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Rennes Emotion Map 2010-11
19 octobre 2011, par
Mis à jour : Juillet 2013
Langue : français
Type : Texte
Autres articles (37)
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Les autorisations surchargées par les plugins
27 avril 2010, parMediaspip core
autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs -
List of compatible distributions
26 avril 2011, parThe table below is the list of Linux distributions compatible with the automated installation script of MediaSPIP. Distribution nameVersion nameVersion number Debian Squeeze 6.x.x Debian Weezy 7.x.x Debian Jessie 8.x.x Ubuntu The Precise Pangolin 12.04 LTS Ubuntu The Trusty Tahr 14.04
If you want to help us improve this list, you can provide us access to a machine whose distribution is not mentioned above or send the necessary fixes to add (...) -
Sélection de projets utilisant MediaSPIP
29 avril 2011, parLes exemples cités ci-dessous sont des éléments représentatifs d’usages spécifiques de MediaSPIP pour certains projets.
Vous pensez avoir un site "remarquable" réalisé avec MediaSPIP ? Faites le nous savoir ici.
Ferme MediaSPIP @ Infini
L’Association Infini développe des activités d’accueil, de point d’accès internet, de formation, de conduite de projets innovants dans le domaine des Technologies de l’Information et de la Communication, et l’hébergement de sites. Elle joue en la matière un rôle unique (...)
Sur d’autres sites (8829)
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Starnge behavior of ffmpeg while opening rtsp with .sdp
7 avril 2014, par Pawel RutkaI have problem with reading rtsp stream(.sdp) over ffmpeg - is here anyone who is able to help me or have something incommon with this kind of troubles ?
url stream :
rtsp://192.168.1.7/moja.sdp
Here is a log with -loglevel debug , why is there 127.0.0.1 ?
[rtsp @ 0x9fef0a0] SDP:
v=0
o=- 1 1 IN IP4 127.0.0.1
s=IPCam
c=IN IP4 0.0.0.0
t=0 0
a=type:broadcast
m=video 0 RTP/AVP 96
a=rtpmap:96 MP4V-ES/90000
a=fmtp:96 profile-level-id=1;config=000001B001000001B58913000001000000012000C488800F514043C1463F
a=control:track0
m=audio 0 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=control:track1
[rtsp @ 0x9fef0a0] video codec set to: mpeg4
[rtsp @ 0x9fef0a0] audio codec set to: pcm_alaw
[rtsp @ 0x9fef0a0] audio samplerate set to: 8000
[rtsp @ 0x9fef0a0] audio channels set to: 1
[rtsp @ 0x9fef0a0] hello state=0
[rtsp @ 0x9fef0a0] UDP timeout, retrying with TCP
[rtsp @ 0x9fef0a0] method PAUSE failed: 501 Not Implemented -
Why is the audio recording on Chrome missing duration
16 août 2019, par Ivan SedelkinI’m sending some user created audio to a server which later passes it to googles speech to text api for transcription. Everything works perfectly on firefox but when I try it on Chrome it doesn’t work. I then installed FFmpeg to check if the metadata somehow changed for the file on chrome and noticed that the duration on the file is tagged "N/A". This does not happen on firefox.
This is the audio recorded from Chrome
Input #0, matroska,webm, from 'PATH_TO_FILE':
Metadata:
encoder : Chrome
Duration: N/A, start: 0.000000, bitrate: N/A
Stream #0:0(eng): Audio: opus, 48000 Hz, mono, fltp (default)This is the audio recorded from Firefox
Input #0, ogg, from 'PATH_TO_FILE':
Duration: 00:00:01.26, start: 0.000000, bitrate: 53 kb/s
Stream #0:0: Audio: opus, 48000 Hz, mono, fltp
Metadata:
ENCODER : Mozilla68.0The audio itself is recorded using the mediarecorder-api where the blob is later converted to base64-url and sent to my server.
This is the code that I use to record the audio :
navigator.mediaDevices
.getUserMedia(
// constraints - only audio needed for this app
{
audio: true
}
)
// Success callback
.then(function(stream) {
console.log(navigator.mediaDevices.getSupportedConstraints());
var mediaRecorder = new MediaRecorder(stream, { sampleRate: 44100 });
var chunks = [];
$(".rec-button")
.mousedown(function() {
console.log("rec start");
$(".rec-button i").addClass("recStart");
mediaRecorder.start();
console.log(mediaRecorder.state);
console.log("recorder started");
})
.mouseup(function() {
console.log("rec end");
$(".rec-button i").removeClass("recStart");
mediaRecorder.stop();
mediaRecorder.ondataavailable = function(e) {
chunks.push(e.data);
var blob = new Blob(chunks, { type: "audio/ogg; codecs=opus" });
var player = document.getElementById("player");
player.src = URL.createObjectURL(blob);
chunks = [];
var reader = new window.FileReader();
reader.readAsDataURL(blob);
reader.onloadend = function() {
var base64 = reader.result;
var audioArr = {
audio: base64
};
$.ajax({
url: "http://localhost:4242/api/1.0/post",
type: "POST",
contentType: "application/json",
dataType: "json",
data: JSON.stringify(audioArr),
success: function(response) {
console.log(response);
},
error: function(err) {
console.log(err);
}
});
};
console.log(mediaRecorder.state);
console.log("recorder stopped");
};
});
})
// Error callback
.catch(function(err) {
console.log("The following getUserMedia error occured: " + err);
});My goal is that the audio file recorded from chrome has a duration so that the google api can transcribe it. If you guys have any way of overcoming this problem I would be grateful
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movenc : take packet dts shifting into mention in check_pkt
30 septembre 2017, par Jan Ekströmmovenc : take packet dts shifting into mention in check_pkt
This FFmpeg-specific "fuzzer fix" was never perfect, but now it
stopped encoding of actual content with a big enough DTS shift.
This returns the function to its original state of results
before negative CTS offsets were added.I remember dealing with this function before, but somehow had
forgotten about it during VDD. The test cases not tripping this
over also didn't help.