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Autres articles (41)

  • Websites made ​​with MediaSPIP

    2 mai 2011, par

    This page lists some websites based on MediaSPIP.

  • Creating farms of unique websites

    13 avril 2011, par

    MediaSPIP platforms can be installed as a farm, with a single "core" hosted on a dedicated server and used by multiple websites.
    This allows (among other things) : implementation costs to be shared between several different projects / individuals rapid deployment of multiple unique sites creation of groups of like-minded sites, making it possible to browse media in a more controlled and selective environment than the major "open" (...)

  • Support de tous types de médias

    10 avril 2011

    Contrairement à beaucoup de logiciels et autres plate-formes modernes de partage de documents, MediaSPIP a l’ambition de gérer un maximum de formats de documents différents qu’ils soient de type : images (png, gif, jpg, bmp et autres...) ; audio (MP3, Ogg, Wav et autres...) ; vidéo (Avi, MP4, Ogv, mpg, mov, wmv et autres...) ; contenu textuel, code ou autres (open office, microsoft office (tableur, présentation), web (html, css), LaTeX, Google Earth) (...)

Sur d’autres sites (4848)

  • No such file or directory Error with FFMPEG + CarrierWave screenshot method

    10 juillet 2013, par dodgerogers747

    I am using AWS CORS to upload videos to my site, all of which works as planned.

    I have the following model method which runs as an after_create callback (for speed) to take a screenshot from the video file on AWS. I plan to move this out into a delayed job but I don't think this will solve this particular issue. Please advise if mistaken.

    I use FFMPEG to take a screenshot from the AWS self.file location, I then send the file to CarrierWave by saving the file to self.screenshot where it is uploaded to AWS.

    Approx. 50% of the time it errors out with Errno::ENOENT - No such file or directory for the location of the screenshot image.

    How can I rectify my code to remove this error and how come it only occurs around 50% of the time ? If anyone needs more code just shout.

    video.rb

    after_create :take_screenshot

    mount_uploader :screenshot, ImageUploader

     def take_screenshot
       location = "#{Rails.root}/public/uploads/tmp/screenshots/#{unique}_#{File.basename(file)}.jpg"
       system `ffmpeg #{log_level} -i #{self.file} -ss 00:00:0#{time_frame} -vframes 1 #{location}`
       logger.debug "Trying to take screenshot from #{self.file}"
       #pass the actual file to CarrierWave to handle the image upload
       self.screenshot = File.open(location)
       self.save
       logger.debug "Deleting tmp file: #{location}: #{File.delete(location)}" if self.screenshot.present?
     end

    def unique
       (0..6).map{(65+rand(26)).chr}.join
     end

    def log_level
       "-loglevel panic"
     end

     def time_frame
       rand(0..3)
     end

    Stack trace :

    Started POST "/videos" for 127.0.0.1 at 2013-07-10 03:58:49 +0800
    Processing by VideosController#create as JS
     Parameters: {"utf8"=>"✓", "authenticity_token"=>"6M1Ia+Ag2E3HVKH2PO/p7jewxSpMPdWeVHGA933Bzjw=", "video"=>{"file"=>"http://bucketname.s3.amazonaws.com/uploads/video/file/671a87fb-91de-4eaf-a38a-1b25c51798e5/Good_7iron.m4v"}}
     User Load (0.3ms)  SELECT `users`.* FROM `users` WHERE `users`.`id` = 9 LIMIT 1
      (0.1ms)  BEGIN
     SQL (0.2ms)  INSERT INTO `videos` (`created_at`, `file`, `question_id`, `screenshot`, `updated_at`, `user_id`) VALUES ('2013-07-09 19:58:49', 'http://bucketname.s3.amazonaws.com/uploads/video/file/671a87fb-91de-4eaf-a38a-1b25c51798e5/Good_7iron.m4v', NULL, NULL, '2013-07-09 19:58:49', 9)
    Trying to take screenshot from http://bucketname.s3.amazonaws.com/uploads/video/file/671a87fb-91de-4eaf-a38a-1b25c51798e5/Good_7iron.m4v
      (0.8ms)  ROLLBACK
    Completed 500 Internal Server Error in 3550ms

    Errno::ENOENT - No such file or directory - /Users/me/rails/project/public/uploads/tmp/screenshots/WCACLIC_Good_7iron.m4v.jpg:
     app/models/video.rb:24:in `initialize'
     app/models/video.rb:24:in `open'
     app/models/video.rb:24:in `take_screenshot'
  • 'C' program to pipeout audio file to FFMPEG and generate Video file

    9 mai 2017, par soflow

    I am attempting to write a short ’C’ program which reads in an Audio file using FFMPEG, processes that file using a ’C’ program, and then outputs a file via FFMEPG, which combines the new, modified audio together with a Video representation using the FFMPEG showwaves filter.

    At present the program attempts to do the following :-

    i) Read in an audio file, using pipein thorugh FFMPEG
    ii) Process the audio file using a portion of the ’C’ program
    iii) Pipeout the modified audio to FFMPEG, and generate a file using the ’showwaves’ filter in FFMEPG to create an MP4 file with audio and video.

