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La conservation du net art au musée. Les stratégies à l’œuvre
26 mai 2011
Mis à jour : Juillet 2013
Langue : français
Type : Texte
Autres articles (112)
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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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Script d’installation automatique de MediaSPIP
25 avril 2011, parAfin de palier aux difficultés d’installation dues principalement aux dépendances logicielles coté serveur, un script d’installation "tout en un" en bash a été créé afin de faciliter cette étape sur un serveur doté d’une distribution Linux compatible.
Vous devez bénéficier d’un accès SSH à votre serveur et d’un compte "root" afin de l’utiliser, ce qui permettra d’installer les dépendances. Contactez votre hébergeur si vous ne disposez pas de cela.
La documentation de l’utilisation du script d’installation (...) -
Ajouter des informations spécifiques aux utilisateurs et autres modifications de comportement liées aux auteurs
12 avril 2011, parLa manière la plus simple d’ajouter des informations aux auteurs est d’installer le plugin Inscription3. Il permet également de modifier certains comportements liés aux utilisateurs (référez-vous à sa documentation pour plus d’informations).
Il est également possible d’ajouter des champs aux auteurs en installant les plugins champs extras 2 et Interface pour champs extras.
Sur d’autres sites (9817)
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FFmpeg remove silence with exact duration detected by detect silence
17 mars 2021, par davI have an audio file, that have some silences, which I am detecting with ffmpeg detectsilence and then trying to remove with removesilence, however there is some strange behavior. Specifically :


1) File's Basic info based on ffprobe show_streams


Input #0, mp3, from 'my_file.mp3':
 Metadata:
 encoder : Lavf58.64.100
 Duration: 00:00:25.22, start: 0.046042, bitrate: 32 kb/s
 Stream #0:0: Audio: mp3, 24000 Hz, mono, fltp, 32 kb/s



2) Using detectsilence


ffmpeg -i my_file.mp3 -af silencedetect=noise=-50dB:d=0.2 -f null -



I get this result


[mp3float @ 000001ee50074280] overread, skip -7 enddists: -1 -1
[silencedetect @ 000001ee5008a1c0] silence_start: 6.21417
[silencedetect @ 000001ee5008a1c0] silence_end: 6.91712 | silence_duration: 0.702958
[silencedetect @ 000001ee5008a1c0] silence_start: 16.44
[silencedetect @ 000001ee5008a1c0] silence_end: 17.1547 | silence_duration: 0.714708
[mp3float @ 000001ee50074280] overread, skip -10 enddists: -3 -3
[mp3float @ 000001ee50074280] overread, skip -5 enddists: -4 -4
[silencedetect @ 000001ee5008a1c0] silence_start: 24.4501
size=N/A time=00:00:25.17 bitrate=N/A speed=1.32e+03x
video:0kB audio:1180kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
[silencedetect @ 000001ee5008a1c0] silence_end: 25.176 | silence_duration: 0.725917



That also match the values and points based on Adobe Audition




So far all good.


3) Now, based on some calculations (which is based on application's logic on what should be the final duration of the audio) I am trying to delete the silence with "0.725917"s duration. For that, based on ffmpeg docs (https://ffmpeg.org/ffmpeg-filters.html#silencedetect)




Trim all silence encountered from beginning to end where there is more
than 1 second of silence in audio :
silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-90dB




I run this command


ffmpeg -i my_file.mp3 -af silenceremove=stop_periods=-1:stop_threshold=-50dB:stop_duration=0.72 result1.mp3



So, I am expecting that it should delete only the silence with "0.725917" duration (the last one in the above image), however it is deleting the silence that starts at 16.44s with duration of "0.714708"s. Please see the following comparison :




