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Amélioration de la version de base
13 septembre 2013Jolie sélection multiple
Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
Il suffit pour cela d’activer le plugin Chosen (Configuration générale du site > Gestion des plugins), puis de configurer le plugin (Les squelettes > Chosen) en activant l’utilisation de Chosen dans le site public et en spécifiant les éléments de formulaires à améliorer, par exemple select[multiple] pour les listes à sélection multiple (...) -
Emballe médias : à quoi cela sert ?
4 février 2011, parCe 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" ; -
Le plugin : Gestion de la mutualisation
2 mars 2010, parLe plugin de Gestion de mutualisation permet de gérer les différents canaux de mediaspip depuis un site maître. Il a pour but de fournir une solution pure SPIP afin de remplacer cette ancienne solution.
Installation basique
On installe les fichiers de SPIP sur le serveur.
On ajoute ensuite le plugin "mutualisation" à la racine du site comme décrit ici.
On customise le fichier mes_options.php central comme on le souhaite. Voilà pour l’exemple celui de la plateforme mediaspip.net :
< ?php (...)
Sur d’autres sites (7612)
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avfilter/vf_xmedian : allow to control eof handling
30 décembre 2020, par Paul B Mahol -
Increase/Decrease audio volume using FFmpeg
9 juin 2016, par williamtroupI’m am currently using C# invokes to call the FFmpeg APIs to handle video and audio. I have the following code in place to extract the audio from a video and write it to a file.
while (ffmpeg.av_read_frame(formatContext, &packet) >= 0)
{
if (packet.stream_index == streamIndex)
{
while (packet.size > 0)
{
int frameDecoded;
int frameDecodedResult = ffmpeg.avcodec_decode_audio4(codecContext, frame, &frameDecoded, packet);
if (frameDecoded > 0 && frameDecodedResult >= 0)
{
//writeAudio.WriteFrame(frame);
packet.data += totalBytesDecoded;
packet.size -= totalBytesDecoded;
}
}
frameIndex++;
}
Avcodec.av_free_packet(&packet);
}This is all working correctly. I’m currently using the FFmpeg.AutoGen project for the API access.
I want to be able to increase/decrease the volume of the audio before its written to the file, but I cannot seem to find a command or any help with this. Does it have to be done manually ?
Update 1 :
After receiving some help, this is the class layout I have :
public unsafe class FilterVolume
{
#region Private Member Variables
private AVFilterGraph* m_filterGraph = null;
private AVFilterContext* m_aBufferSourceFilterContext = null;
private AVFilterContext* m_aBufferSinkFilterContext = null;
#endregion
#region Private Constant Member Variables
private const int EAGAIN = 11;
#endregion
public FilterVolume(AVCodecContext* codecContext, AVStream* stream, float volume)
{
CodecContext = codecContext;
Stream = stream;
Volume = volume;
Initialise();
}
public AVFrame* Adjust(AVFrame* frame)
{
AVFrame* returnFilteredFrame = ffmpeg.av_frame_alloc();
if (m_aBufferSourceFilterContext != null && m_aBufferSinkFilterContext != null)
{
int bufferSourceAddFrameResult = ffmpeg.av_buffersrc_add_frame(m_aBufferSourceFilterContext, frame);
if (bufferSourceAddFrameResult < 0)
{
}
int bufferSinkGetFrameResult = ffmpeg.av_buffersink_get_frame(m_aBufferSinkFilterContext, returnFilteredFrame);
if (bufferSinkGetFrameResult < 0 && bufferSinkGetFrameResult != -EAGAIN)
{
}
}
return returnFilteredFrame;
}
public void Dispose()
{
Cleanup(m_filterGraph);
}
#region Private Properties
private AVCodecContext* CodecContext { get; set; }
private AVStream* Stream { get; set; }
private float Volume { get; set; }
#endregion
#region Private Setup Helper Functions
private void Initialise()
{
m_filterGraph = GetAllocatedFilterGraph();
