
Recherche avancée
Médias (3)
-
GetID3 - Bloc informations de fichiers
9 avril 2013, par
Mis à jour : Mai 2013
Langue : français
Type : Image
-
GetID3 - Boutons supplémentaires
9 avril 2013, par
Mis à jour : Avril 2013
Langue : français
Type : Image
-
Collections - Formulaire de création rapide
19 février 2013, par
Mis à jour : Février 2013
Langue : français
Type : Image
Autres articles (45)
-
Supporting all media types
13 avril 2011, parUnlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)
-
Contribute to translation
13 avril 2011You can help us to improve the language used in the software interface to make MediaSPIP more accessible and user-friendly. You can also translate the interface into any language that allows it to spread to new linguistic communities.
To do this, we use the translation interface of SPIP where the all the language modules of MediaSPIP are available. Just subscribe to the mailing list and request further informantion on translation.
MediaSPIP is currently available in French and English (...) -
Librairies et binaires spécifiques au traitement vidéo et sonore
31 janvier 2010, parLes logiciels et librairies suivantes sont utilisées par SPIPmotion d’une manière ou d’une autre.
Binaires obligatoires FFMpeg : encodeur principal, permet de transcoder presque tous les types de fichiers vidéo et sonores dans les formats lisibles sur Internet. CF ce tutoriel pour son installation ; Oggz-tools : outils d’inspection de fichiers ogg ; Mediainfo : récupération d’informations depuis la plupart des formats vidéos et sonores ;
Binaires complémentaires et facultatifs flvtool2 : (...)
Sur d’autres sites (3315)
-
C# How do I set the volume of sound bytes[]
23 juillet 2016, par McLucarioIm trying to change the volume of sound bytes[] in C#. Im reading a sound file with FFMPEG and and wanna change the volume on the fly. I found some examples and but I didnt understand them.
public void SendAudio(string pathOrUrl)
{
cancelVid = false;
isPlaying = true;
mProcess = Process.Start(new ProcessStartInfo
{ // FFmpeg requireqs us to spawn a process and hook into its stdout, so we will create a Process
FileName = "ffmpeg",
Arguments = "-i " + (char)34 + pathOrUrl + (char)34 + // Here we provide a list of arguments to feed into FFmpeg. -i means the location of the file/URL it will read from
" -f s16le -ar 48000 -ac 2 pipe:1", // Next, we tell it to output 16-bit 48000Hz PCM, over 2 channels, to stdout.
UseShellExecute = false,
RedirectStandardOutput = true, // Capture the stdout of the process
Verb = "runas"
});
while (!isRunning(mProcess)) { Task.Delay(1000); }
int blockSize = 3840; // The size of bytes to read per frame; 1920 for mono
byte[] buffer = new byte[blockSize];
byte[] gainBuffer = new byte[blockSize];
int byteCount;
while (true && !cancelVid) // Loop forever, so data will always be read
{
byteCount = mProcess.StandardOutput.BaseStream // Access the underlying MemoryStream from the stdout of FFmpeg
.Read(buffer, 0, blockSize); // Read stdout into the buffer
if (byteCount == 0) // FFmpeg did not output anything
break; // Break out of the while(true) loop, since there was nothing to read.
if (cancelVid)
break;
disAudioClient.Send(buffer, 0, byteCount); // Send our data to Discord
}
disAudioClient.Wait(); // Wait for the Voice Client to finish sending data, as ffMPEG may have already finished buffering out a song, and it is unsafe to return now.
isPlaying = false;
Console.Clear();
Console.WriteLine("Done Playing!"); -
Distorted audio output from microphone input
25 août 2016, par Mohammad Abu MusaI am writing an ogg encoder with chrome native client, I managed to successfully export the files as the wave samples come in put I have two issues
- Timing in the file
- The voice is distorted, I hear noises instead of the voice collected from the microphone
Here how the code works, first I collect the microphone samples and process it as the following
const char* data = static_cast<const>(buffer.GetDataBuffer());
audio_buffer_size = buffer.GetDataBufferSize();
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();
}
}
memcpy(samples_.data(), data,
sample_count_ * channel_count_ * sizeof(char));
</const>The result is a
vector
called samples_ of typechar
for the other information they are as follows- Samples 480
- Channels 1
- Sample Rate 48000
- Time Interval is 15
On the function
ScheduleNextTimer
I make the encoding and the storage of the packets toogg
file. I used the standard code provided in the encoder example but I do not think I am doing write since I am not importing from a wave file, I am rather collecting raw bytes so I do not think the loops I am using are correct.
Attempt #1for (int i = 0; i < sampread; i++) {
for (int j = 0; j < audioCh; j++) {
vorbis_buffer[j][i] = ((samples_[count + 1] << 8)
| (0x00ff & (int) samples_[count])) / 32768.f;
count += 2;
}
}Attempt #2
for(int i=0;i4;i++){
vorbis_buffer[0][i]=((samples_[i*4+1]<<8)|
(0x00ff&(int)samples_[i*4]))/32768.f;
vorbis_buffer[1][i]=((samples_[i*4+3]<<8)|
(0x00ff&(int)samples_[i*4+2]))/32768.f;
}Both of these attempts did not work. My question is should I keep using the
vector
or should I change it to something else. what are these values in theloop
I do not get them or what they represent and how does the timing work.Here is what I get from
ffprob
Input #0, ogg, from 'file.ogg':
Duration: 00:01:21.00, start: 0.000000, bitrate: 118 kb/s
Stream #0:0: Audio: vorbis, 44100 Hz, stereo, fltp, 80 kb/s -
Matomo vs WP-Statistics – which web analytics plugin suits you best ?
2 avril 2020, par Joselyn Khor — Analytics Tips, Plugins