Bepeho

  I’m designing a simple camera switching system with ffmpeg and ffserver.
  I’m using ffmpeg to redirect one camera stream to a single ffserver.

  I’ve setup the ffserver to generate webm output :

  <feed>
  
      File /tmp/PREVIEW.ffm
  
      FileMaxSize 1M
  
  </feed>
  
  <stream>           
  
         Feed PREVIEW.ffm             
  
         Format webm
  
          AudioBitRate 64
  
          AudioSampleRate 24000
  
          AudioChannels 1
  
          VideoCodec libvpx
  
          VideoSize 640x380
  
          VideoFrameRate 25
  
          AVOptionVideo flags +global_header
  
          AVOptionAudio flags +global_header
  
          AVOptionVideo cpu-used 0
  
          AVOptionVideo qmin 10
  
          AVOptionVideo qmax 42
  
          AVOptionVideo quality good
  
          PreRoll 0
  
          StartSendOnKey
  
          VideoBitRate 1M
  
  </stream>

  It works well when i feed it with a ffmpeg commmand line such as :

  ffmpeg -re -rtbufsize 1500M  -thread_queue_size 512  -rtsp_transport tcp -i rtsp://admin:admin@192.168.0.10:554  -f lavfi -i anullsrc  http://localhost:8090/PREVIEW.ffm

  Now I want to insert dynamic text within the generated video stream, so I’ve added to the previous command line a drawtext directive :

  ffmpeg -re -rtbufsize 1500M -re -rtbufsize 1500M -thread_queue_size 512 -override_ffserver -rtsp_transport tcp -i rtsp://admin:admin@192.168.0.10:554 -f lavfi -i anullsrc -vf drawtext='fontfile=/usr/share/fonts/truetype/dejavu/DejaVuSans.ttf:textfile=mytextfile.txt:reload=1:fontcolor=white:fontsize=48:box=1:boxcolor=black@0.5:boxborderw=5:x=(w-text_w):y=(h-text_h)' http://localhost:8090/PREVIEW.ffm

  But of course, it’s not working !

  I suppose that it’s because ffserver commands ffmpeg stream generation directives.

  So i’ve added a -override_ffserver parameter and audio+video generation parameters :

  ffmpeg -re -rtbufsize 1500M  -thread_queue_size 512  -rtsp_transport tcp -i rtsp://admin:admin@192.168.0.10:554  -f lavfi -i anullsrc -vf drawtext='fontfile=/usr/share/fonts/truetype/dejavu/DejaVuSans.ttf:textfile=mytextfile.txt:reload=1:fontcolor=white:fontsize=48:box=1:boxcolor=black@0.5:boxborderw=5:x=(w-text_w):y=(h-text_h)' -vcodec libvpx -flags:v +global_header -b:v 1M -acodec libopus -ac 1 -ar 24000 -flags:a +global_header -s 320x190 -r 25 -override_ffserver http://localhost:8090/PREVIEW.ffm

  Still, the stream looks incorrect as ffplay logs lines such as :

  vp8 @ 0000000004e88220] Unknown profile 4
  
  vp8 @ 0000000004e88220] Header size larger than data 

  Can anyone give me an hint on how to solve this problem ?

  Many Thanks.

  EDIT :
  I’ve found a workaround by piping the video+audio+text mix result to a second ffmpeg process.

  Hence, There’s no need to use the override_ffserver parameter anymore :

  ffmpeg -y -re -rtbufsize 1500M  -thread_queue_size 512 -rtsp_transport tcp -i rtsp://admin:admin@192.168.0.10:554  -f lavfi -i anullsrc  -vf drawtext='fontfile=/usr/share/fonts/truetype/dejavu/DejaVuSans.ttf:textfile=mytextfile.txt:reload=1:fontcolor=white:fontsize=48:box=1:boxcolor=black@0.5:boxborderw=5:x=(w-text_w):y=(h-text_h)' -f nut pipe:1 | ffmpeg  -i pipe:0 http://localhost:8090/PREVIEW.ffm

  But, if anyone has a better, leaner and simpler solution, I’m for it.