4) Running detectsilence on result1.mp3 with same options gives even stranger results


ffmpeg -i result1.mp3 -af silencedetect=noise=-50dB:d=0.2 -f null -



result


[mp3float @ 0000017723404280] overread, skip -5 enddists: -4 -4
[silencedetect @ 0000017723419540] silence_start: 6.21417
[silencedetect @ 0000017723419540] silence_end: 6.92462 | silence_duration: 0.710458
[mp3float @ 0000017723404280] overread, skip -7 enddists: -6 -6
[mp3float @ 0000017723404280] overread, skip -7 enddists: -2 -2
[mp3float @ 0000017723404280] overread, skip -6 enddists: -1 -1
 Last message repeated 1 times
[silencedetect @ 0000017723419540] silence_start: 23.7308
size=N/A time=00:00:24.45 bitrate=N/A speed=1.33e+03x
video:0kB audio:1146kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
[silencedetect @ 0000017723419540] silence_end: 24.456 | silence_duration: 0.725167



So, the results are :


- 

- With command to remove silences that are longer than "0.72 second", a silence that was "0.714708"s, got removed and - a silence with "0.725917"s remained as is (well, actually changed a little - as per 3rd point)
- The first silence that had started at "6.21417" and had a duration of "0.702958"s, suddenly now has a duration of "0.710458"s
- The 3rd silence that had started at "24.4501" (which now starts at 23.7308 - obviously because the 2nd silence was removed) and had a duration of "0.725917", now suddenly is "0.725167"s (this one is not a big difference, but still why even removing other silence, this silence's duration should change at all).








Accordingly the expected results are :


- 

- Only the silences that match the provided condition (stop_duration=0.72) should be removed. In this specific example only the last one, but in general any silence that matches the condition of the length - irrelevant of their positioning (start, end or in the middle)
- Other silences should remain with same exact duration they were before






FFMpeg : 4.2.4-1ubuntu0.1, Ubuntu : 20.04.2


Some attempts and results, while playing with ffmpeg options


a)


ffmpeg -i my_file.mp3 -af silenceremove=stop_periods=-1:stop_threshold=-50dB:stop_duration=0.72:detection=peak tmp1.mp3



result :
First and second silences are removed, 3rd silence's duration remains exactly the same


b)


ffmpeg -i my_file.mp3 -af silenceremove=stop_periods=-1:stop_threshold=-50dB:stop_duration=0.71 tmp_0.71.mp3



result :
First and second silences are removed, 3rd silence remains, but the duration becomes "0.72075"s


c)


ffmpeg -i my_file.mp3 -af silenceremove=stop_periods=-1:stop_threshold=-50dB:stop_duration=0.7 tmp_0.7.mp3



result :
all 3 silence are removed


d) the edge case


this command still removes the second silence (after which the first silence become exactly as in point #4 and last silence becomes "0.721375")


ffmpeg -i my_file.mp3 -af silenceremove=stop_periods=-1:stop_threshold=-50dB:stop_duration=0.72335499999 tmp_0.72335499999.mp3



but this one, again does not remove any silence :


ffmpeg -i my_file.mp3 -af silenceremove=stop_periods=-1:stop_threshold=-50dB:stop_duration=0.723355 tmp_0.723355.mp3



e) window param case 0.03


ffmpeg -i my_file.mp3 -af silenceremove=stop_periods=-1:stop_threshold=-50dB:stop_duration=0.72:window=0.03 window_0.03.mp3



does not remove any silence, but the detect silence


ffmpeg -i window_0.03.mp3 -af silencedetect=noise=-50dB:d=0.2 -f null -



gives this result (compare with silences in result1.mp3 - from point #4 )


[mp3float @ 000001c5c8824280] overread, skip -5 enddists: -4 -4
[silencedetect @ 000001c5c883a040] silence_start: 6.21417
[silencedetect @ 000001c5c883a040] silence_end: 6.92462 | silence_duration: 0.710458
[mp3float @ 000001c5c8824280] overread, skip -7 enddists: -6 -6
[mp3float @ 000001c5c8824280] overread, skip -7 enddists: -2 -2
[silencedetect @ 000001c5c883a040] silence_start: 16.4424
[silencedetect @ 000001c5c883a040] silence_end: 17.1555 | silence_duration: 0.713167
[mp3float @ 000001c5c8824280] overread, skip -6 enddists: -1 -1
 Last message repeated 1 times
[silencedetect @ 000001c5c883a040] silence_start: 24.4508
size=N/A time=00:00:25.17 bitrate=N/A speed=1.24e+03x
video:0kB audio:1180kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
[silencedetect @ 000001c5c883a040] silence_end: 25.176 | silence_duration: 0.725167



f) window case 0.01


ffmpeg -i my_file.mp3 -af silenceremove=stop_periods=-1:stop_threshold=-50dB:stop_duration=0.72:window=0.01 window_0.01.mp3



removes first and second silences, the detect silence with same params has the following result