    The following code run form the ommand line in FFMPEG generates the Audio/Video MP4 I want to create :-

    ffmpeg -y -f s16le -ar 44100 -ac 1 -i 12345678.wav  -i 12345678.wav  -filter_complex  "[0:a]showwaves=s=1280x720:mode=line:rate=25,format=yuv420p[v]" -map "[v]" -map 1:a:0 -codec:v libx264 -crf 21 -bf 2 -flags +cgop -pix_fmt yuv420p -codec:a aac -strict -2 -b:a 384k -r:a 48000 -movflags faststart  12345678.mp4

    "

    This code generates a processed audio file, and outputs it to a .wav file as required :-

    #include
    #include
    #include

    void main()
    {
    // Launch two instances of FFmpeg, one to read the original WAV
    // file and another to write the modified WAV file. In each case,
    // data passes between this program and FFmpeg through a pipe.
    FILE *pipein;
    FILE *pipeout;
    pipein  = popen("ffmpeg -i 12345678.wav -f s16le -ac 1 -", "r");
    pipeout = popen("ffmpeg -y -f s16le -ar 44100 -ac 1 -i - out.wav", "w");

    // Read, modify and write one sample at a time
    int16_t sample;
    int count, n=0;
    while(1)
    {
       count = fread(&sample, 2, 1, pipein); // read one 2-byte sample
       if (count != 1) break;
       ++n;
       sample = sample * sin(n * 5.0 * 2*M_PI / 44100.0);
       fwrite(&sample, 2, 1, pipeout);
    }

    // Close input and output pipes
    pclose(pipein);    
    pclose(pipeout);
    }

    (This code borrowed from ted Burke’s excellent post here)

    I have made an attempt as shown below, but this is not working :-

    #include
    #include
    #include

    void main()
    {
    // Launch two instances of FFmpeg, one to read the original WAV
    // file and another to write the modified WAV file. In each case,
    // data passes between this program and FFmpeg through a pipe.
    FILE *pipein;
    FILE *pipeout;
    pipein  = popen("ffmpeg -i 12345678.wav -f s16le -ac 1 -", "r");
    pipeout = popen("ffmpeg -y -f s16le -ar 44100 -ac 1 -i 12345678.wav  -i
    12345678.wav  -filter_complex  "
    [0:a]showwaves=s=1280x720:mode=line:rate=25,format=yuv420p[v]" -map "[v]"
    -map 1:a:0 -codec:v libx264 -crf 21 -bf 2 -flags +cgop -pix_fmt yuv420p -
    codec:a aac -strict -2 -b:a 384k -r:a 48000 -movflags faststart  
    12345678.mp4
    ", "w");


    // Read, modify and write one sample at a time
    int16_t sample;
    int count, n=0;
    while(1)
    {
       count = fread(&sample, 2, 1, pipein); // read one 2-byte sample
       if (count != 1) break;
       ++n;
       sample = sample * sin(n * 5.0 * 2*M_PI / 44100.0);
       fwrite(&sample, 2, 1, pipeout);
    }

    // Close input and output pipes
    pclose(pipein);    
    pclose(pipeout);
    }    

    Ideally someone can suggest an improved version of the pipeout command above - alternately another process to achieve this would be interesting

    * EDIT *

    Thanks to @Mulvya, the revised pipeout line is now :-

    pipeout = popen("ffmpeg -y -f s16le -ar 44100 -ac 1 -i -  -filter_complex  "[0:a]showwaves=s=1280x720:mode=line:rate=25,format=yuv420p[v]" -map "[v]" -map 1:a:0 -codec:v libx264 -crf 21 -bf 2 -flags +cgop -pix_fmt yuv420p -codec:a aac -strict -2 -b:a 384k -r:a 48000 -movflags faststart  12345678.mp4

    ", "w") ;

    On compiling with gcc I get the following error messages :-

    avtovid2.c: In function \u2018main\u2019:

    wavtovid2.c:13:83: error: expected \u2018]\u2019 before \u2018:\u2019
    token
    pipeout = popen("ffmpeg -y -f s16le -ar 44100 -ac 1 -i -  -
    filter_complex  "
    [0:a]showwaves=s=1280x720:mode=line:rate=25,format=yuv420p[v]" -map "[v]"
    -map 1:a:0 -codec:v libx264 -crf 21 -bf 2 -flags +cgop -pix_fmt yuv420p -
    codec:a aac -strict -2 -b:a 384k -r:a 48000 -movflags faststart  
    12345678.mp4

    ^
    wavtovid2.c:13:86: error: expected \u2018)\u2019 before
    \u2018showwaves\u2019
    pipeout = popen("ffmpeg -y -f s16le -ar 44100 -ac 1 -i -  -
    filter_complex  "
    [0:a]showwaves=s=1280x720:mode=line:rate=25,format=yuv420p[v]" -map "[v]"
    -map 1:a:0 -codec:v libx264 -crf 21 -bf 2 -flags +cgop -pix_fmt yuv420p -
    codec:a aac -strict -2 -b:a 384k -r:a 48000 -movflags faststart  
    12345678.mp4

