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The pirate bay depuis la Belgique
1er avril 2013, par
Mis à jour : Avril 2013
Langue : français
Type : Image
Autres articles (25)
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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 (...)
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ANNEXE : Les plugins utilisés spécifiquement pour la ferme
5 mars 2010, parLe site central/maître de la ferme a besoin d’utiliser plusieurs plugins supplémentaires vis à vis des canaux pour son bon fonctionnement. le plugin Gestion de la mutualisation ; le plugin inscription3 pour gérer les inscriptions et les demandes de création d’instance de mutualisation dès l’inscription des utilisateurs ; le plugin verifier qui fournit une API de vérification des champs (utilisé par inscription3) ; le plugin champs extras v2 nécessité par inscription3 (...)
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Encoding and processing into web-friendly formats
13 avril 2011, parMediaSPIP automatically converts uploaded files to internet-compatible formats.
Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
All uploaded files are stored online in their original format, so you can (...)
Sur d’autres sites (3983)
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aacenc : use the decoder’s lcg PRNG
8 octobre 2016, par Rostislav Pehlivanovaacenc : use the decoder’s lcg PRNG
Using lfg was an overkill in this case where the random numbers
were only used for encoder descisions. Should increase result
uniformity between different FPUs and gives a slight speedup.Signed-off-by : Rostislav Pehlivanov <atomnuker@gmail.com>
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Using ffmpeg libraries to decode wav audio as PCM samples and display waveform
26 septembre 2016, par Lorenzo MonniI’m using the ffmpeg libraries to process audio files.
I need to decode .wav audio files to make some operations having their samples in an understandable format, i.e. decimal numbers comprised between [-1,1] as a normal audio waveform.
I have written the code for decoding and it’s apparently working well, but when I see the decoded samples it seems something in the sample numbers translation went bad. I paste here only the part of code where I translate the samples from the audio frame in PCM 16 bits :
while(av_read_frame(pFormatCtx, &apkt)>=0) {
if(apkt.stream_index==audioStream->index)
{
// Try to decode the packet into a frame
int frameFinished = 0;
avcodec_decode_audio4(aCodecCtx, aFrame, &frameFinished, &apkt);
int data_size = av_samples_get_buffer_size(&plane_size, aCodecCtx->channels,
aFrame->nb_samples,
aCodecCtx->sample_fmt, 1);
// Some frames rely on multiple packets, so we have to make sure the frame is finished before
// we can use it
if (frameFinished)
{
for(int a=0;a < plane_size/sizeof(int16_t);a++)
{
fprintf(fd,"%d\n",(int16_t*)aFrame->data[a]);
}
}
}
av_free_packet(&apkt);
// Free the packet that was allocated by av_read_frame
}Additional information and issues :
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the sample_fmt in my allocated AVCodecContext is "AV_SAMPLE_FMT_S16" so the samples numbers should be 16bit signed binaries, I guess if translated in decimal format numbers comprised between -32768 and 32767 (I don’t remember how the problem of disparity between positives and negatives is solved). However when I decode them in int16_t I see much higher numbers that seem to fall in the 32bit signed format (but the file is in 16bit anyway). E.g., the max of my decoded audio (after int16_t translation) is 2044951012 ;
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My .wav file has two channels, but I can’t access both two, if I use extended_data of the audio frame struct pointing to the second channel (index 1) the execution returns a segmentation fault. The same happens with the data pointer. I’m able to recover only one channel, from data[0].
Here is how my audio file decoded with the aforementioned code and saved in a txt looks like :
Here is how the trend of the signal should look like :
If I play my decoded signal the sound shows some similarities with the original audio file, but with a lot of destructing artifacts in it.
Final remarks : ffmpeg documentation and past questions of Stackoverflow are not working well to solve this problem.
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Is Google Analytics Accurate ? 6 Important Caveats
8 novembre 2022, par Erin