Recherche avancée

Médias (1)

Mot : - Tags -/iphone

Autres articles (104)

  • Soumettre améliorations et plugins supplémentaires

    10 avril 2011

    Si vous avez développé une nouvelle extension permettant d’ajouter une ou plusieurs fonctionnalités utiles à MediaSPIP, faites le nous savoir et son intégration dans la distribution officielle sera envisagée.
    Vous pouvez utiliser la liste de discussion de développement afin de le faire savoir ou demander de l’aide quant à la réalisation de ce plugin. MediaSPIP étant basé sur SPIP, il est également possible d’utiliser le liste de discussion SPIP-zone de SPIP pour (...)

  • Emballe médias : à quoi cela sert ?

    4 février 2011, par

    Ce plugin vise à gérer des sites de mise en ligne de documents de tous types.
    Il crée des "médias", à savoir : un "média" est un article au sens SPIP créé automatiquement lors du téléversement d’un document qu’il soit audio, vidéo, image ou textuel ; un seul document ne peut être lié à un article dit "média" ;

  • ANNEXE : Les plugins utilisés spécifiquement pour la ferme

    5 mars 2010, par

    Le site central/maître de la ferme a besoin d’utiliser plusieurs plugins supplémentaires vis à vis des canaux pour son bon fonctionnement. le plugin Gestion de la mutualisation ; le plugin inscription3 pour gérer les inscriptions et les demandes de création d’instance de mutualisation dès l’inscription des utilisateurs ; le plugin verifier qui fournit une API de vérification des champs (utilisé par inscription3) ; le plugin champs extras v2 nécessité par inscription3 (...)

Sur d’autres sites (10828)

  • Facing issue with mp4Parser

    20 novembre 2014, par Rohit

    I am trying to trim video with the help of mp4Parser. In output am getting trimmed video but just before video finishes it give error "Sorry cant Play this video". Video am making is H264.

       public static void main(String args) throws IOException {

       Movie movie = new MovieCreator()
               .build(new RandomAccessFileIsoBufferWrapperImpl(
                       new File(
                               "/sdcard/Videos11/"+args+".mp4")));

       List<track> tracks = movie.getTracks();
       Log.e("Tracks","Following are the Tracks:- "+tracks);
       movie.setTracks(new LinkedList<track>());
       // remove all tracks we will create new tracks from the old

       double startTime = 0.000;
       double endTime = 6.000;

       boolean timeCorrected = false;

       // Here we try to find a track that has sync samples. Since we can only
       // start decoding
       // at such a sample we SHOULD make sure that the start of the new
       // fragment is exactly
       // such a frame
       for (Track track : tracks) {
           if (track.getSyncSamples() != null
                   &amp;&amp; track.getSyncSamples().length > 0) {
               if (timeCorrected) {
                   // This exception here could be a false positive in case we
                   // have multiple tracks
                   // with sync samples at exactly the same positions. E.g. a
                   // single movie containing
                   // multiple qualities of the same video (Microsoft Smooth
                   // Streaming file)

                   throw new RuntimeException(
                           "The startTime has already been corrected by another track with SyncSample. Not Supported.");
               }
               startTime = correctTimeToNextSyncSample(track, startTime);
               endTime = correctTimeToNextSyncSample(track, endTime);
               timeCorrected = true;
           }
       }

       for (Track track : tracks) {
           long currentSample = 0;
           double currentTime = 0;
           long startSample = -1;
           long endSample = -1;

           for (int i = 0; i &lt; track.getDecodingTimeEntries().size(); i++) {
               TimeToSampleBox.Entry entry = track.getDecodingTimeEntries().get(i);
               for (int j = 0; j &lt; entry.getCount(); j++) {
                   // entry.getDelta() is the amount of time the current sample
                   // covers.

                   if (currentTime &lt;= startTime) {
                       // current sample is still before the new starttime
                       startSample = currentSample;
                   }
                   if (currentTime &lt;= endTime) {
                       // current sample is after the new start time and still
                       // before the new endtime
                       endSample = currentSample;
                   } else {
                       // current sample is after the end of the cropped video
                       break;
                   }
                   currentTime += (double) entry.getDelta()
                           / (double) track.getTrackMetaData().getTimescale();
                   currentSample++;
               }
           }
           movie.addTrack(new CroppedTrack(track, startSample, endSample));
       }

       IsoFile out = new DefaultMp4Builder().build(movie);

       String filePath = "sdcard/test"+i+".mp4";
       i++;
       File f = new File(filePath);
       FileOutputStream fos = new FileOutputStream(f);
       BufferedOutputStream bos = new BufferedOutputStream(fos, 65535);
       out.getBox(new IsoOutputStream(fos));
       bos.close();
       fos.flush();
       fos.close();

