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Autres articles (52)
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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. -
Support audio et vidéo HTML5
10 avril 2011MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...) -
Support de tous types de médias
10 avril 2011Contrairement à beaucoup de logiciels et autres plate-formes modernes de partage de documents, MediaSPIP a l’ambition de gérer un maximum de formats de documents différents qu’ils soient de type : images (png, gif, jpg, bmp et autres...) ; audio (MP3, Ogg, Wav et autres...) ; vidéo (Avi, MP4, Ogv, mpg, mov, wmv et autres...) ; contenu textuel, code ou autres (open office, microsoft office (tableur, présentation), web (html, css), LaTeX, Google Earth) (...)
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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; // FLAGCan 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 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; // FLAGCan anyone help me ?
Thanks.
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