string aBufferFilterArguments = string.Format("sample_fmt={0}:channel_layout={1}:sample_rate={2}:time_base={3}/{4}",
(int)CodecContext->sample_fmt,
CodecContext->channel_layout,
CodecContext->sample_rate,
Stream->time_base.num,
Stream->time_base.den);
AVFilterContext* aBufferSourceFilterContext = CreateFilter("abuffer", m_filterGraph, aBufferFilterArguments);
AVFilterContext* volumeFilterContext = CreateFilter("volume", m_filterGraph, string.Format("volume={0}", Volume));
AVFilterContext* aBufferSinkFilterContext = CreateFilter("abuffersink", m_filterGraph);
LinkFilter(aBufferSourceFilterContext, volumeFilterContext);
LinkFilter(volumeFilterContext, aBufferSinkFilterContext);
SetFilterGraphConfiguration(m_filterGraph, null);
m_aBufferSourceFilterContext = aBufferSourceFilterContext;
m_aBufferSinkFilterContext = aBufferSinkFilterContext;
}
#endregion
#region Private Cleanup Helper Functions
private static void Cleanup(AVFilterGraph* filterGraph)
{
if (filterGraph != null)
{
ffmpeg.avfilter_graph_free(&filterGraph);
}
}
#endregion
#region Provate Helpers
private AVFilterGraph* GetAllocatedFilterGraph()
{
AVFilterGraph* filterGraph = ffmpeg.avfilter_graph_alloc();
if (filterGraph == null)
{
}
return filterGraph;
}
private AVFilter* GetFilterByName(string name)
{
AVFilter* filter = ffmpeg.avfilter_get_by_name(name);
if (filter == null)
{
}
return filter;
}
private void SetFilterGraphConfiguration(AVFilterGraph* filterGraph, void* logContext)
{
int filterGraphConfigResult = ffmpeg.avfilter_graph_config(filterGraph, logContext);
if (filterGraphConfigResult < 0)
{
}
}
private AVFilterContext* CreateFilter(string filterName, AVFilterGraph* filterGraph, string filterArguments = null)
{
AVFilter* filter = GetFilterByName(filterName);
AVFilterContext* filterContext;
int aBufferFilterCreateResult = ffmpeg.avfilter_graph_create_filter(&filterContext, filter, filterName, filterArguments, null, filterGraph);
if (aBufferFilterCreateResult < 0)
{
}
return filterContext;
}
private void LinkFilter(AVFilterContext* source, AVFilterContext* destination)
{
int filterLinkResult = ffmpeg.avfilter_link(source, 0, destination, 0);
if (filterLinkResult < 0)
{
}
}
#endregion
}The Adjust() function is called after a frame is decoded. I’m currently getting a -22 error when av_buffersrc_add_frame() is called. This indicates that a parameter is invalid, but after debugging, I cannot see anything that would be causing this.
This is how the code is called :
while (ffmpeg.av_read_frame(formatContext, &packet) >= 0)
{
if (packet.stream_index == streamIndex)
{
while (packet.size > 0)
{
int frameDecoded;
int frameDecodedResult = ffmpeg.avcodec_decode_audio4(codecContext, frame, &frameDecoded, packet);
if (frameDecoded > 0 && frameDecodedResult >= 0)
{
AVFrame* filteredFrame = m_filterVolume.Adjust(frame);
//writeAudio.WriteFrame(filteredFrame);
packet.data += totalBytesDecoded;
packet.size -= totalBytesDecoded;
}
}
frameIndex++;
}
Avcodec.av_free_packet(&packet);
}Update 2 :
Cracked it. The "channel_layout" option in the filter argument string is supposed to be a hexadecimal. This is what the string formatting should look like :
string aBufferFilterArguments = string.Format("sample_fmt={0}:channel_layout=0x{1}:sample_rate={2}:time_base={3}/{4}",
(int)CodecContext->sample_fmt,
CodecContext->channel_layout,
CodecContext->sample_rate,
Stream->time_base.num,
Stream->time_base.den); -
how can i control Nodejs server app by internet
22 décembre 2020, par Khales NaimI have deploy a node js app on a windows server that decode my ip cam by ffmpeg and jsmpeg that working very well .


On client side i can call the live cam rtsp stream by websocket on my phone .


What i want to do is to control app.js from my phone to stop or restart the app.js


any help is welcom


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