    ^
    wavtovid2.c:13:98: error: invalid suffix "x720" on integer constant
    pipeout = popen("ffmpeg -y -f s16le -ar 44100 -ac 1 -i -  -
    filter_complex  "
    [0:a]showwaves=s=1280x720:mode=line:rate=25,format=yuv420p[v]" -map "[v]"
    -map 1:a:0 -codec:v libx264 -crf 21 -bf 2 -flags +cgop -pix_fmt yuv420p -
    codec:a aac -strict -2 -b:a 384k -r:a 48000 -movflags faststart  
    12345678.mp4

    ^
    wavtovid2.c:13:153: warning: missing terminating " character
    pipeout = popen("ffmpeg -y -f s16le -ar 44100 -ac 1 -i -  -
    filter_complex  "
    [0:a]showwaves=s=1280x720:mode=line:rate=25,format=yuv420p[v]" -map "[v]"
    -map 1:a:0 -codec:v libx264 -crf 21 -bf 2 -flags +cgop -pix_fmt yuv420p -
    codec:a aac -strict -2 -b:a 384k -r:a 48000 -movflags faststart  
    12345678.mp4

    ^
    wavtovid2.c:13:86: error: missing terminating " character
    pipeout = popen("ffmpeg -y -f s16le -ar 44100 -ac 1 -i -  -
    filter_complex  "
    [0:a]showwaves=s=1280x720:mode=line:rate=25,format=yuv420p[v]" -map "[v]"
    -map 1:a:0 -codec:v libx264 -crf 21 -bf 2 -flags +cgop -pix_fmt yuv420p -
    codec:a aac -strict -2 -b:a 384k -r:a 48000 -movflags faststart  
    12345678.mp4

    ^
    wavtovid2.c:14:6: warning: missing terminating " character
    ", "w");
     ^
    wavtovid2.c:14:1: error: missing terminating " character
    ", "w");
    ^
    wavtovid2.c:13:21: warning: passing argument 1 of \u2018popen\u2019 makes
    pointer from integer without a cast
    pipeout = popen("ffmpeg -y -f s16le -ar 44100 -ac 1 -i -  -
    filter_complex  "
    [0:a]showwaves=s=1280x720:mode=line:rate=25,format=yuv420p[v]" -map "[v]"
    -map 1:a:0 -codec:v libx264 -crf 21 -bf 2 -flags +cgop -pix_fmt yuv420p -
    codec:a aac -strict -2 -b:a 384k -r:a 48000 -movflags faststart  
    12345678.mp4
                    ^
    In file included from wavtovid2.c:1:0:
    /usr/include/stdio.h:872:14: note: expected \u2018const char *\u2019 but
    argument is of type \u2018char\u2019
    extern FILE *popen (const char *__command, const char *__modes) __wur;
             ^
    wavtovid2.c:13:15: error: too few arguments to function \u2018popen\u2019
    pipeout = popen("ffmpeg -y -f s16le -ar 44100 -ac 1 -i -  -
    filter_complex  "
    [0:a]showwaves=s=1280x720:mode=line:rate=25,format=yuv420p[v]" -map "[v]"
    -map 1:a:0 -codec:v libx264 -crf 21 -bf 2 -flags +cgop -pix_fmt yuv420p -
    codec:a aac -strict -2 -b:a 384k -r:a 48000 -movflags faststart  
    12345678.mp4
              ^
    In file included from wavtovid2.c:1:0:
    /usr/include/stdio.h:872:14: note: declared here
    extern FILE *popen (const char *__command, const char *__modes) __wur;
             ^
    wavtovid2.c:32:1: error: expected \u2018;\u2019 before \u2018}\u2019
    token
    }
  • Speech recognition with python-telegram-bot without downloading an audio file

    25 juin 2022, par linz

    I'm developing a telegram bot in which the user sends a voice message, the bot transcribes it and sends back what was said in text.
For that I am using the python-telegram-bot library and the speech_recognition library with the google engine.
My problem is, the voice messages sent by the users are .mp3, however in order to transcribe them i need to convert them to .wav. In order to do that I have to download the file sent to the bot.
Is there a way to avoid that ? I understand this is not an efficient and a safe way to do this since many active users at once will result in race conditions and takes a lot of space.

    


    
def voice_handler(update, context):
    bot = context.bot
    file = bot.getFile(update.message.voice.file_id)
    file.download('voice.mp3')
    filename = "voice.wav"
    
    # convert mp3 to wav file
    subprocess.call(['ffmpeg', '-i', 'voice.mp3',
                         'voice.wav', '-y'])

    # initialize the recognizer
    r = sr.Recognizer()
    
    # open the file
    with sr.AudioFile(filename) as source:
    
        # listen for the data (load audio to memory)
        audio_data = r.record(source)
        # recognize (convert from speech to text)
        text = r.recognize_google(audio_data, language='ar-AR')
        
        
def main() -> None:
    updater.dispatcher.add_handler(MessageHandler(Filters.voice, voice_handler))