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• Shell Script for passing multiple audio sources as arguments to a script that uses ffmpeg for recording video/audio ? [duplicate]

  29 novembre 2023, par Raul Chiarella

  I have the following Shell/Bash Script, that records video with audio in my Linux :

  


  #!/bin/bash

# Command to record audio from one microphone and video from one webcam
ffmpeg_cmd="ffmpeg -f video4linux2 -s 320x240 -i /dev/video0 -f alsa -ac 1 -i hw:1,0 -acodec libmp3lame -ab 96k camera.mp4"

# Execute the command
echo "Executing: $ffmpeg_cmd"
eval $ffmpeg_cmd


  


  This one works... But now :

  


  What I want is to be able to execute this Bash/Shell script passing multiple audio sources parameters and changing if video is going to be recorded or not (Video 1 Enabled, Video 0 Disabled), for example, using ./recordVideo.sh -a "audio1,audio2" -v 1...

  


  I tried recording a video accepting multiple audio sources arguments and passing it to FFMPEG using the following script :

  


  #!/bin/bash

# Default variables
audio_sources=""
video_flag=0

# Function to display help
usage() {
    echo "Usage: $0 -a 'Microphone1,Microphone2' -v 1"
    echo "  -a : List of audio sources, separated by commas"
    echo "  -v : Video flag (1 to record video, 0 not to record)"
    exit 1
}

# Parse arguments
while getopts "a,v" opt; do
  case $opt in
    a) audio_sources=$OPTARG ;;
    v) video_flag=$OPTARG ;;
    *) usage ;;
  esac
done

# Check if audio arguments were provided
if [ -z "$audio_sources" ]; then
    echo "Error: Audio sources not specified."
    usage
fi

# Initial ffmpeg command configuration
cmd="ffmpeg"

# Counter for source mapping
source_count=0

# Audio settings
IFS=',' read -ra ADDR <<< "$audio_sources"
for source in $ADDR; do
    cmd+=" -f alsa -ac 1 -i $source"
    cmd+=" -map $source_count"
    ((source_count++))
done

# Video settings
if [ "$video_flag" -eq 1 ]; then
    cmd+=" -f video4linux2 -s 320x240 -i /dev/video0 -map $source_count"
    ((source_count++))
fi

# Audio codec configuration
cmd+=" -acodec libmp3lame -ab 96k"

# Output file name
cmd+=" recordedVideo.mp4"

# Execute command
echo "Executing: $cmd"
eval $cmd


  


  But it is throwing the following error when I execute ./recordVideo.sh -a 'Microphone1,Microphone2' -v 1 :

  


  


  Error : Audio sources not specified.
Usage : ./recordVideo.sh -a 'Microphone1,Microphone2' -v 1
-a : List of audio sources, separated by commas
-v : Video flag (1 to record video, 0 not to record)

  


  


  Can someone help me ? :(
What am I doing wrong ? Is it the Shell Syntax or FFMPEG arguments that are wrong ?

  


• How to make audio sound batter ? (C + FFMpeg audio generation example)

  31 janvier 2014, par Spender

  So I found this grate C FFMpeg official example which I simplified :

  #include 
  
  #include 
  
  #include 
  
  
  
  #ifdef HAVE_AV_CONFIG_H
  
  #undef HAVE_AV_CONFIG_H
  
  #endif
  
  
  
  #include "libavcodec/avcodec.h"
  
  #include "libavutil/mathematics.h"
  
  
  
  #define INBUF_SIZE 4096
  
  #define AUDIO_INBUF_SIZE 20480
  
  #define AUDIO_REFILL_THRESH 4096
  
  
  
  /*
  
   * Audio encoding example
  
   */
  
  static void audio_encode_example(const char *filename)
  
  {
  
      AVCodec *codec;
  
      AVCodecContext *c= NULL;
  
      int frame_size, i, j, out_size, outbuf_size;
  
      FILE *f;
  
      short *samples;
  
      float t, tincr;
  
      uint8_t *outbuf;
  
  
  
      printf("Audio encoding\n");
  
  
  
      /* find the MP2 encoder */
  
      codec = avcodec_find_encoder(CODEC_ID_MP2);
  
      if (!codec) {
  
          fprintf(stderr, "codec not found\n");
  
          exit(1);
  
