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• Compiled FFmpeg not accepting -c:v and -c:a

  13 mars 2020, par King Horse

  I compiled FFmpeg with libsrt, with the online compile guide. https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu & how to compile ffmpeg with enabling libsrt

  It seems to compile correctly.

  ffmpeg version N-96575-g843c24a Copyright (c) 2000-2020 the FFmpeg developers
  
  built with gcc 7 (Ubuntu 7.4.0-1ubuntu1~18.04.1)
  
  configuration: --prefix=/home/ubuntu/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/ubuntu/ffmpeg_build/include --extra-ldflags=-L/home/ubuntu/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/ubuntu/bin --enable-gpl --enable-libaom --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libsrt --enable-nonfree
  
  libavutil      56. 38.100 / 56. 38.100
  
  libavcodec     58. 67.100 / 58. 67.100
  
  libavformat    58. 37.100 / 58. 37.100
  
  libavdevice    58.  9.103 / 58.  9.103
  
  libavfilter     7. 72.100 /  7. 72.100
  
  libswscale      5.  6.100 /  5.  6.100
  
  libswresample   3.  6.100 /  3.  6.100
  
  libpostproc    55.  6.100 / 55.  6.100

  But when running this command to convert a incoming SRT stream to HLS, it doesn’t know the -c:a command. When switching the order, it runs that it doesn’t know about the -c:v command.

  ffmpeg -re -i srt://0.0.0.0:25000?pkt_size=1316&mode=listener -c:a copy -c:v copy -strict -f hls -hls_time 4 -hls_playlist_type event stream.m3u8

  ~$ ffmpeg -re -i srt://0.0.0.0:25000?pkt_size=1316&mode=listener -c:a copy -c:v copy -strict -f hls -hls_time 4 -hls_playlist_type event stream.m3u8
  
  [2] 9930
  
  ffmpeg version N-96575-g843c24a Copyright (c) 2000-2020 the FFmpeg developers
  
    built with gcc 7 (Ubuntu 7.4.0-1ubuntu1~18.04.1)
  
    configuration: --prefix=/home/ubuntu/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/ubuntu/ffmpeg_build/include --extra-ldflags=-L/home/ubuntu/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/ubuntu/bin --enable-gpl --enable-libaom --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libsrt --enable-nonfree
  
    libavutil      56. 38.100 / 56. 38.100
  
    libavcodec     58. 67.100 / 58. 67.100
  
    libavformat    58. 37.100 / 58. 37.100
  
    libavdevice    58.  9.103 / 58.  9.103
  
    libavfilter     7. 72.100 /  7. 72.100
  
    libswscale      5.  6.100 /  5.  6.100
  
    libswresample   3.  6.100 /  3.  6.100
  
    libpostproc    55.  6.100 / 55.  6.100
  
  -c:a: command not found
  
  
  
  [2]+  Stopped                 ffmpeg -re -i srt://0.0.0.0:25000?pkt_size=1316

  I have searched the issue, but I could not find anything similar.
  Does someone know what I have missed in the setup ?

  Everything is manually compiled through the guide, this was the final command I run to compile FFmpeg :

  cd ~/ffmpeg_sources && \
  
  wget -O ffmpeg-snapshot.tar.bz2 https://ffmpeg.org/releases/ffmpeg-snapshot.tar.bz2 && \
  
  tar xjvf ffmpeg-snapshot.tar.bz2 && \
  
  cd ffmpeg && \
  
  PATH="$HOME/bin:$PATH" PKG_CONFIG_PATH="$HOME/ffmpeg_build/lib/pkgconfig" ./configure \
  
    --prefix="$HOME/ffmpeg_build" \
  
    --pkg-config-flags="--static" \
  
    --extra-cflags="-I$HOME/ffmpeg_build/include" \
  
    --extra-ldflags="-L$HOME/ffmpeg_build/lib" \
  
    --extra-libs="-lpthread -lm" \
  
    --bindir="$HOME/bin" \
  
    --enable-gpl \
  
    --enable-libaom \
  
    --enable-libass \
  
    --enable-libfdk-aac \
  
    --enable-libfreetype \
  
    --enable-libmp3lame \
  
    --enable-libopus \
  
    --enable-libvorbis \
  
    --enable-libvpx \
  
    --enable-libx264 \
  
    --enable-libx265 \
  
    --enable-libsrt \
  
    --enable-nonfree && \
  
  PATH="$HOME/bin:$PATH" make && \
  
  make install && \
  
  hash -r

• doc/platform : drop reference to ffmpeg.zeranoe.com

  28 mars 2023, par Stefano Sabatini

  doc/platform : drop reference to ffmpeg.zeranoe.com
  It was closed in September 2020.
  Fix issue :
  
  http://trac.ffmpeg.org/ticket/9734
  

  • [DH] doc/platform.texi

• Unable to open audio file on Heroku using Librosa

  15 mars 2020, par Rohan Bojja

  I have a feature extraction REST API written in Python using the Librosa library (Extracting audio features), it receives an audio file through HTTP POST and responds with a list of features(such as MFCC,etc).

  Since librosa depends on SoundFile (libsndfile1 / libsndfile-dev), it doesn’t support all the formats, I’m converting the audio file using ffmpeg-python wrapper (https://kkroening.github.io/ffmpeg-python/) .

  It works just fine on my Windows 10 machine with Conda, but when I deploy it on Heroku, the librosa.load() functions returns an unknown format error, no matter what format I convert it to. I have tried FLAC, AIFF and WAV.

