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Exemple de boutons d’action pour une collection collaborative
27 février 2013, par
Mis à jour : Mars 2013
Langue : français
Type : Image
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Exemple de boutons d’action pour une collection personnelle
27 février 2013, par
Mis à jour : Février 2013
Langue : English
Type : Image
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Collections - Formulaire de création rapide
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Mis à jour : Février 2013
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Type : Image
Autres articles (39)
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Les autorisations surchargées par les plugins
27 avril 2010, parMediaspip core
autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs -
MediaSPIP Player : problèmes potentiels
22 février 2011, parLe lecteur ne fonctionne pas sur Internet Explorer
Sur Internet Explorer (8 et 7 au moins), le plugin utilise le lecteur Flash flowplayer pour lire vidéos et son. Si le lecteur ne semble pas fonctionner, cela peut venir de la configuration du mod_deflate d’Apache.
Si dans la configuration de ce module Apache vous avez une ligne qui ressemble à la suivante, essayez de la supprimer ou de la commenter pour voir si le lecteur fonctionne correctement : /** * GeSHi (C) 2004 - 2007 Nigel McNie, (...) -
Keeping control of your media in your hands
13 avril 2011, parThe vocabulary used on this site and around MediaSPIP in general, aims to avoid reference to Web 2.0 and the companies that profit from media-sharing.
While using MediaSPIP, you are invited to avoid using words like "Brand", "Cloud" and "Market".
MediaSPIP is designed to facilitate the sharing of creative media online, while allowing authors to retain complete control of their work.
MediaSPIP aims to be accessible to as many people as possible and development is based on expanding the (...)
Sur d’autres sites (7090)
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How to get FFplay to stop looping last audio packet ?
29 décembre 2020, par occvtechI'm trying to get FFplay to simply stop (pause) on the last image when playing through.



The default behavior for FFplay appears to use the
-loopsperimeter, which causes the last audio packet to be looped - even though the image appears in a paused state.


Is there a way to also stop playing audio on end of file ?


