Recherche avancée

Sur d’autres sites (5350)

• How to Kill ffmpeg process in node.js

  8 février 2017, par Sanjay

  I am using node.js code that convert Axis Ipcamera live stream into mp4 using FFMPEG

   var childProcess=require('child_process');
  
   var childArguments = [];
  
  var child=[];
  
  var cmd='ffmpeg -i rtsp://172.24.22.117:554/axis-media/media.amp -vcodec libx264 -pix_fmt yuv420p -profile:v baseline -preset slower -crf 18 -vf "scale=trunc(in_w/2)*2:trunc(in_h/2)*2"'+' '+__dirname+'/uploads/ouput.mp4';
  
  
  
    child=childProcess.exec(       
  
          cmd,
  
          childArguments,
  
          {            
  
              env: process.env,
  
              silent:true
  
          },  function (err, stdout, stderr) {
  
              if (err) {
  
                  throw err;
  
              }
  
              console.log(stdout);
  
  
  
          });     
  
  
  
          //    here generate events for listen child process (works properly)
  
      // Listen for incoming(stdout) data
  
      child.stdout.on('data', function (data) {
  
          console.log("Got data from child: " + data);
  
      });
  
  
  
      // Listen for any errors:
  
      child.stderr.on('data', function (data) {
  
          console.log('There was an error: ' + data);
  
      });
  
  
  
      // Listen for exit event
  
      child.on('exit', function(code) {
  
          console.log('Child process exited with exit code ' + code);
  
          child.stdout.pause();
  
          child.kill();
  
      });

  my above code works perfectly. It gives the output as I want, but I am not able to kill(stop) the ffmpeg command. I am using the code below for stopping the process, but in background it still continues.

  child.kill("SIGTERM"); 

  I also used following commands : child.kill(’SIGUSR1’) ; child.kill("SIGHUP") ; child.kill("SIGINT") ;child.kill(’SIGUSR2’) ; for killing this process but it not works.

  Currently I forcefully kill the node application to stop ffmpeg command and generate mp4 file. I do not want this.
  But I want commands that stop ffmpeg process and generate mp4 file, without killing the node application.

• How to set sample rate when read frames from ALSA [FFMPEG C/C++]

  26 juin 2023, par Dion

  I'm trying to get audio data from a microphone using ALSA. By default stream has a sampling rate of 44100 Hz, but I need to get 8000 Hz.

  


  Trying to use AVDictionary options for avformat_open_input doesn't change anything.

  


  The code is maximum simplified :

  


  AVFormatContext *format_context = nullptr;
AVInputFormat   *input_format   = nullptr;

avdevice_register_all();

input_format = av_find_input_format("alsa");

AVDictionary* options = NULL;
av_dict_set(&options, "sample_rate", "8000", 0);

int res = avformat_open_input(&format_context, "hw:0", input_format, &options);

if(res < 0)
{
    exit(1);
}

res = avformat_find_stream_info(format_context, 0);

if(res < 0)
{
    exit(1);
}

av_dump_format(format_context, 0, "alsa", 0);


  


  


  Input #0, alsa, from 'alsa' : Duration : N/A, start :
1685994324.766645, bitrate : 1411 kb/s
Stream #0:0 : Audio : pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s

  


  


  Is there probably some way to tell ALSA to output a lower sampling rate ?

  


  Thanks !

  


• mp3 decoding using ffmpeg API (Header missing)

  24 juin 2019, par WIN

  i have been trying to decode an MP3 file to pcm, using ffmpeg API, but i keep getting an error

  [mp3 @ 0x8553020]Header missing

  this is the code i use :

  #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
  
  
  
  
  
  static void audio_decode_example(const char *outfilename, const char *filename)
  
  {
  
      AVCodec *codec;
  
      AVCodecContext *c= NULL;
  
      int out_size, len;
  
      FILE *f, *outfile;
  
      uint8_t *outbuf;
  
      uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
  
      AVPacket avpkt;
  
  
  
      av_init_packet(&avpkt);
  
  
  
      printf("Audio decoding\n");
  
  
  
      /* find the mpeg audio decoder */
  
      codec = avcodec_find_decoder(CODEC_ID_MP3ON4);
  
      if (!codec) {
  
      fprintf(stderr, "codec not found\n");
  
      exit(1);
  
      }
  
  
  
      c= avcodec_alloc_context();
  
  
  
      /* open it */
  
      if (avcodec_open(c, codec) < 0) {
  
      fprintf(stderr, "could not open codec\n");
  
      exit(1);
  
      }
  
  
  
      outbuf = malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);
  
  
  
      f = fopen(filename, "rb");
  
      if (!f) {
  
      fprintf(stderr, "could not open %s\n", filename);
  
      exit(1);
  
      }
  
      outfile = fopen(outfilename, "wb");
  
      if (!outfile) {
  
      av_free(c);
  
      exit(1);
  
      }
  
  
  
      /* decode until eof */
  
      avpkt.data = inbuf;
  
      avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
  
  
  
      while (avpkt.size > 0) {
  
      out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
  
      len = avcodec_decode_audio3(c, (short *)outbuf, &out_size, &avpkt);
  
      if (len < 0) {
  
          fprintf(stderr, "Error while decoding\n");
  
          exit(1);
  
      }
  
      if (out_size > 0) {
  
          /* if a frame has been decoded, output it */
  
          fwrite(outbuf, 1, out_size, outfile);
  
      }
  
      avpkt.size -= len;
  
      avpkt.data += len;
  
      if (avpkt.size < AUDIO_REFILL_THRESH) {
  
          /* Refill the input buffer, to avoid trying to decode
  
           * incomplete frames. Instead of this, one could also use
  
           * a parser, or use a proper container format through
  
           * libavformat. */
  
          memmove(inbuf, avpkt.data, avpkt.size);
  
          avpkt.data = inbuf;
  
          len = fread(avpkt.data + avpkt.size, 1,
  
                      AUDIO_INBUF_SIZE - avpkt.size, f);
  
          if (len > 0)
  
              avpkt.size += len;
  
      }
  
      }
  
  
  
      fclose(outfile);
  
      fclose(f);
  
      free(outbuf);
  
  
  
      avcodec_close(c);
  
      av_free(c);
  
  }
  
  
  
  int main(int argc, char **argv)
  
  {
  
      const char *filename;
  
  
  
      /* must be called before using avcodec lib */
  
      avcodec_init();
  
  
  
      /* register all the codecs */
  
      avcodec_register_all();
  
  
  
      audio_decode_example("test.wav", argv[1]);
  
  
  
      return 0;
  
  }

  when i use the same code to directly play the sound, like this :

  if (out_size > 0) {
  
      /* if a frame has been decoded, output it *
  
      play_sound(outbuf, out_size);
  
  }

  i have no problem at all with some files, other mp3 files just gives an error without even starting ... is there any ideas ?

  PS : this code is from libavcodec/api-example.c , modified as needed