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  • MediaSPIP version 0.1 Beta

    16 avril 2011, par

    MediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
    Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
    Pour avoir une installation fonctionnelle, 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 (...)

  • MediaSPIP 0.1 Beta version

    25 avril 2011, par

    MediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
    The zip file provided here only contains the sources of MediaSPIP in its standalone version.
    To get a working installation, you must manually install all-software dependencies on the server.
    If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)

  • Personnaliser en ajoutant son logo, sa bannière ou son image de fond

    5 septembre 2013, par

    Certains 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 ;

Sur d’autres sites (11126)

  • FFMPEG pushed RTMP stream not working on Android & iPhone

    1er décembre 2015, par BlackDivine

    I have to make a semi-live-stream. I used Nginx-rtmp module and then pushed content to it via ffmpeg using :

    ffmpeg -re -i content.mp4 -r 25 -f fvl "rtmp://rtmp.server.here"

    The stream runs fine when I open it in VLC from "rtmp ://rtmp.server.here"

    But I also have to make iPhone and Android apps that play these streams. And that’s the problem, the stream doesn’t work on Android and iPhone.

    If I use Wowza streaming cloud and stream to Wowza cloud instead of my own nginx-rtmp server then the same app written for Android & iPhone can playback the stream just fine.

    Now either nginx-rtmp is not working right, or what else ? I’ve also tried crtmpserver and the same thing happens.

    What I want to acheive :
    I have to develop a system where we can upstream a TV-Channel (have rights for it) to a server and then make a website, android app & iPhone app so consumers can watch the live channel.

    The uploading part I have a clue of, probably a TV tuner card and Open Broadcast Software to stream it to server. But the Live playback is new to me.


    UPDATE : I also used ffprobe and here’s the output. (See the last line)

    munir@munir-HP-ProBook-450-G2:~$ ffprobe rtmp://rtmp.server.here
    ffprobe version 2.6.2 Copyright (c) 2007-2015 the FFmpeg developers
     built with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
     configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable-libvo-aacenc --enable-libvidstab
     libavutil      54. 20.100 / 54. 20.100
     libavcodec     56. 26.100 / 56. 26.100
     libavformat    56. 25.101 / 56. 25.101
     libavdevice    56.  4.100 / 56.  4.100
     libavfilter     5. 11.102 /  5. 11.102
     libavresample   2.  1.  0 /  2.  1.  0
     libswscale      3.  1.101 /  3.  1.101
     libswresample   1.  1.100 /  1.  1.100
     libpostproc    53.  3.100 / 53.  3.100
    [flv @ 0x267cc60] Stream discovered after head already parsed
       Last message repeated 1 times
    Input #0, flv, from 'rtmp://stage.funworldpk.com/live':
     Metadata:
       Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
       displayWidth    : 320
       displayHeight   : 240
       fps             : 20
       profile         :
       level           :
     Duration: 00:00:00.00, start: 288.763000, bitrate: N/A
       Stream #0:0: Video: h264 (High), yuv420p, 320x240 [SAR 1:1 DAR 4:3], 20 fps, 20 tbr, 1k tbn, 40 tbc
       Stream #0:1: Data: none
       Stream #0:2: Audio: aac (LC), 22050 Hz, stereo, fltp
    Unsupported codec with id 0 for input stream 1

    Update 2 :
    I got my stream working by using Licensed copy of Wowza streaming server. Everything works now. But obviously this will not be an option for everyone that’s why I am not posting it as an answer.

  • Recording audio with MediaRecorder on iPhone with Safari and Chrome only 1 second long ? Mimetype and FFMPEG problem ?

    9 mai 2023, par Avatar

    I am using MediaRecorder to record the Microphone audio on a website.

    


    Javascript :

    


    var blob;
var blob_url;
var stream;
var recorder;
var chunks;

var media = {
    tag: 'audio',
    type: 'audio/ogg',
    ext: '.ogg',
    gUM: {audio: true}
};

navigator.mediaDevices.getUserMedia(media.gUM).then(_stream => 
{
    stream = _stream;

    recorder = new MediaRecorder(stream);

    recorder.ondataavailable = e => 
    {
        // push data to chunks
        chunks.push(e.data);

        // recording has been stopped
        if(recorder.state == 'inactive') 
        {
            // audio data available
            blob = new Blob(chunks, {type: media.type });
            blob_url = URL.createObjectURL(blob);
            
            // send data to server
            uploadfile_audio();
        }
    };

    if(typeof(recorder)=='undefined')
    {
        alert('No microphone access');
        return;
    }

    chunks = [];
    recorder.start();
}


// when stop button is clicked
recorder.stop();
stream.getTracks().forEach( track => track.stop() );


    


    The audio stream (ogg format) is sent to the server.

