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Carte de Schillerkiez
13 mai 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Texte
Autres articles (107)
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Diogene : création de masques spécifiques de formulaires d’édition de contenus
26 octobre 2010, parDiogene est un des plugins ? SPIP activé par défaut (extension) lors de l’initialisation de MediaSPIP.
A quoi sert ce plugin
Création de masques de formulaires
Le plugin Diogène permet de créer des masques de formulaires spécifiques par secteur sur les trois objets spécifiques SPIP que sont : les articles ; les rubriques ; les sites
Il permet ainsi de définir en fonction d’un secteur particulier, un masque de formulaire par objet, ajoutant ou enlevant ainsi des champs afin de rendre le formulaire (...) -
MediaSPIP version 0.1 Beta
16 avril 2011, parMediaSPIP 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 (...) -
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 ;
Sur d’autres sites (15105)
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wrong play audio samples
8 août 2014, par Ivan LisovichI have a problem with a ffmpeg and NAudio libs.
I worked with the old ffmpeg library and there the audio plays correctly.
read video in manage c++// Read frames and save to list audio frames
while(av_read_frame(pFormatCtx, &packet) >= 0)
{
if(packet.stream_index == videoStream)
{
// reade image
}
else if(packet.stream_index == audioStream)
{
int b = av_dup_packet(&packet);
if(b >= 0) {
int audio_pkt_size = packet.size;
libffmpeg::uint8_t* audio_pkt_data = packet.data;
while(audio_pkt_size > 0)
{
int got_frame = 0;
int len1 = libffmpeg::avcodec_decode_audio4(aCodecCtx, &frame, &got_frame, &packet);
if(len1 < 0)
{
/* if error, skip frame */
audio_pkt_size = 0;
break;
}
audio_pkt_data += len1;
audio_pkt_size -= len1;
if (got_frame)
{
int data_size = libffmpeg::av_samples_get_buffer_size ( NULL, aCodecCtx->channels, frame.nb_samples, aCodecCtx->sample_fmt, 1 );
array<byte>^ managedBuf = gcnew array<byte>(data_size);
System::IntPtr iptr = System::IntPtr( frame.data[0] );
System::Runtime::InteropServices::Marshal::Copy( iptr, managedBuf, 0, data_size );
audioData->Add(managedBuf);
}
}
}
}
// Free the packet that was allocated by av_read_frame
libffmpeg::av_free_packet(&packet);
}
</byte></byte>I return audioData to c# code and play in NAudio library
play in c#var recordingFormat = new WaveFormat(reader.SampleRate, 16, reader.Channels);
var waveProvider = new BufferedWaveProvider(recordingFormat) { DiscardOnBufferOverflow = true, BufferDuration = TimeSpan.FromMilliseconds(10000) };
var waveOut = new DirectSoundOut();
waveOut.Init(waveProvider);
waveOut.Play();
foreach (byte[] data in audioData)
{
waveProvider.AddSamples(data, 0, data.Length);
}but audio not playing.
what am I doing wrong ? -
wrong play audio sasamples
7 août 2014, par Ivan LisovichI have a problem with a ffmpeg and NAudio libs.
I worked with the old ffmpeg library and there the audio plays correctly.
read video in manage c++// Read frames and save to list audio frames
while(av_read_frame(pFormatCtx, &packet) >= 0)
{
if(packet.stream_index == videoStream)
{
// reade image
}
else if(packet.stream_index == audioStream)
{
int b = av_dup_packet(&packet);
if(b >= 0) {
int audio_pkt_size = packet.size;
libffmpeg::uint8_t* audio_pkt_data = packet.data;
while(audio_pkt_size > 0)
{
int got_frame = 0;
int len1 = libffmpeg::avcodec_decode_audio4(aCodecCtx, &frame, &got_frame, &packet);
if(len1 < 0)
{
/* if error, skip frame */
audio_pkt_size = 0;
break;
}
audio_pkt_data += len1;
audio_pkt_size -= len1;
if (got_frame)
{
int data_size = libffmpeg::av_samples_get_buffer_size ( NULL, aCodecCtx->channels, frame.nb_samples, aCodecCtx->sample_fmt, 1 );
array<byte>^ managedBuf = gcnew array<byte>(data_size);
System::IntPtr iptr = System::IntPtr( frame.data[0] );
System::Runtime::InteropServices::Marshal::Copy( iptr, managedBuf, 0, data_size );
audioData->Add(managedBuf);
}
}
}
}
// Free the packet that was allocated by av_read_frame
libffmpeg::av_free_packet(&packet);
}
</byte></byte>I return audioData to c# code and play in NAudio library
play in c#var recordingFormat = new WaveFormat(reader.SampleRate, 16, reader.Channels);
var waveProvider = new BufferedWaveProvider(recordingFormat) { DiscardOnBufferOverflow = true, BufferDuration = TimeSpan.FromMilliseconds(10000) };
var waveOut = new DirectSoundOut();
waveOut.Init(waveProvider);
waveOut.Play();
foreach (byte[] data in audioData)
{
waveProvider.AddSamples(data, 0, data.Length);
}but audio not playing.
what am I doing wrong ? -
Yes or no, will ffmpeg api do hardware decoding on iOS ?
15 janvier 2019, par FattieThere seems to be conflicting information on this.
https://trac.ffmpeg.org/wiki/HWAccelIntro
notice the first diagram, it firmly marks iOS as “Y” on VideoToolbox
however in the comments down the bottom it says
VideoToolbox. VideoToolbox, only supported on macOS. H.264 decoding is available in FFmpeg/libavcodec.
And in the confusing second diagram it says "Standalone" is not done for VideoToolbox.
We have found that using ffmpeg compiled in to iOS .... it seems to not use hardware decoding, which is really a pain.
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With
avcodec_get_hw_config()
we getAV_PIX_FMT_VIDEOTOOLBOX
,AV_HWDEVICE_TYPE_VIDEOTOOLBOX
which is seemingly correct. -
But usage and framerates clearly shows everything is being done in CPU. The code is in
ff_hevc_hls_residual_coding
all the time. (That’s fffmpeg’s software decoder.) -
This very diff very long git.videolan.org URL here seems to suggest again it should all be working.
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Have tried every iPhone etc. of course
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