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Autres articles (94)
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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 ;
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Des sites réalisés avec MediaSPIP
2 mai 2011, parCette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page. -
Soumettre améliorations et plugins supplémentaires
10 avril 2011Si vous avez développé une nouvelle extension permettant d’ajouter une ou plusieurs fonctionnalités utiles à MediaSPIP, faites le nous savoir et son intégration dans la distribution officielle sera envisagée.
Vous pouvez utiliser la liste de discussion de développement afin de le faire savoir ou demander de l’aide quant à la réalisation de ce plugin. MediaSPIP étant basé sur SPIP, il est également possible d’utiliser le liste de discussion SPIP-zone de SPIP pour (...)
Sur d’autres sites (15664)
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how minimize an image using @ffmpeg/ffmpeg then send it to firebase [closed]
4 juillet 2023, par Yassin Samirhow minimize an image using @ffmpeg/ffmpeg then send it to firebase storage with the original image


I couldn't find any solution to problem except firebase cloud function and my project is free and open source. I need like a function give it the image blob it returns to me the image minimized in a blob or an api


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libswresample : swr_convert() not producing enough samples
20 septembre 2016, par TsherrI’m trying to use ffmpeg/libswresample to resample streaming audio in my c++ application. Changing the sample width works well and the result sounds as one would expect ; however, when changing the sample rate the result is somewhat crackly. I am unsure if it is due to incorrect usage of the libswresample library, or if I’m misunderstanding the resampling theory.
Here is my resampling process, simplified for demonstration’s sake :
//Externally supplied data
const uint8_t* in_samples //contains the audio data to be resampled
int in_num_samples = 256
//Set up resampling context
SwrContext *swr = swr_alloc();
av_opt_set_channel_layout(swr, "in_channel_layout", AV_CH_LAYOUT_STEREO, 0);
av_opt_set_channel_layout(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
av_opt_set_int(swr, "in_sample_rate", 44100, 0);
av_opt_set_int(swr, "out_sample_rate", 22050, 0);
av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLT, 0);
av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_FLT, 0);
swr_init(swr);
//Perform the resampe
uint8_t* out_samples;
int out_num_samples = av_rescale_rnd(swr_get_delay(swr, in_samplerate) + in_num_samples, out_samplerate, in_samplerate, AV_ROUND_UP);
av_samples_alloc(&out_samples, NULL, out_num_channels, out_num_samples, AV_SAMPLE_FMT_FLT, 0);
out_num_samples = swr_convert(swr, &out_samples, out_num_samples, &in_samples, in_num_samples);
av_freep(&out_samples);
swr_free(&swr);I suspect that the reason the resampled audio does not sound right is because
swr_convert()
returns 112, where I expect it to return 128 (the number of samples of the resampled audio) :
Downsampling 256 samples from a samplerate of 44100 to a samplerate of 22050 should yield 128 samples, yetswr_convert()
is producing 112 samples. When expressed in terms of audio duration this is also puzzling. 256 samples at 44100 = 5.8 ms, but 112 samples at 22050 = 5.07 ms. Shouldn’t the downsampling process not alter the duration of the resampled audio ?I have also stepped through an example provided with ffmpeg, in which swr_convert() also returns a smaller number than I would expect. So, I suspect that the problem is not due to a bug in libswresample but rather my own lack of understanding.
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FFmpeg AVERROR(EAGAIN) error when call avcodec receive for h264
7 mai 2019, par KsilonI’m working with ffmpeg 4.1 and I’m showing live streams of multiple cameras, h264 and h265.
My program collects packets of the same frame and then calls decodeVideo function. Actually it sends all packets of a frame at once.
Program works well if there is no missing packets. When I remove packet in random I-frames, both h264 and h265 streams work as expected (jumps some seconds but continues streaming).
When I remove packet in random P-frame from h265 streams, avcodec_send_packet function gives AVERROR_INVALIDDATA and streams continue.
However when I remove packet in random P-frame from h264 streams, avcodec_send_packet function gives 0. Then avcodec_receive_frame function gives AVERROR(EAGAIN) continuously and streams freeze.
void decodeVideo(array^ data, int length, AvFrame^ finishedFrame)
{
AVPacket* videoPacket = new AVPacket();
av_init_packet(videoPacket);
pin_ptr<unsigned char="char"> dataPtr = &data[0];
videoPacket->data = dataPtr;
videoPacket->size = length;
int retVal = avcodec_send_packet((AVCodecContext*)context, videoPacket);
if(retVal < 0)
{
if (retVal == AVERROR_EOF)
Utility::Log->ErrorFormat("avcodec_send_packet() return value is AVERROR_EOF.");
else if( retVal == AVERROR_INVALIDDATA)
Utility::Log->ErrorFormat("avcodec_send_packet() INVALID DATA!");
else
Utility::Log->ErrorFormat("avcodec_send_packet() return value is negative:{0}",retVal);
}
else
{
int receive_frame = avcodec_receive_frame((AVCodecContext*)context, (AVFrame*)finishedFrame);
if (receive_frame == AVERROR(EAGAIN))
Utility::Log->ErrorFormat("avcodec_receive_frame() returns AVERROR(EAGAIN)");
else if(receive_frame == AVERROR_EOF)
Utility::Log->ErrorFormat("avcodec_receive_frame() returns AVERROR(AVERROR_EOF)");
else
Utility::Log->ErrorFormat("avcodec_receive_frame() return value is negative:{0}",receive_frame);
}
av_packet_unref(videoPacket);
delete videoPacket;
}
</unsigned>EDIT
When I add
avcodec_flush_buffers
like shown, my problem is temporarily solved. However it freeze again after a while.if(receive_frame == AVERROR(EAGAIN))
{
Utility::Log->ErrorFormat("avcodec_receive_frame() returns AVERROR(EAGAIN)");
avcodec_flush_buffers((AVCodecContext*)context);
}Tested with ffmpeg version 4.1.1 same results.
Find an ffmpeg version like 2.5 decode function is different but there is no problem when i remove packets. However I’m working with h265 streams too.
EDIT2
AVCodecID id = AVCodecID::AV_CODEC_ID_H264;
AVCodec* dec = avcodec_find_decoder(id);
AVCodecContext* decContext = avcodec_alloc_context3(dec);After these lines, my code included the following lines. When i delete them, there is no problem now.
if(dec->capabilities & AV_CODEC_CAP_TRUNCATED)
decContext->flags |= AV_CODEC_FLAG_TRUNCATED;
decContext->flags2 |= AV_CODEC_FLAG2_CHUNKS;