[mp3float @ 000001ea631d4280] overread, skip -5 enddists: -4 -4
 Last message repeated 1 times
[mp3float @ 000001ea631d4280] overread, skip -7 enddists: -2 -2
[mp3float @ 000001ea631d4280] overread, skip -6 enddists: -1 -1
 Last message repeated 1 times
[silencedetect @ 000001ea631ea1c0] silence_start: 23.0108
size=N/A time=00:00:23.73 bitrate=N/A speed=1.2e+03x
video:0kB audio:1113kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
[silencedetect @ 000001ea631ea1c0] silence_end: 23.736 | silence_duration: 0.725167




Any thoughts, ideas, points are much appreciated.


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Hacking the Popcorn Hour C-200
Update : A new firmware version has been released since the publication of this article. I do not know if the procedure described below will work with the new version.
The Popcorn Hour C-200 is a Linux-based media player with impressive specifications. At its heart is a Sigma Designs SMP8643 system on chip with a 667MHz MIPS 74Kf as main CPU, several co-processors, and 512MB of DRAM attached. Gigabit Ethernet, SATA, and USB provide connectivity with the world around it. With a modest $299 on the price tag, the temptation to repurpose the unit as a low-power server or cheap development board is hard to resist. This article shows how such a conversion can be achieved.
Kernel
The PCH runs a patched Linux 2.6.22.19 kernel. A source tarball is available from the manufacturer. This contains the sources with Sigma support patches, Con Kolivas’ patch set (scheduler tweaks), and assorted unrelated changes. Properly split patches are unfortunately not available. I have created a reduced patch against vanilla 2.6.22.19 with only Sigma-specific changes, available here.
The installed kernel has a number of features disabled, notably PTY support and oprofile. We will use kexec to load a more friendly one.
As might be expected, the PCH kernel does not have kexec support enabled. It does however, by virtue of using closed-source components, support module loading. This lets us turn kexec into a module and load it. A patch for this is available here. To build the module, apply the patch to the PCH sources and build using this configuration. This will produce two modules, kexec.ko and mips_kexec.ko. No other products of this build will be needed.
The replacement kernel can be built from the PCH sources or, if one prefers, from vanilla 2.6.22.19 with the Sigma-only patch. For the latter case, this config provides a minimal starting point suitable for NFS-root.
When configuring the kernel, make sure CONFIG_TANGOX_IGNORE_CMDLINE is enabled. Otherwise the command line will be overridden by a useless one stored in flash. A good command line can be set with CONFIG_CMDLINE (under “Kernel hacking” in menuconfig) or passed from kexec.
Taking control
In order to load our kexec module, we must first gain root privileges on the PCH, and here a few features of the system are working to our advantage :
- The PCH allows mounting any NFS export to access media files stored there.
- There is an HTTP server running. As root.
- This HTTP server can be readily instructed to fetch files from an NFS mount.
- Files with a name ending in .cgi are executed. As root.
All we need do to profit from this is place the kexec modules, the kexec userspace tools, and a simple script on an NFS export. Once this is done, and the mount point configured on the PCH, a simple HTTP request will send the old kernel screaming to /dev/null, our shiny new kernel taking its place.
The rootfs
A kernel is mostly useless without a root filesystem containing tools and applications. A number of tools for cross-compiling a full system exist, each with its strengths and weaknesses. The only thing to look out for is the version of kernel headers used (usually a linux-headers package). As we will be running an old kernel, chances are the default version is too recent. Other than this, everything should be by the book.
Assembling the parts
