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Richard Stallman et le logiciel libre
19 octobre 2011, par
Mis à jour : Mai 2013
Langue : français
Type : Texte
Autres articles (84)
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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 (...) -
Amélioration de la version de base
13 septembre 2013Jolie sélection multiple
Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
Il suffit pour cela d’activer le plugin Chosen (Configuration générale du site > Gestion des plugins), puis de configurer le plugin (Les squelettes > Chosen) en activant l’utilisation de Chosen dans le site public et en spécifiant les éléments de formulaires à améliorer, par exemple select[multiple] pour les listes à sélection multiple (...) -
Le plugin : Podcasts.
14 juillet 2010, parLe problème du podcasting est à nouveau un problème révélateur de la normalisation des transports de données sur Internet.
Deux formats intéressants existent : Celui développé par Apple, très axé sur l’utilisation d’iTunes dont la SPEC est ici ; Le format "Media RSS Module" qui est plus "libre" notamment soutenu par Yahoo et le logiciel Miro ;
Types de fichiers supportés dans les flux
Le format d’Apple n’autorise que les formats suivants dans ses flux : .mp3 audio/mpeg .m4a audio/x-m4a .mp4 (...)
Sur d’autres sites (9524)
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How to understand the given ffplay C code snippet ?
20 juillet 2015, par Jerikc XIONGThe following code snippet is from ffplay :
static int decoder_decode_frame(Decoder *d, AVFrame *frame, AVSubtitle *sub) {
int got_frame = 0;
do {
int ret = -1;
if (d->queue->abort_request)
return -1;
if (!d->packet_pending || d->queue->serial != d->pkt_serial) {
AVPacket pkt;
do {
if (d->queue->nb_packets == 0)
SDL_CondSignal(d->empty_queue_cond);
if (packet_queue_get(d->queue, &pkt, 1, &d->pkt_serial) < 0)
return -1;
if (pkt.data == flush_pkt.data) {
avcodec_flush_buffers(d->avctx);
d->finished = 0;
d->next_pts = d->start_pts;
d->next_pts_tb = d->start_pts_tb;
}
} while (pkt.data == flush_pkt.data || d->queue->serial != d->pkt_serial);
av_free_packet(&d->pkt);
d->pkt_temp = d->pkt = pkt;
d->packet_pending = 1;
}
switch (d->avctx->codec_type) {
case AVMEDIA_TYPE_VIDEO:
ret = avcodec_decode_video2(d->avctx, frame, &got_frame, &d->pkt_temp);
if (got_frame) {
if (decoder_reorder_pts == -1) {
frame->pts = av_frame_get_best_effort_timestamp(frame);
} else if (decoder_reorder_pts) {
frame->pts = frame->pkt_pts;
} else {
frame->pts = frame->pkt_dts;
}
}
break;
case AVMEDIA_TYPE_AUDIO:
ret = avcodec_decode_audio4(d->avctx, frame, &got_frame, &d->pkt_temp);
if (got_frame) {
AVRational tb = (AVRational){1, frame->sample_rate};
if (frame->pts != AV_NOPTS_VALUE)
frame->pts = av_rescale_q(frame->pts, d->avctx->time_base, tb);
else if (frame->pkt_pts != AV_NOPTS_VALUE)
frame->pts = av_rescale_q(frame->pkt_pts, av_codec_get_pkt_timebase(d->avctx), tb);
else if (d->next_pts != AV_NOPTS_VALUE)
frame->pts = av_rescale_q(d->next_pts, d->next_pts_tb, tb);
if (frame->pts != AV_NOPTS_VALUE) {
d->next_pts = frame->pts + frame->nb_samples;
d->next_pts_tb = tb;
}
}
break;
case AVMEDIA_TYPE_SUBTITLE:
ret = avcodec_decode_subtitle2(d->avctx, sub, &got_frame, &d->pkt_temp);
break;
}
if (ret < 0) {
d->packet_pending = 0;
} else {
d->pkt_temp.dts =
d->pkt_temp.pts = AV_NOPTS_VALUE;
if (d->pkt_temp.data) {
if (d->avctx->codec_type != AVMEDIA_TYPE_AUDIO)
ret = d->pkt_temp.size;
d->pkt_temp.data += ret;
d->pkt_temp.size -= ret;
if (d->pkt_temp.size <= 0)
d->packet_pending = 0;
} else {
if (!got_frame) {
d->packet_pending = 0;
d->finished = d->pkt_serial; // FLAG
}
}
}
} while (!got_frame && !d->finished);
return got_frame;
}It’s difficult for me to understand the following code :
d->finished = d->pkt_serial; // FLAG
Can anyone help me ?