    }

    private static double correctTimeToNextSyncSample(Track track,
           double cutHere) {
       double[] timeOfSyncSamples = new double[track.getSyncSamples().length];
       long currentSample = 0;
       double currentTime = 0;
       for (int i = 0; i &lt; track.getDecodingTimeEntries().size(); i++) {
           TimeToSampleBox.Entry entry = track.getDecodingTimeEntries().get(i);
           for (int j = 0; j &lt; entry.getCount(); j++) {
               if (Arrays.binarySearch(track.getSyncSamples(),
                       currentSample + 1) >= 0) {
                   // samples always start with 1 but we start with zero
                   // therefore +1
                   timeOfSyncSamples[Arrays.binarySearch(
                           track.getSyncSamples(), currentSample + 1)] = currentTime;
               }
               currentTime += (double) entry.getDelta()
                       / (double) track.getTrackMetaData().getTimescale();
               currentSample++;
           }
       }
       for (double timeOfSyncSample : timeOfSyncSamples) {
           if (timeOfSyncSample > cutHere) {
               return timeOfSyncSample;
           }
       }
       return timeOfSyncSamples[timeOfSyncSamples.length - 1];
    }
    </track></track>

    Making video of 30 seconds and want to trim first 6 seconds. Working but in last(between 5 to 6 sec) it shows Cant play this video. Any help will be appreciated.

  • Pydub FFMPEG issue [closed]

    14 janvier, par Nikolai van den Hoven

    I am attempting to use FFMPEG with Pydub to create a program that chops .mp3 files into different words, each contained in their own .mp3 file, but when I run the script I am getting the following error :

    &#xA;

    PS C:\Users\nik> &amp; C:/Users/nik/AppData/Local/Microsoft/WindowsApps/python3.12.exe "d:/Python/Word Splitter.py"&#xA;C:\Users\nik\AppData\Local\Packages\PythonSoftwareFoundation.Python.3.12_qbz5n2kfra8p0\LocalCache\local-packages\Python312\site-packages\pydub\utils.py:170: RuntimeWarning: Couldn&#x27;t find ffmpeg or avconv - defaulting to ffmpeg, but may not work&#xA;  warn("Couldn&#x27;t find ffmpeg or avconv - defaulting to ffmpeg, but may not work", RuntimeWarning)&#xA;

    &#xA;

    This is the code I am using.

    &#xA;

    import os&#xA;from pydub import AudioSegment&#xA;from pydub.silence import split_on_silence&#xA;import speech_recognition as sr&#xA;AudioSegment.ffmpeg = r"D:\Python\ffmpeg\bin\ffmpeg.exe"&#xA;def mp3_to_words(mp3_file, output_folder):&#xA;    # Ensure output folder exists&#xA;    os.makedirs(output_folder, exist_ok=True)&#xA;&#xA;    # Load MP3 file&#xA;    print("Loading audio file...")&#xA;    audio = AudioSegment.from_mp3(mp3_file)&#xA;&#xA;    # Split audio into chunks using silence detection&#xA;    print("Splitting audio into chunks...")&#xA;    chunks = split_on_silence(&#xA;        audio,&#xA;        min_silence_len=200,  # Minimum silence duration in ms to consider as a split point&#xA;        silence_thresh=audio.dBFS - 14,  # Silence threshold relative to average loudness&#xA;        keep_silence=100  # Retain some silence in chunks&#xA;    )&#xA;&#xA;    recognizer = sr.Recognizer()&#xA;&#xA;    for i, chunk in enumerate(chunks):&#xA;        print(f"Processing chunk {i &#x2B; 1}/{len(chunks)}...")&#xA;&#xA;        # Save the chunk temporarily&#xA;        temp_file = os.path.join(output_folder, f"chunk_{i}.wav")&#xA;        chunk.export(temp_file, format="wav")&#xA;&#xA;        # Recognize words in the chunk&#xA;        with sr.AudioFile(temp_file) as source:&#xA;            audio_data = recognizer.record(source)&#xA;            try:&#xA;                text = recognizer.recognize_google(audio_data)&#xA;                words = text.split()&#xA;&#xA;                # Export each word as its own MP3&#xA;                word_start = 0&#xA;                for j, word in enumerate(words):&#xA;                    word_duration = len(chunk) // len(words)  # Approximate duration per word&#xA;                    word_audio = chunk[word_start:word_start &#x2B; word_duration]&#xA;                    word_file = os.path.join(output_folder, f"word_{i}_{j}.mp3")&#xA;                    word_audio.export(word_file, format="mp3")&#xA;                    word_start &#x2B;= word_duration&#xA;&#xA;            except sr.UnknownValueError:&#xA;                print(f"Could not understand chunk {i &#x2B; 1}.")&#xA;            except sr.RequestError as e:&#xA;                print(f"Could not request results; {e}")&#xA;&#xA;        # Clean up temporary file&#xA;        os.remove(temp_file)&#xA;&#xA;    print(f"Processed {len(chunks)} chunks. Word MP3s saved in {output_folder}.")&#xA;&#xA;if __name__ == "__main__":&#xA;    input_file = input("Enter the path to the MP3 file: ").strip()&#xA;    output_dir = input("Enter the output folder path: ").strip()&#xA;&#xA;    mp3_to_words(input_file, output_dir)&#xA;

    &#xA;

    I have added the Base FFMPEG folder and the bin folder within it to Windows PATH&#xA;My PATH variable on Windows 11,&#xA;But it does not show up in the variable when I typed PATH into cmd