Having gathered all the pieces, it is now time to assemble the hack. The following steps are suitable for an NFS-root system. Adaptation to a disk-based system is left as an exercise.
- Build a rootfs for MIPS 74Kf little endian. Make sure kernel headers used are no more recent than 2.6.22.x. Include a recent version of the kexec userspace tools.
- Fetch and unpack the PCH kernel sources.
- Apply the modular kexec patch.
- Using this config, build the modules and install them as usual to the rootfs. The version string must be 2.6.22.19-19-4.
- From either the same kernel sources or plain 2.6.22.19 with Sigma patches, build a vmlinux and (optionally) modules using this config. Modify the compiled-in command line to point to the correct rootfs. Set the version string to something other than in the previous step.
- Copy vmlinux to any directory in the rootfs.
- Copy kexec.sh and kexec.cgi to the same directory as vmlinux.
- Export the rootfs over NFS with full read/write permissions for the PCH.
- Power on the PCH, and update to latest firmware.
- Configure an NFS mount of the rootfs.
- Navigate to the rootfs in the PCH UI. A directory listing of bin, dev, etc. should be displayed.
- On the host system, run the kexec.sh script with the target hostname or IP address as argument.
- If all goes well, the new kernel will boot and mount the rootfs.
Serial console
A serial console is indispensable for solving boot problems. The PCH board has two UART connectors. We will use the one labeled UART0. The pinout is as follows (not standard PC pinout).
+-----------+ 2| * * * * * |10 1| * * * * * |9 -----------+ J7 UART0 /---------------------/ board edge
Pin Function 1 +5V 5 Rx 6 Tx 10 GND The signals are 3.3V so a converter, e.g. MAX202, is required for connecting this to a PC serial port. The default port settings are 115200 bps 8n1.
-
Recording voice using HTML5 and processing it with ffmpeg
22 mars 2015, par user3789242I need to use ffmpeg in my javascript/HTML5 project which allows the user to select the format he wants the audio to open with.I don’t know anything about ffmpeg and I’ve been doing lots of research I don’t know how to use it in my project. I found an example https://github.com/sopel39/audioconverter.js but the problem how can I install the ffmpeg.js which is 8 mg to m project. please if someone can help me I’ll be very thankfull
here is my full code :the javascript page :
// variables
var leftchannel = [];
var rightchannel = [];
var recorder = null;
var recording = false;
var recordingLength = 0;
var volume = null;
var audioInput = null;
var sampleRate = 44100;
var audioContext = null;
var context = null;
var outputString;
if (!navigator.getUserMedia)
navigator.getUserMedia = navigator.getUserMedia ||
navigator.webkitGetUserMedia ||
navigator.mozGetUserMedia ||
navigator.msGetUserMedia;
if (navigator.getUserMedia){
navigator.getUserMedia({audio:true}, success, function(e) {
alert('Error capturing audio.');
});
} else alert('getUserMedia not supported in this browser.');
function getVal(value)
{
// if R is pressed, we start recording
if ( value == "record"){
recording = true;
// reset the buffers for the new recording
leftchannel.length = rightchannel.length = 0;
recordingLength = 0;
document.getElementById('output').innerHTML="Recording now...";
// if S is pressed, we stop the recording and package the WAV file
} else if ( value == "stop" ){
// we stop recording
recording = false;
document.getElementById('output').innerHTML="Building wav file...";
// we flat the left and right channels down
var leftBuffer = mergeBuffers ( leftchannel, recordingLength );
var rightBuffer = mergeBuffers ( rightchannel, recordingLength );
// we interleave both channels together
var interleaved = interleave ( leftBuffer, rightBuffer );
var buffer = new ArrayBuffer(44 + interleaved.length * 2);
var view = new DataView(buffer);
// RIFF chunk descriptor
writeUTFBytes(view, 0, 'RIFF');
view.setUint32(4, 44 + interleaved.length * 2, true);
writeUTFBytes(view, 8, 'WAVE');