  My first guess is that the converted format isn’t supported by libsndfile1, but it works on my local server (plus, their documentation says AIFF and WAV are supported), so I’m a little lost.

  I have attached all the relevant snippets of code below, I can provide anything extra if necessary. Any help is highly appreciated. Thanks.

  UPDATE1 :

  I am using pipes instead of writing and reading from disk, worth a mention as the question could be misleading otherwise.

  The log :

  File "/app/app.py", line 31, in upload
  
  x , sr = librosa.load(audioFile,mono=True,duration=5)
  
  File "/app/.heroku/python/lib/python3.6/site-packages/librosa/core/audio.py", line 164, in load
  
  six.reraise(*sys.exc_info())
  
  File "/app/.heroku/python/lib/python3.6/site-packages/six.py", line 703, in reraise
  
  raise value
  
  File "/app/.heroku/python/lib/python3.6/site-packages/librosa/core/audio.py", line 129, in load
  
  with sf.SoundFile(path) as sf_desc:
  
  File "/app/.heroku/python/lib/python3.6/site-packages/soundfile.py", line 629, in __init__
  
  self._file = self._open(file, mode_int, closefd)
  
  File "/app/.heroku/python/lib/python3.6/site-packages/soundfile.py", line 1184, in _open
  
  "Error opening {0!r}: ".format(self.name))
  
  File "/app/.heroku/python/lib/python3.6/site-packages/soundfile.py", line 1357, in _error_check
  
  raise RuntimeError(prefix + _ffi.string(err_str).decode('utf-8', 'replace'))
  
  RuntimeError: Error opening <_io.BytesIO object at 0x7f46ad28beb8>: File contains data in an unknown format.
  
  10.69.244.94 - - [15/Mar/2020:12:37:28 +0000] "POST /receiveWav HTTP/1.1" 500 290 "-" "curl/7.55.1"

  Flask/Librosa code deployed on Heroku (app.py) :

  from flask import Flask, jsonify, request
  
  import scipy.optimize
  
  import os,pickle
  
  import numpy as np
  
  from sklearn.preprocessing import StandardScaler
  
  import librosa
  
  import logging
  
  import soundfile as sf
  
  from pydub import AudioSegment
  
  import subprocess as sp
  
  import ffmpeg
  
  from io import BytesIO
  
  
  
  logging.basicConfig(level=logging.DEBUG)
  
  
  
  app = Flask(__name__) 
  
  
  
  @app.route('/receiveWav',methods = ['POST'])
  
  def upload():
  
      if(request.method == 'POST'):
  
          f = request.files['file']
  
          app.logger.info(f'AUDIO FORMAT\n\n\n\n\n\n\n\n\n\n: {f}')
  
          proc = (
  
              ffmpeg.input('pipe:')
  
              .output('pipe:', format='aiff')
  
              .run_async(pipe_stdin=True,pipe_stdout=True, pipe_stderr=True)
  
          )
  
          audioFile,err = proc.communicate(input=f.read())
  
          audioFile =  BytesIO(audioFile)
  
          scaler = pickle.load(open("scaler.ok","rb"))
  
          x , sr = librosa.load(audioFile,mono=True,duration=5)
  
          y=x
  
          #Extract the features
  
          chroma_stft = librosa.feature.chroma_stft(y=y, sr=sr)
  
          spec_cent = librosa.feature.spectral_centroid(y=y, sr=sr)
  
          spec_bw = librosa.feature.spectral_bandwidth(y=y, sr=sr)
  
          rolloff = librosa.feature.spectral_rolloff(y=y, sr=sr)
  
          zcr = librosa.feature.zero_crossing_rate(y)
  
          rmse = librosa.feature.rms(y=y)
  
          mfcc = librosa.feature.mfcc(y=y, sr=sr)
  
          features = f'{np.mean(chroma_stft)} {np.mean(rmse)} {np.mean(spec_cent)} {np.mean(spec_bw)} {np.mean(rolloff)} {np.mean(zcr)}'    
  
          for e in mfcc:
  
              features += f' {np.mean(e)}'
  
          input_data2 = np.array([float(i) for i in features.split(" ")]).reshape(1,-1)
  
          input_data2 = scaler.transform(input_data2)
  
          return jsonify(input_data2.tolist())
  
  
  
  # driver function 
  
  if __name__ == '__main__':   
  
      app.run(debug = True) 

  Aptfile :

  libsndfile1
  
  libsndfile-dev
  
  libav-tools
  
  libavcodec-extra-53 
  
  libavcodec-extra-53
  
  ffmpeg 

  requirements.txt :

  aniso8601==8.0.0
  
  audioread==2.1.8
  
  certifi==2019.11.28
  
  cffi==1.14.0
  
  Click==7.0
  
  decorator==4.4.2
  
  ffmpeg-python==0.2.0
  
  Flask==1.1.1
  
  Flask-RESTful==0.3.8
  
  future==0.18.2
  
  gunicorn==20.0.4
  
  itsdangerous==1.1.0
  
  Jinja2==2.11.1
  
  joblib==0.14.1
  
  librosa==0.7.2
  
  llvmlite==0.31.0
  
  MarkupSafe==1.1.1
  
  marshmallow==3.2.2
  
  numba==0.48.0
  
  numpy==1.18.1
  
  pycparser==2.20
  
  pydub==0.23.1
  
  pytz==2019.3
  
  resampy==0.2.2
  
  scikit-learn==0.22.2.post1
  
  scipy==1.4.1
  
  six==1.14.0
  
  SoundFile==0.10.3.post1
  
  Werkzeug==1.0.0
  
  wincertstore==0.2