      }
  
  
  
      c= avcodec_alloc_context();
  
  
  
      /* put sample parameters */
  
      c->bit_rate = 64000;
  
      c->sample_rate = 44100;
  
      c->channels = 2;
  
  
  
      /* open it */
  
      if (avcodec_open(c, codec) < 0) {
  
          fprintf(stderr, "could not open codec\n");
  
          exit(1);
  
      }
  
  
  
      /* the codec gives us the frame size, in samples */
  
      frame_size = c->frame_size;
  
      samples = malloc(frame_size * 2 * c->channels);
  
      outbuf_size = 10000;
  
      outbuf = malloc(outbuf_size);
  
  
  
      f = fopen(filename, "wb");
  
      if (!f) {
  
          fprintf(stderr, "could not open %s\n", filename);
  
          exit(1);
  
      }
  
  
  
      /* encode a single tone sound */
  
      t = 0;
  
      tincr = 2 * M_PI * 440.0 / c->sample_rate;
  
      for(i=0;i<200;i++) {
  
          for(j=0;j* encode the samples */
  
          out_size = avcodec_encode_audio(c, outbuf, outbuf_size, samples);
  
          fwrite(outbuf, 1, out_size, f);
  
      }
  
      fclose(f);
  
      free(outbuf);
  
      free(samples);
  
  
  
      avcodec_close(c);
  
      av_free(c);
  
  }
  
  
  
  int main(int argc, char **argv)
  
  {
  
  
  
      /* must be called before using avcodec lib */
  
      avcodec_init();
  
  
  
      /* register all the codecs */
  
      avcodec_register_all();
  
  
  
      audio_encode_example("test.mp2");
  
  
  
      return 0;
  
  }

  How should it sound like ? May be I don't get something but it sounds awful =( how to make audio generation sound batter/ more interesting/ melodical in a wary shourt way (no special functions just how to change this code to make it sound batter) ?

• ffmpeg replacing audio track with track from another video but different length (sync audio)

  27 août 2019, par Matt

  Background

  I have digitized some old Canon Video8 tapes. I used a SONY Digital8 camera (which is backwards compatible with Video8) to output DV (using it’s inbuilt ADC). The conversion process worked well for the video but the audio came through jumpy/distorted in places. This left me with a problem, was it the Camera or the tape ? So I bought another Samsung Video8 camera (just analog) and using the SONY’s passthrough feature output from the Samsung (Composite & mono audio) into the SONY which output the DV. Much to my delight it worked ! The audio was clear.

  Result

  DV01.avi - Good Video / Crap Audio
  
  DV02.avi - Crap Video / Good Audio

  Ok so obviously what I would like to do is take the video track from DV01 and the audio track from DV02 and join/mux ? them WITHOUT re-encoding.

  Problem 1 : They have different start times so just copying over the audio track will result it not being in sync.

  After some googling I found you can use ffmpeg to take care of this :

  Firstly here is the Video info using : ffmpeg -i DV01.avi

  Input #0, avi, from 'DV01.avi':
  
    Duration: 02:53:06.68, start: 0.000000, bitrate: 28878 kb/s
  
      Stream #0:0: Video: dvvideo, yuv420p, 720x576 [SAR 16:15 DAR 4:3], 25000 kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc
  
      Stream #0:1: Audio: pcm_s16le, 32000 Hz, stereo, s16, 1024 kb/s
  
      Stream #0:2: Audio: pcm_s16le, 32000 Hz, stereo, s16, 1024 kb/s

  Muxing :

  ffmpeg -itsoffset 4 -i DV01.avi -i DV02.avi -map 0:v -map 1:a -c copy output.avi

  In the example above I am delaying the start of the video 4 seconds (I think ?) This is just an example I haven’t actually tried it as the file sizes are 40Gb !

  So my QUESTION is :

  Given the background above what would be the best way to join/mux the two streams together (without re-encoding) with the audio being in sync. Given that syncing the audio to the video may need millisecond tweaking I don’t believe trial and error is a good idea (I don’t want to tweak it by 10ms then rinse and repeat for a 40Gb file) ?

  I just had a thought, could I say create a 10 second clip (from the start) of each video and use them to find the reference/sync start point then use that when muxing the 40Gb versions ?

  Anyway, you get the idea. Looking for ways to solve this problem. Thanks !