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• what is ffmpeg exit code 2 ? and also output error array is empty and not working in php

  14 juin 2019, par Mehdi Zamani

  I use ffmpeg to convert bitrate to 128 but not working in php

   exec("ffmpeg -i input.mp3 -codec:a libmp3lame -b:a 128k output().mp3 2>&1", 
  
   $output, $exit_code);
  
   if ($exit_code!= 0) {
  
      $data['message'][] = "Error";
  
   }
  
  
  
   print_r($output);
  
   print_r($exit_code);
  
   exit;

  After running this code show error code 2.
  The output is an empty array and also exit_code is 2 and not create output.mp3 file.

  I already study How can I find out what this ffmpeg error code means ? but this isn’t my problem and don’t explain error code 2 or error code 2 is not defined. My problem is dont show any error and error message is empty just exit_code show 2 that means is some error happened.

• x264 static lib libx264.a occur an error : asm code, recompile with -fPIC [closed]

  20 juin 2019, par thunvan

  compile with static libs :

  yasm -I. -I. -DARCH_X86_64=1 -I./common/x86/ -f elf64 -Worphan-labels -DSTACK_ALIGNMENT=32 -DPIC -DHIGH_BIT_DEPTH=0 -DBIT_DEPTH=8 -o common/x86/cabac-a.o common/x86/cabac-a.asm

  link the libx264.a to another so libs

  occur : (cabac-a.o): relocation R_X86_64_PC32 against symbol `x264_cabac_range_lps' can not be used when making a shared object; recompile with -fPIC

  used -DPIC with yasm, also have error.

  gcc -Wno-maybe-uninitialized -Wshadow -O3 -ffast-math -m64  -Wall -I. -I. -std=gnu99 -D_GNU_SOURCE -mpreferred-stack-boundary=5  -I/usr/local/include  -I/usr/local/include -fPIC -fomit-frame-pointer -fno-tree-vectorize   -c -o encoder/cavlc.o encoder/cavlc.c
  
  gcc -Wno-maybe-uninitialized -Wshadow -O3 -ffast-math -m64  -Wall -I. -I. -std=gnu99 -D_GNU_SOURCE -mpreferred-stack-boundary=5  -I/usr/local/include  -I/usr/local/include -fPIC -fomit-frame-pointer -fno-tree-vectorize   -c -o encoder/encoder.o encoder/encoder.c
  
  gcc -Wno-maybe-uninitialized -Wshadow -O3 -ffast-math -m64  -Wall -I. -I. -std=gnu99 -D_GNU_SOURCE -mpreferred-stack-boundary=5  -I/usr/local/include  -I/usr/local/include -fPIC -fomit-frame-pointer -fno-tree-vectorize   -c -o encoder/lookahead.o encoder/lookahead.c
  
  gcc -Wno-maybe-uninitialized -Wshadow -O3 -ffast-math -m64  -Wall -I. -I. -std=gnu99 -D_GNU_SOURCE -mpreferred-stack-boundary=5  -I/usr/local/include  -I/usr/local/include -fPIC -fomit-frame-pointer -fno-tree-vectorize   -c -o common/threadpool.o common/threadpool.c
  
  gcc -Wno-maybe-uninitialized -Wshadow -O3 -ffast-math -m64  -Wall -I. -I. -std=gnu99 -D_GNU_SOURCE -mpreferred-stack-boundary=5  -I/usr/local/include  -I/usr/local/include -fPIC -fomit-frame-pointer -fno-tree-vectorize   -c -o common/x86/mc-c.o common/x86/mc-c.c
  
  gcc -Wno-maybe-uninitialized -Wshadow -O3 -ffast-math -m64  -Wall -I. -I. -std=gnu99 -D_GNU_SOURCE -mpreferred-stack-boundary=5  -I/usr/local/include  -I/usr/local/include -fPIC -fomit-frame-pointer -fno-tree-vectorize   -c -o common/x86/predict-c.o common/x86/predict-c.c
  
  yasm -I. -I. -DARCH_X86_64=1 -I./common/x86/ -f elf64 -Worphan-labels -DSTACK_ALIGNMENT=32 -DPIC -DHIGH_BIT_DEPTH=0 -DBIT_DEPTH=8 -o common/x86/const-a.o common/x86/const-a.asm
  
  yasm -I. -I. -DARCH_X86_64=1 -I./common/x86/ -f elf64 -Worphan-labels -DSTACK_ALIGNMENT=32 -DPIC -DHIGH_BIT_DEPTH=0 -DBIT_DEPTH=8 -o common/x86/cabac-a.o common/x86/cabac-a.asm
  
  yasm -I. -I. -DARCH_X86_64=1 -I./common/x86/ -f elf64 -Worphan-labels -DSTACK_ALIGNMENT=32 -DPIC -DHIGH_BIT_DEPTH=0 -DBIT_DEPTH=8 -o common/x86/dct-a.o common/x86/dct-a.asm
  
  yasm -I. -I. -DARCH_X86_64=1 -I./common/x86/ -f elf64 -Worphan-labels -DSTACK_ALIGNMENT=32 -DPIC -DHIGH_BIT_DEPTH=0 -DBIT_DEPTH=8 -o common/x86/deblock-a.o common/x86/deblock-a.asm
  
  yasm -I. -I. -DARCH_X86_64=1 -I./common/x86/ -f elf64 -Worphan-labels -DSTACK_ALIGNMENT=32 -DPIC -DHIGH_BIT_DEPTH=0 -DBIT_DEPTH=8 -o common/x86/mc-a.o common/x86/mc-a.asm
  
  yasm -I. -I. -DARCH_X86_64=1 -I./common/x86/ -f elf64 -Worphan-labels -DSTACK_ALIGNMENT=32 -DPIC -DHIGH_BIT_DEPTH=0 -DBIT_DEPTH=8 -o common/x86/mc-a2.o common/x86/mc-a2.asm
  
  yasm -I. -I. -DARCH_X86_64=1 -I./common/x86/ -f elf64 -Worphan-labels -DSTACK_ALIGNMENT=32 -DPIC -DHIGH_BIT_DEPTH=0 -DBIT_DEPTH=8 -o common/x86/pixel-a.o common/x86/pixel-a.asm
  
  yasm -I. -I. -DARCH_X86_64=1 -I./common/x86/ -f elf64 -Worphan-labels -DSTACK_ALIGNMENT=32 -DPIC -DHIGH_BIT_DEPTH=0 -DBIT_DEPTH=8 -o common/x86/predict-a.o common/x86/predict-a.asm

  how to compile asm code with yasm, and result to pic code successfully ?

• 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!