    


    Since iPad/iPhone do not play ogg files, the recording file is converted to "mp3" using FFMPEG.

    


    This file is stored on the server.

    

    


    This works on Windows and MAC (Chrome and Safari), also on iPad (Safari) but not properly on iPhone (Chrome/Safari). Version : iPhone iOS 15.1.

    


    On iPhone the recording file is only 0:01 min in length. Size is always 17277 Bytes.

    


    What could be the issue ? (I cannot debug because I don't have a Mac.)

    


    Does the stream get interrupted ? Is the recording stopped after 1 second ?

    


    Update 1 :

    


    I have checked the incoming filesize of the browser-recorded file serverside. It seems to be coming in properly, because there are different sizes such as 184 kB.

    


    My guess is now that FFMPEG cannot handle the incoming file correctly. Which might have the wrong mimetype set in Javascript with type: 'audio/ogg',. Is another format needed ?

    


    The conversion code serverside :

    


    PHP

    


    $mp3file = shell_exec("ffmpeg -i ".$file_locationtmp." -vn -ar 44100 -ac 2 -b:a 128k ".$file_locationtmp.".mp3");


    


    I would need to find out the audio recording format used by iPhone but I couldn't.

    


    I tried to find the supporting mimetypes using https://developer.mozilla.org/en-US/docs/Web/API/MediaRecorder/isTypeSupported - however, it shows that NO mimetypes are supported on iPhone (neither in Chrome nor Safari).

    


    Update 2 :

    


    I used ffprobe to get the recording file information. It says Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 2234 kb/s (default)

    


    Update 3 :

    


    It seems to be a problem with FFMPEG. See my new question How to convert AAC/MP4A to MP3 using FFMPEG in full length ? Audio file gets cut off after 1 second

    


  • Recording MP3 file from iphone using FFMPEG

    16 octobre 2014, par harit shah

    I am trying to record an mp3 file from iphone. It seems it is not possible so next option I am trying is recording it with aac encoding and convert it with ffmpeg exe to mp3.

    I am recording the m4a file with following code :

    // Set the audio file
    NSArray *pathComponents = [NSArray arrayWithObjects:
                              [NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES) lastObject],
                              @"MyAudioMemo.aac",
                              nil];
    NSURL *outputFileURL = [NSURL fileURLWithPathComponents:pathComponents];

    // Setup audio session
    AVAudioSession *session = [AVAudioSession sharedInstance];
    [session setCategory:AVAudioSessionCategoryPlayAndRecord error:nil];
    UInt32 audioRouteOverride = kAudioSessionOverrideAudioRoute_Speaker;
    AudioSessionSetProperty (kAudioSessionProperty_OverrideAudioRoute,sizeof (audioRouteOverride),&audioRouteOverride);

    NSMutableDictionary *recordSettings = [[NSMutableDictionary alloc] initWithCapacity:10];
    NSNumber *formatObject;
    formatObject = [NSNumber numberWithInt: kAudioFormatMPEG4AAC];

    [recordSettings setObject:formatObject forKey: AVFormatIDKey];
    [recordSettings setObject:[NSNumber numberWithFloat:44100.0] forKey: AVSampleRateKey];
    [recordSettings setObject:[NSNumber numberWithInt:1] forKey:AVNumberOfChannelsKey];
    // [recordSettings setObject:[NSNumber numberWithInt:12800] forKey:AVEncoderBitRateKey];
    // [recordSettings setObject:[NSNumber numberWithInt:16] forKey:AVLinearPCMBitDepthKey];
    [recordSettings setObject:[NSNumber numberWithInt: AVAudioQualityHigh] forKey: AVEncoderAudioQualityKey];

    so when I try to convert it using ffmpeg -i 1.m4a 1.mp3 it says invalid input found while reading 1.m4a. I tried different encoding but everytime I get the same error. Any help on this would be life saving.

    Thanks.