
Recherche avancée
Médias (1)
-
1 000 000 (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
Autres articles (100)
-
MediaSPIP 0.1 Beta version
25 avril 2011, parMediaSPIP 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 (...) -
Participer à sa traduction
10 avril 2011Vous pouvez nous aider à améliorer les locutions utilisées dans le logiciel ou à traduire celui-ci dans n’importe qu’elle nouvelle langue permettant sa diffusion à de nouvelles communautés linguistiques.
Pour ce faire, on utilise l’interface de traduction de SPIP où l’ensemble des modules de langue de MediaSPIP sont à disposition. ll vous suffit de vous inscrire sur la liste de discussion des traducteurs pour demander plus d’informations.
Actuellement MediaSPIP n’est disponible qu’en français et (...) -
MediaSPIP v0.2
21 juin 2013, parMediaSPIP 0.2 est la première version de MediaSPIP stable.
Sa date de sortie officielle est le 21 juin 2013 et est annoncée ici.
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Comme pour la version précédente, 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 (...)
Sur d’autres sites (11977)
-
Graph-based video processing for .NET
23 octobre 2016, par BorvDoes anyone know a good object-oriented library (preferably high-level, like C# or Java) for working with video and audio streams ?
I wrote an app which fiddles with video and audio streams, feeds and such. The original task was simple :
- grab an RTSP feed
- display original feed(s) on the display
- convert it to a series of h264 ts files
- extract audio into separate MP3 files
- upload videos and audio to the web site (preferably in real time, few minute delay is acceptable)
As you may have already guessed it is about recording events (e.g. lectures) and publishing them on the web.
To pull this out I needed some graph-based non-linear editing for media. Two weeks in, I tried ffmpeg, vlc and WMF. The only library I got to work is ffmpeg, and that comes with lots of "however". WMF required a lot of coding (and I abandoned this path), vlc looked great on paper, but I stumbled across some bugs with input splitting I could not get around (e.g. transcode:es combination flat out refused to work).
So, the question. What are good non-linear editing libraries besides ffmpeg, vlc and wmf/directshow that allow for building video processing graphs with sources, sinks and filters ? Or perhaps good bindings over ffmpeg and vlc allowing to build such graphs ?
-
Wave bytes to buffer
24 août 2016, par Mohammad Abu MusaI am encoding wav input from microphone which comes in four bytes format to ogg format. I think I have a problem shifting the bytes to the correct format here is the code I am using
To explain more I get the audio frames from Google Chrome where I get
data
asconst8
andchannels
, andsamples
.data
field is always in 4 bytes format.I copy the data to a vector of type
int16_t
then I loopuninterleave samples
which I think I am doing wrong. my question is how can I make sure the data is formatted correctly forogg
encoder to handle them correctly ?void EncoderInstance::OnGetBuffer(int32_t result, pp::AudioBuffer buffer) {
if (result != PP_OK)
return;
assert(buffer.GetSampleSize() == PP_AUDIOBUFFER_SAMPLESIZE_16_BITS);
const char* data = static_cast<const>(buffer.GetDataBuffer());
uint32_t channels = buffer.GetNumberOfChannels();
uint32_t samples = buffer.GetNumberOfSamples() / channels;
if (channel_count_ != channels || sample_count_ != samples) {
channel_count_ = channels;
sample_count_ = samples;
samples_.resize(sample_count_ * channel_count_);
// Try (+ 5) to ensure that we pick up a new set of samples between each
// timer-generated repaint.
timer_interval_ = (sample_count_ * 1000) / buffer.GetSampleRate() + 5;
// Start the timer for the first buffer.
if (first_buffer_) {
first_buffer_ = false;
ScheduleNextTimer();
}
}
if(is_audio_recording && is_audio_header_written_)
{
memcpy(samples_.data(), data,
sample_count_ * channel_count_ * sizeof(int16_t));
float **buffer=vorbis_analysis_buffer(&vd,samples);
/* uninterleave samples */
for(i=0;i4;i++)
{
buffer[0][i]=((samples_.at(i*4+1)<<8)|
(0x00ff&(int16_t)samples_.at(i*4)))/32768.f;
buffer[1][i]=((samples_.at(i*4+3)<<8)|
(0x00ff&(int16_t)samples_.at(i*4+2)))/32768.f;
}
vorbis_analysis_wrote(&vd,i);
while(vorbis_analysis_blockout(&vd,&vb)==1){
/* analysis, assume we want to use bitrate management */
vorbis_analysis(&vb,NULL);
vorbis_bitrate_addblock(&vb);
while(vorbis_bitrate_flushpacket(&vd,&op)){
/* weld the packet into the bitstream */
ogg_stream_packetin(&os,&op);
/* write out pages (if any) */
while(!eos){
int result=ogg_stream_pageout(&os,&og);
if(result==0)break;
glb_app_thread.message_loop().PostWork(callback_factory_.NewCallback(&EncoderInstance::writeAudioHeader));
if(ogg_page_eos(&og))eos=1;
}
}
}
}
audio_track_.RecycleBuffer(buffer);
audio_track_.GetBuffer(callback_factory_.NewCallbackWithOutput(
&EncoderInstance::OnGetBuffer));
}
</const> -
10 Matomo Features You Possibly Didn’t Know About
28 octobre 2022, par Erin