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The pirate bay depuis la Belgique
1er avril 2013, par
Mis à jour : Avril 2013
Langue : français
Type : Image
Autres articles (71)
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Keeping control of your media in your hands
13 avril 2011, parThe vocabulary used on this site and around MediaSPIP in general, aims to avoid reference to Web 2.0 and the companies that profit from media-sharing.
While using MediaSPIP, you are invited to avoid using words like "Brand", "Cloud" and "Market".
MediaSPIP is designed to facilitate the sharing of creative media online, while allowing authors to retain complete control of their work.
MediaSPIP aims to be accessible to as many people as possible and development is based on expanding the (...) -
Personnaliser en ajoutant son logo, sa bannière ou son image de fond
5 septembre 2013, parCertains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;
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MediaSPIP v0.2
21 juin 2013, parMediaSPIP 0.2 est la première version de MediaSPIP stable.
Sa date de sortie officielle est le 21 juin 2013 et est annoncée ici.
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Comme pour la version précédente, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...)
Sur d’autres sites (8827)
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FFMPEG commands isn't working in Android Q
5 février, par Krupali ShingalaTry 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

my Error log:
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]
 java.io.IOException: Cannot run program "/data/user/0/com.merger.cut/files/ffmpeg": error=13, Permission denied
 at java.lang.ProcessBuilder.start(ProcessBuilder.java:1050)
 at nl.bravobit.ffmpeg.ShellCommand.run(ShellCommand.java:15)
 at nl.bravobit.ffmpeg.FFcommandExecuteAsyncTask.doInBackground(FFcommandExecuteAsyncTask.java:43)
 at nl.bravobit.ffmpeg.FFcommandExecuteAsyncTask.doInBackground(FFcommandExecuteAsyncTask.java:12)
 at android.os.AsyncTask$3.call(AsyncTask.java:378)
 at java.util.concurrent.FutureTask.run(FutureTask.java:266)
 at java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1167)
 at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:641)
 at java.lang.Thread.run(Thread.java:919)
 Caused by: java.io.IOException: error=13, Permission denied
 at java.lang.UNIXProcess.forkAndExec(Native Method)
 at java.lang.UNIXProcess.<init>(UNIXProcess.java:133)
 at java.lang.ProcessImpl.start(ProcessImpl.java:141)
 at java.lang.ProcessBuilder.start(ProcessBuilder.java:1029)
 at nl.bravobit.ffmpeg.ShellCommand.run(ShellCommand.java:15) 
 at nl.bravobit.ffmpeg.FFcommandExecuteAsyncTask.doInBackground(FFcommandExecuteAsyncTask.java:43) 
 at nl.bravobit.ffmpeg.FFcommandExecuteAsyncTask.doInBackground(FFcommandExecuteAsyncTask.java:12) 
 at android.os.AsyncTask$3.call(AsyncTask.java:378) 
 at java.util.concurrent.FutureTask.run(FutureTask.java:266) 
 at java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1167) 
 at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:641) 
 at java.lang.Thread.run(Thread.java:919) 
2019-09-28 13:48:32.146 16041-16041/com.merger.cut E/FFMPEG :: on finish
</init>


Give me solution for above problem.


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ffmpeg - Merge back frames to a video with the same encoding
24 septembre 2019, par VuwoxI 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.movWhere 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 azail765This 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):
 File "C:\Users\zach.blair\Downloads\speechRecognition\New folder\speechRecognition3.py", line 17, in <module>
 audio = AudioSegment.from_file(filename,format="wav")
 File "C:\Users\zach.blair\AppData\Local\Programs\Python\Python37-32\lib\site-packages\pydub\audio_segment.py", line 704, in from_file
 p.returncode, p_err))
pydub.exceptions.CouldntDecodeError: Decoding failed. ffmpeg returned error code: 1
</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!


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