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• FFMPEG commands isn't working in Android Q

  5 février, par Krupali Shingala

  Try this command for merge two audio files, but its not working in Android 10.0(Q) while targeting sdk 29.
But, this command is completely working on targeting sdk 24 to 28.

  



  I am using this library of FFMPEG implementation 'nl.bravobit:android-ffmpeg:1.1.7'

  



  "-y", "-i", path1, "-i", path2, "-filter_complex", "[0:0][1:0] amix=inputs=2:duration=longest", "-c:a", "libmp3lame", savedPath&#xA;&#xA;my Error log:&#xA;2019-09-28 13:48:32.037 16041-16166/com.merger.cut E/FFmpeg: Exception while trying to run: [/data/user/0/com..merger.cut/files/ffmpeg, -y, -i, /storage/emulated/0/Music/song1.mp3, -i, /storage/emulated/0/Music/song2.mp3, -filter_complex, [0:0][1:0] amix=inputs=2:duration=longest, -c:a, libmp3lame, /storage/emulated/0/merger/Merge_1569658695254.mp3]&#xA;    java.io.IOException: Cannot run program "/data/user/0/com.merger.cut/files/ffmpeg": error=13, Permission denied&#xA;        at java.lang.ProcessBuilder.start(ProcessBuilder.java:1050)&#xA;        at nl.bravobit.ffmpeg.ShellCommand.run(ShellCommand.java:15)&#xA;        at nl.bravobit.ffmpeg.FFcommandExecuteAsyncTask.doInBackground(FFcommandExecuteAsyncTask.java:43)&#xA;        at nl.bravobit.ffmpeg.FFcommandExecuteAsyncTask.doInBackground(FFcommandExecuteAsyncTask.java:12)&#xA;        at android.os.AsyncTask$3.call(AsyncTask.java:378)&#xA;        at java.util.concurrent.FutureTask.run(FutureTask.java:266)&#xA;        at java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1167)&#xA;        at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:641)&#xA;        at java.lang.Thread.run(Thread.java:919)&#xA;     Caused by: java.io.IOException: error=13, Permission denied&#xA;        at java.lang.UNIXProcess.forkAndExec(Native Method)&#xA;        at java.lang.UNIXProcess.<init>(UNIXProcess.java:133)&#xA;        at java.lang.ProcessImpl.start(ProcessImpl.java:141)&#xA;        at java.lang.ProcessBuilder.start(ProcessBuilder.java:1029)&#xA;        at nl.bravobit.ffmpeg.ShellCommand.run(ShellCommand.java:15)&#xA0;&#xA;        at nl.bravobit.ffmpeg.FFcommandExecuteAsyncTask.doInBackground(FFcommandExecuteAsyncTask.java:43)&#xA0;&#xA;        at nl.bravobit.ffmpeg.FFcommandExecuteAsyncTask.doInBackground(FFcommandExecuteAsyncTask.java:12)&#xA0;&#xA;        at android.os.AsyncTask$3.call(AsyncTask.java:378)&#xA0;&#xA;        at java.util.concurrent.FutureTask.run(FutureTask.java:266)&#xA0;&#xA;        at java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1167)&#xA0;&#xA;        at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:641)&#xA0;&#xA;        at java.lang.Thread.run(Thread.java:919)&#xA0;&#xA;2019-09-28 13:48:32.146 16041-16041/com.merger.cut E/FFMPEG&#xA0;:: on finish&#xA;</init>

  



  Give me solution for above problem.

  


• ffmpeg - Merge back frames to a video with the same encoding

  24 septembre 2019, par Vuwox

  I have a video encoded using H264 at 23.98 fps, for a duration of 00:00:06.42.

  I extracted the frames from that video, and then I processed those images one-by-one. Now I want to put them back together as a video, but I want to be the same as the source video (same duration, same audio, etc).

  Whatever I tried gives something different. The duration is always greater (around 00:00:06.59), the audio seems to be up to the end of the video (as expected), but the frame are not encoded properly, and they seems to freeze at the end and the audio continue.

  The one that look almost the same except the freeze at the end look like this :

  ffmpeg -i input.mov \
  
         -pattern_type glob -i 'result_*.tif' 
  
         -map 1 -map 0:a \
  
         -map_metadata 0 \
  
         -map_metadata:s:v 0:s:v \
  
         -map_metadata:s:a 0:s:a \
  
         output.mov

  Where I use the metadata and the audio from the input video, and use the frames from my second input.

  EDIT : As suggested here the details of the source video.

  ffmpeg version 2.8.15 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-28)
  
    configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --extra-ldflags='-Wl,-z,relro ' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd --enable-gnutls --enable-ladspa --enable-libass --enable-libcdio --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libx264 --enable-libx265 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Transparent.mov':
  
    Metadata:
  
      major_brand     : qt
  
      minor_version   : 0
  
      compatible_brands: qt
  
      creation_time   : 2019-09-17 22:06:44
  
    Duration: 00:00:06.42, start: 0.000000, bitrate: 47798 kb/s
  
      Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 113 kb/s (default)
  
      Metadata:
  
        creation_time   : 2019-09-17 22:06:44
  
        handler_name    : Core Media Data Handler
  
      Stream #0:1(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 3840x2160 [SAR 1:1 DAR 16:9], 47541 kb/s, 23.98 fps, 23.98 tbr, 24k tbn, 48k tbc (default)
  
      Metadata:
  
        creation_time   : 2019-09-17 22:06:44
  
        handler_name    : Core Media Data Handler
  
        encoder         : H.264
  
        timecode        : 00:00:00:00
  
      Stream #0:2(und): Data: none (tmcd / 0x64636D74), 0 kb/s (default)
  
      Metadata:
  
        creation_time   : 2019-09-17 22:06:44
  
        handler_name    : Core Media Data Handler
  
        timecode        : 00:00:00:00
  
  At least one output file must be specified

• pydub.exceptions.CouldntDecodeError : Decoding failed. ffmpeg returned error code : 1

  9 avril, par azail765

  This script will work on a 30 second wav file but not a 10 minutes phone call also in wav format. Any help would be appreciated

  


  I've downloaded ffmpeg.