Thanks.
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Set H264 high quality encoding in FFMPEG
25 décembre 2015, par hiitigerI’m working on a function, using ffmpeg to encode a serial images to a MP4 file.
I use H264 codec, and want to have a high quality output.Other things now goes well, but the video quality looks obvious loss.
After some Google, I set codec context like this
video_st = avformat_new_stream(oc, 0);
....
pVideoCodec = avcodec_find_encoder(fmt->video_codec); //AV_CODEC_ID_H264
....
avcodec_get_context_defaults3(video_st->codec, pVideoCodec);
video_st->codec->codec_id = fmt->video_codec;
video_st->codec->width = m_outWidth;
video_st->codec->height = m_outHeight;
video_st->codec->time_base.den = m_fps;
video_st->codec->time_base.num = 1;
video_st->codec->pix_fmt = AV_PIX_FMT_YUV420P;
AVDictionary *param = nullptr;
av_dict_set(&param, "qp", "0", 0);
av_dict_set(&param, "preset", "medium", 0);I also try to set bit_rate. But all turned out to be bad.
I want a visually loss less output video.
So what parameters should I set or anything else did I missed ?
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How to explain the given ffplay C code snippet ?
20 juillet 2015, par Jerikc XIONGThe following code snippet is from ffplay :
static int decoder_decode_frame(Decoder *d, AVFrame *frame, AVSubtitle *sub) {
int got_frame = 0;
do {
int ret = -1;
if (d->queue->abort_request)
return -1;
if (!d->packet_pending || d->queue->serial != d->pkt_serial) {
AVPacket pkt;
do {
if (d->queue->nb_packets == 0)
SDL_CondSignal(d->empty_queue_cond);
if (packet_queue_get(d->queue, &pkt, 1, &d->pkt_serial) < 0)
return -1;
if (pkt.data == flush_pkt.data) {
avcodec_flush_buffers(d->avctx);
d->finished = 0;
d->next_pts = d->start_pts;
d->next_pts_tb = d->start_pts_tb;
}
} while (pkt.data == flush_pkt.data || d->queue->serial != d->pkt_serial);
av_free_packet(&d->pkt);
d->pkt_temp = d->pkt = pkt;
d->packet_pending = 1;
}
switch (d->avctx->codec_type) {
case AVMEDIA_TYPE_VIDEO:
ret = avcodec_decode_video2(d->avctx, frame, &got_frame, &d->pkt_temp);
if (got_frame) {
if (decoder_reorder_pts == -1) {
frame->pts = av_frame_get_best_effort_timestamp(frame);
} else if (decoder_reorder_pts) {
frame->pts = frame->pkt_pts;
} else {
frame->pts = frame->pkt_dts;
}
}
break;
case AVMEDIA_TYPE_AUDIO:
ret = avcodec_decode_audio4(d->avctx, frame, &got_frame, &d->pkt_temp);
if (got_frame) {
AVRational tb = (AVRational){1, frame->sample_rate};
if (frame->pts != AV_NOPTS_VALUE)
frame->pts = av_rescale_q(frame->pts, d->avctx->time_base, tb);
else if (frame->pkt_pts != AV_NOPTS_VALUE)
frame->pts = av_rescale_q(frame->pkt_pts, av_codec_get_pkt_timebase(d->avctx), tb);
else if (d->next_pts != AV_NOPTS_VALUE)
frame->pts = av_rescale_q(d->next_pts, d->next_pts_tb, tb);
if (frame->pts != AV_NOPTS_VALUE) {
d->next_pts = frame->pts + frame->nb_samples;
d->next_pts_tb = tb;
}
}
break;
case AVMEDIA_TYPE_SUBTITLE:
ret = avcodec_decode_subtitle2(d->avctx, sub, &got_frame, &d->pkt_temp);
break;
}
if (ret < 0) {
d->packet_pending = 0;
} else {
d->pkt_temp.dts =
d->pkt_temp.pts = AV_NOPTS_VALUE;
if (d->pkt_temp.data) {
if (d->avctx->codec_type != AVMEDIA_TYPE_AUDIO)
ret = d->pkt_temp.size;
d->pkt_temp.data += ret;
d->pkt_temp.size -= ret;
if (d->pkt_temp.size <= 0)
d->packet_pending = 0;
} else {
if (!got_frame) {
d->packet_pending = 0;
d->finished = d->pkt_serial; // FLAG
}
}
}
} while (!got_frame && !d->finished);
return got_frame;
}It’s difficult for me to understand the following code :
d->finished = d->pkt_serial; // FLAG
Can anyone help me ?
Thanks.