    &#xA;

  • ffmpeg skips frames/packets while playing a file over http

    24 janvier 2020, par coban
    Microsoft Windows [Version 10.0.18362.592]  
    libavutil      56. 31.100 / 56. 31.100
    libavcodec     58. 54.100 / 58. 54.100
    libavformat    58. 29.100 / 58. 29.100
    libavdevice    58.  8.100 / 58.  8.100
    libavfilter     7. 57.100 /  7. 57.100
    libswscale      5.  5.100 /  5.  5.100
    libswresample   3.  5.100 /  3.  5.100
    libpostproc    55.  5.100 / 55.  5.100
    [hls @ 0600cac0] Skip ('#EXT-X-VERSION:3')    0KB sq=    0B f=0/0
    [hls @ 0600cac0] Opening 'http://198.144.145.236:1935/ondemand-1/_definst_/mp4:path2/FRENCH.MOVIES/2017.JOHN.WICK.CHAPTER.2.FRN.mp4/chunklist_w1481451763.m3u8' for reading
    [hls @ 0600cac0] Skip ('#EXT-X-VERSION:3')    0KB sq=    0B f=0/0
    [hls @ 0600cac0] Opening 'http://198.144.145.236:1935/ondemand-1/_definst_/mp4:path2/FRENCH.MOVIES/2017.JOHN.WICK.CHAPTER.2.FRN.mp4/media_w1481451763_0.ts' for reading
    [hls @ 0600cac0] Opening 'http://198.144.145.236:1935/ondemand-1/_definst_/mp4:path2/FRENCH.MOVIES/2017.JOHN.WICK.CHAPTER.2.FRN.mp4/media_w1481451763_1.ts' for reading
    Input #0, hls, from 'http://ok2.se:8000/movie/xxxxxxxxxx/67933.m3u8':
    Duration: 02:02:22.46, start: 0.000000, bitrate: N/A
    Program 0
    Metadata: 0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0
     variant_bitrate : 2788912
    Stream #0:0: Data: timed_id3 (ID3  / 0x20334449)
    Metadata:
     variant_bitrate : 2788912
    Stream #0:1: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 720x304 [SAR 1:1 DAR 45:19], 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc
    Metadata:
     variant_bitrate : 2788912
    Stream #0:2: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp
    Metadata:
     variant_bitrate : 2788912
    [http @ 060662c0] Opening 'http://198.144.145.236:1935/ondemand-1/_definst_/mp4:path2/FRENCH.MOVIES/2017.JOHN.WICK.CHAPTER.2.FRN.mp4/media_w1481451763_2.ts' for reading
    [hls @ 0600cac0] Opening 'http://198.144.145.236:1935/ondemand-1/_definst_/mp4:path2/FRENCH.MOVIES/2017.JOHN.WICK.CHAPTER.2.FRN.mp4/media_w1481451763_2.ts' for reading
    [http @ 0606fb40] Opening 'http://198.144.145.236:1935/ondemand-1/_definst_/mp4:path2/FRENCH.MOVIES/2017.JOHN.WICK.CHAPTER.2.FRN.mp4/media_w1481451763_3.ts' for reading
    [hls @ 0600cac0] Opening 'http://198.144.145.236:1935/ondemand-1/_definst_/mp4:path2/FRENCH.MOVIES/2017.JOHN.WICK.CHAPTER.2.FRN.mp4/media_w1481451763_3.ts' for reading
    [http @ 0c2442c0] Opening 'http://198.144.145.236:1935/ondemand-1/_definst_/mp4:path2/FRENCH.MOVIES/2017.JOHN.WICK.CHAPTER.2.FRN.mp4/media_w1481451763_4.ts' for reading
    [http @ 0606f8c0] Opening 'http://198.144.145.236:1935/ondemand-1/_definst_/mp4:path2/FRENCH.MOVIES/2017.JOHN.WICK.CHAPTER.2.FRN.mp4/media_w1481451763_5.ts' for reading

    I am trying to create an A/V player using ffmpeg library and Delphi. After puzzling and puzzling I found out how to read/decode/sync/display etc...

    I think this behaviour is in some specific situations, because I tested live streams and noticed no problems. In this case this is a video file, while playing my app and ffplay.exe (from ffmpeg not compiled by myself) behaving the same, and skipping packets "every time when the app has to wait while packetqueue is full ?" and loses about 10-30 seconds of audio and video.

    VLC doesn’t seem to have such a like behaviour, while as I understand from VLC credits, they are using ffmpeg or ffmpeg methods imported for decoding/encoding. inspecting vlc source doesn’t make much sense for me because I am a (noob) Delphi programmer. It’s like my main language is Dutch and try to translate form French to German language.

    Please any suggestions how to avoid this, because I think it has to do something with delivery of data from the server and/or read buffer of ffmpeg. If I increase the max size of packetqueue the skipping does happen later and if decrease the size, the skipping happens sooner.

    Including code from my app is not necessary because ffplay.exe has exact the same behavior. You can see debug from ffplay.exe.