// FMT sub-chunk
writeUTFBytes(view, 12, 'fmt ');
view.setUint32(16, 16, true);
view.setUint16(20, 1, true);
// stereo (2 channels)
view.setUint16(22, 2, true);
view.setUint32(24, sampleRate, true);
view.setUint32(28, sampleRate * 4, true);
view.setUint16(32, 4, true);
view.setUint16(34, 16, true);
// data sub-chunk
writeUTFBytes(view, 36, 'data');
view.setUint32(40, interleaved.length * 2, true);
var lng = interleaved.length;
var index = 44;
var volume = 1;
for (var i = 0; i < lng; i++){
view.setInt16(index, interleaved[i] * (0x7FFF * volume), true);
index += 2;
}
var blob = new Blob ( [ view ], { type : 'audio/wav' } );
// let's save it locally
document.getElementById('output').innerHTML='Handing off the file now...';
var url = (window.URL || window.webkitURL).createObjectURL(blob);
var li = document.createElement('li');
var au = document.createElement('audio');
var hf = document.createElement('a');
au.controls = true;
au.src = url;
hf.href = url;
hf.download = 'audio_recording_' + new Date().getTime() + '.wav';
hf.innerHTML = hf.download;
li.appendChild(au);
li.appendChild(hf);
recordingList.appendChild(li);
}
}
function success(e){
audioContext = window.AudioContext || window.webkitAudioContext;
context = new audioContext();
volume = context.createGain();
// creates an audio node from the microphone incoming stream(source)
source = context.createMediaStreamSource(e);
// connect the stream(source) to the gain node
source.connect(volume);
var bufferSize = 2048;
recorder = context.createScriptProcessor(bufferSize, 2, 2);
//node for the visualizer
analyser = context.createAnalyser();
analyser.smoothingTimeConstant = 0.3;
analyser.fftSize = 512;
splitter = context.createChannelSplitter();
//when recording happens
recorder.onaudioprocess = function(e){
if (!recording) return;
var left = e.inputBuffer.getChannelData (0);
var right = e.inputBuffer.getChannelData (1);
leftchannel.push (new Float32Array (left));
rightchannel.push (new Float32Array (right));
recordingLength += bufferSize;
// get the average for the first channel
var array = new Uint8Array(analyser.frequencyBinCount);
analyser.getByteFrequencyData(array);
var c=document.getElementById("myCanvas");
var ctx = c.getContext("2d");
// clear the current state
ctx.clearRect(0, 0, 1000, 325);
var gradient = ctx.createLinearGradient(0,0,0,300);
gradient.addColorStop(1,'#000000');
gradient.addColorStop(0.75,'#ff0000');
gradient.addColorStop(0.25,'#ffff00');
gradient.addColorStop(0,'#ffffff');
// set the fill style
ctx.fillStyle=gradient;
drawSpectrum(array);
function drawSpectrum(array) {
for ( var i = 0; i < (array.length); i++ ){
var value = array[i];
ctx.fillRect(i*5,325-value,3,325);
}
}
}
function getAverageVolume(array) {
var values = 0;
var average;
var length = array.length;
// get all the frequency amplitudes
for (var i = 0; i < length; i++) {
values += array[i];
}
average = values / length;
return average;
}
// we connect the recorder(node to destination(speakers))
volume.connect(splitter);
splitter.connect(analyser, 0, 0);
analyser.connect(recorder);
recorder.connect(context.destination);
}
function mergeBuffers(channelBuffer, recordingLength){
var result = new Float32Array(recordingLength);
var offset = 0;
var lng = channelBuffer.length;
for (var i = 0; i < lng; i++){
var buffer = channelBuffer[i];
result.set(buffer, offset);
offset += buffer.length;
}
return result;
}
function interleave(leftChannel, rightChannel){
var length = leftChannel.length + rightChannel.length;
var result = new Float32Array(length);
var inputIndex = 0;
for (var index = 0; index < length; ){
result[index++] = leftChannel[inputIndex];
result[index++] = rightChannel[inputIndex];
inputIndex++;
}
return result;
}
function writeUTFBytes(view, offset, string){
var lng = string.length;
for (var i = 0; i < lng; i++){
view.setUint8(offset + i, string.charCodeAt(i));
}
}and here is the html code :
<code class="echappe-js"><script src="http://stackoverflow.com/feeds/tag/js/functions.js"></script>