  


  # Import necessary libraries 
from pydub import AudioSegment 
import speech_recognition as sr 
import os
import pydub


chunk_count = 0
directory = os.fsencode(r'C:\Users\zach.blair\Downloads\speechRecognition\New folder')
# Text file to write the recognized audio 
fh = open("recognized.txt", "w+")
for file in os.listdir(directory):
     filename = os.fsdecode(file)
     if filename.endswith(".wav"):
        chunk_count += 1
             # Input audio file to be sliced 
        audio = AudioSegment.from_file(filename,format="wav") 
          
        ''' 
        Step #1 - Slicing the audio file into smaller chunks. 
        '''
        # Length of the audiofile in milliseconds 
        n = len(audio) 
          
        # Variable to count the number of sliced chunks 
        counter = 1
          
         
          
        # Interval length at which to slice the audio file. 
        interval = 20 * 1000
          
        # Length of audio to overlap.  
        overlap = 1 * 1000
          
        # Initialize start and end seconds to 0 
        start = 0
        end = 0
          
        # Flag to keep track of end of file. 
        # When audio reaches its end, flag is set to 1 and we break 
        flag = 0
          
        # Iterate from 0 to end of the file, 
        # with increment = interval 
        for i in range(0, 2 * n, interval): 
              
            # During first iteration, 
            # start is 0, end is the interval 
            if i == 0: 
                start = 0
                end = interval 
          
            # All other iterations, 
            # start is the previous end - overlap 
            # end becomes end + interval 
            else: 
                start = end - overlap 
                end = start + interval  
          
            # When end becomes greater than the file length, 
            # end is set to the file length 
            # flag is set to 1 to indicate break. 
            if end >= n: 
                end = n 
                flag = 1
          
            # Storing audio file from the defined start to end 
            chunk = audio[start:end] 
          
            # Filename / Path to store the sliced audio 
            filename = str(chunk_count)+'chunk'+str(counter)+'.wav'
          
            # Store the sliced audio file to the defined path 
            chunk.export(filename, format ="wav") 
            # Print information about the current chunk 
            print(str(chunk_count)+str(counter)+". Start = "
                                +str(start)+" end = "+str(end)) 
          
            # Increment counter for the next chunk 
            counter = counter + 1
              
          
            AUDIO_FILE = filename 
            
            # Initialize the recognizer 
            r = sr.Recognizer() 
          
            # Traverse the audio file and listen to the audio 
            with sr.AudioFile(AUDIO_FILE) as source: 
                audio_listened = r.listen(source) 
          
            # Try to recognize the listened audio 
            # And catch expections. 
            try:     
                rec = r.recognize_google(audio_listened) 
                  
                # If recognized, write into the file. 
                fh.write(rec+" ") 
              
            # If google could not understand the audio 
            except sr.UnknownValueError: 
                    print("Empty Value") 
          
            # If the results cannot be requested from Google. 
            # Probably an internet connection error. 
            except sr.RequestError as e: 
                print("Could not request results.") 
          
            # Check for flag. 
            # If flag is 1, end of the whole audio reached. 
            # Close the file and break.                  
fh.close()    


  


  I get this error on audio = AudioSegment.from_file(filename,format="wav") :

  


  Traceback (most recent call last):&#xA;  File "C:\Users\zach.blair\Downloads\speechRecognition\New folder\speechRecognition3.py", line 17, in <module>&#xA;    audio = AudioSegment.from_file(filename,format="wav")&#xA;  File "C:\Users\zach.blair\AppData\Local\Programs\Python\Python37-32\lib\site-packages\pydub\audio_segment.py", line 704, in from_file&#xA;    p.returncode, p_err))&#xA;pydub.exceptions.CouldntDecodeError: Decoding failed. ffmpeg returned error code: 1&#xA;</module>

  


  Output from ffmpeg/avlib :

  


    ffmpeg version N-95027-g8c90bb8ebb Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 9.2.1 (GCC) 20190918
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  libavutil      56. 35.100 / 56. 35.100
  libavcodec     58. 58.101 / 58. 58.101
  libavformat    58. 33.100 / 58. 33.100
  libavdevice    58.  9.100 / 58.  9.100
  libavfilter     7. 58.102 /  7. 58.102
  libswscale      5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, wav, from '2a.wav.wav':
  Duration: 00:09:52.95, bitrate: 64 kb/s
    Stream #0:0: Audio: pcm_mulaw ([7][0][0][0] / 0x0007), 8000 Hz, mono, s16, 64 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_mulaw (native) -> pcm_s8 (native))
Press [q] to stop, [?] for help
[wav @ 0000024307974400] pcm_s8 codec not supported in WAVE format
Could not write header for output file #0 (incorrect codec parameters ?): Function not implemented
Error initializing output stream 0:0 -- 
Conversion failed!