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• Ffmpeg stereo buffer size issue

  8 janvier 2020, par grunt

  When using ffmpeg library (c++), when audio is stereo (channels = 2) there is a problem in data_size : in the case of code below, fwrite line will crash on access violation reading. The channels parameter doubles the data_size when stereo. There is no problem when c->channels is replaced with 1.

  The channels parameter is set to 2 in the avcodec_decode_audio4 function (and I indeed use a stereo file [joint stereo]. when I use a mono file channels is set to 1).

  If I dump input_packet’s to a file it plays ok.

  What could be the reason for this ? My code is very much similar to code examples on the net.

  while (input_packet.size > 0)
  
  {
  
    len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &input_packet);
  
    if (got_frame)
  
    {
  
      int data_size = av_samples_get_buffer_size(NULL, c->channels,
  
                                                     decoded_frame->nb_samples,
  
                                                     c->sample_fmt, 1);
  
      fwrite(decoded_frame->data[0], 1, data_size, outfile); // access violation reading loction
  
      // outArr.insert(outArr.end(), decoded_frame->data[0], decoded_frame->data[0] + data_size); // This will crash in same manner. outArr is vector<byte>.
  
    }
  
  }
  
  </byte>

• FFMPEG, resize and pad a video by and odd number of pixels ?

  18 février 2018, par Jules

  I’m trying to resize and pad a video from 1917 x 1080 to 1920 x 1080.

  I’ve tried various syntax, which works but doesn’t change the size.

  ffmpeg -i input.mp4 -filter:v scale=1920:1080,pad=1920:1080 -c:a copy output.mp4

  However, I have go to this point by resize, joint audio and rotate. The initial size is 640 x 1136, I believe this is the source of the problem.

  ffmpeg -i input.mp4 -filter:v scale=900:1200 -c:a copy output.mp4
  
  
  
  ffmpeg \
  
  -i input.m4a \
  
  -i resize.mp4 -acodec copy -vcodec copy -shortest \
  
  output.mp4
  
  
  
  ffmpeg -i input.mp4" -vf "transpose=2" output/mp4

  So I’m wondering if I should do something different earlier

• Trying to convert an mp3 file to a Numpy Array, and ffmpeg just hangs

  29 mai 2021, par Rich

  I'm working on a music classification methodology with Scikit-learn, and the first step in that process is converting a music file to a numpy array.

  



  After unsuccessfully trying to call ffmpeg from a python script, I decided to simply pipe the file in directly :

  



  FFMPEG_BIN = "ffmpeg"
cwd = (os.getcwd())
dcwd = (cwd + "/temp")
if not os.path.exists(dcwd): os.makedirs(dcwd)

folder_path = sys.argv[1]
f = open("test.txt","a")

for f in glob.glob(os.path.join(folder_path, "*.mp3")):
    ff = f.replace("./", "/")
    print("Name: " + ff)
    aa = (cwd + ff)

    command = [ FFMPEG_BIN,
        '-i',  aa,
        '-f', 's16le',
        '-acodec', 'pcm_s16le',
        '-ar', '22000', # ouput will have 44100 Hz
        '-ac', '1', # stereo (set to '1' for mono)
        '-']

    pipe = sp.Popen(command, stdout=sp.PIPE, bufsize=10**8)
    raw_audio = pipe.proc.stdout.read(88200*4)
    audio_array = numpy.fromstring(raw_audio, dtype="int16")
    print (str(audio_array))
    f.write(audio_array + "\n")


  



  The problem is, when I run the file, it starts ffmpeg and then does nothing :

  



  [mp3 @ 0x1446540] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from '/home/don/Code/Projects/MC/Music/Spaz.mp3':
  Metadata:
    title           : Spaz
    album           : Seeing souns
    artist          : N*E*R*D
    genre           : Hip-Hop
    encoder         : Audiograbber 1.83.01, LAME dll 3.96, 320 Kbit/s, Joint Stereo, Normal quality
    track           : 5/12
    date            : 2008
  Duration: 00:03:50.58, start: 0.000000, bitrate: 320 kb/s
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
Output #0, s16le, to 'pipe:':
  Metadata:
    title           : Spaz
    album           : Seeing souns
    artist          : N*E*R*D
    genre           : Hip-Hop
    date            : 2008
    track           : 5/12
    encoder         : Lavf56.4.101
    Stream #0:0: Audio: pcm_s16le, 22000 Hz, mono, s16, 352 kb/s
    Metadata:
      encoder         : Lavc56.1.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (mp3 (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help


  



  It just sits there, hanging, for far longer than the song is. What am I doing wrong here ?,