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How to get FFplay to stop looping last audio packet ?
2 juillet 2015, par occvtechI’m trying to get FFplay to simply stop (pause) on the last image when playing through.
The default behavior for FFplay appears to use the
-loopsperimeter, which causes the last audio packet to be looped - even though the image appears in a paused state.Is there a way to also stop playing audio on end of file ?
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RTSP client can not be play video
13 novembre 2018, par Harshil MakwanaI added and updated below API inside ffserver code inside ffmpeg code :
AVPacket *dataPacket;
void setAVPacket(AVPacket *packet)
{
if (packet && packet->data)
{
pthread_mutex_lock(&lock);
if (isSend == 1)
{
dataPacket = packet;
}
else
{
if (packet != NULL)
{
av_packet_unref(packet);
free(packet);
packet = NULL;
}
}
pthread_mutex_unlock(&lock);
}
static int http_prepare_data(HTTPContext *c)
{
int i, len, ret;
AVFormatContext *ctx;
av_freep(&c->pb_buffer);
switch(c->state) {
case HTTPSTATE_SEND_DATA_HEADER:
ctx = avformat_alloc_context();
if (!ctx)
return AVERROR(ENOMEM);
c->pfmt_ctx = ctx;
av_dict_copy(&(c->pfmt_ctx->metadata), c->stream->metadata, 0);
for(i=0;istream->nb_streams;i++) {
LayeredAVStream *src;
AVStream *st = avformat_new_stream(c->pfmt_ctx, NULL);
if (!st)
return AVERROR(ENOMEM);
/* if file or feed, then just take streams from FFServerStream
* struct */
if (!c->stream->feed ||
c->stream->feed == c->stream)
src = c->stream->streams[i];
else
src = c->stream->feed->streams[c->stream->feed_streams[i]];
unlayer_stream(c->pfmt_ctx->streams[i], src); //TODO we no longer copy st->internal, does this matter?
av_assert0(!c->pfmt_ctx->streams[i]->priv_data);
if (src->codec->flags & AV_CODEC_FLAG_BITEXACT)
c->pfmt_ctx->flags |= AVFMT_FLAG_BITEXACT;
}
/* set output format parameters */
c->pfmt_ctx->oformat = c->stream->fmt;
av_assert0(c->pfmt_ctx->nb_streams == c->stream->nb_streams);
c->got_key_frame = 0;
/* prepare header and save header data in a stream */
if (avio_open_dyn_buf(&c->pfmt_ctx->pb) < 0) {
/* XXX: potential leak */
return -1;
}
c->pfmt_ctx->pb->seekable = 0;
/*
* HACK to avoid MPEG-PS muxer to spit many underflow errors
* Default value from FFmpeg
* Try to set it using configuration option
*/
c->pfmt_ctx->max_delay = (int)(0.7*AV_TIME_BASE);
if ((ret = avformat_write_header(c->pfmt_ctx, NULL)) < 0) {
http_log("Error writing output header for stream '%s': %s\n",
c->stream->filename, av_err2str(ret));
return ret;
}
av_dict_free(&c->pfmt_ctx->metadata);
len = avio_close_dyn_buf(c->pfmt_ctx->pb, &c->pb_buffer);
c->buffer_ptr = c->pb_buffer;
c->buffer_end = c->pb_buffer + len;
c->state = HTTPSTATE_SEND_DATA;
c->last_packet_sent = 0;
break;
case HTTPSTATE_SEND_DATA:
/* find a new packet */
/* read a packet from the input stream */
if (c->stream->feed)
ffm_set_write_index(c->fmt_in,
c->stream->feed->feed_write_index,
c->stream->feed->feed_size);
if (c->stream->max_time &&
c->stream->max_time + c->start_time - cur_time < 0)
/* We have timed out */
c->state = HTTPSTATE_SEND_DATA_TRAILER;
else {
AVPacket pkt;
redo:
ret = av_read_frame(c->fmt_in, &pkt);
if (ret < 0) {
if (c->stream->feed) {
/* if coming from feed, it means we reached the end of the
* ffm file, so must wait for more data */
c->state = HTTPSTATE_WAIT_FEED;
return 1; /* state changed */
}
if (ret == AVERROR(EAGAIN)) {
/* input not ready, come back later */
return 0;
}
if (c->stream->loop) {
avformat_close_input(&c->fmt_in);
if (open_input_stream(c, "") < 0)
goto no_loop;
goto redo;
} else {
no_loop:
/* must send trailer now because EOF or error */
c->state = HTTPSTATE_SEND_DATA_TRAILER;
}
} else {
int source_index = pkt.stream_index;
/* update first pts if needed */
if (c->first_pts == AV_NOPTS_VALUE && pkt.dts != AV_NOPTS_VALUE) {
c->first_pts = av_rescale_q(pkt.dts, c->fmt_in->streams[pkt.stream_index]->time_base, AV_TIME_BASE_Q);
c->start_time = cur_time;
}
/* send it to the appropriate stream */
if (c->stream->feed) {
/* if coming from a feed, select the right stream */
if (c->switch_pending) {
c->switch_pending = 0;
for(i=0;istream->nb_streams;i++) {
if (c->switch_feed_streams[i] == pkt.stream_index)
if (pkt.flags & AV_PKT_FLAG_KEY)
c->switch_feed_streams[i] = -1;
if (c->switch_feed_streams[i] >= 0)
c->switch_pending = 1;
}
}
for(i=0;istream->nb_streams;i++) {
if (c->stream->feed_streams[i] == pkt.stream_index) {
AVStream *st = c->fmt_in->streams[source_index];
pkt.stream_index = i;
if (pkt.flags & AV_PKT_FLAG_KEY &&
(st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO ||
c->stream->nb_streams == 1))
c->got_key_frame = 1;
if (!c->stream->send_on_key || c->got_key_frame)
goto send_it;
}
}
} else {
AVStream *ist, *ost;
send_it:
ist = c->fmt_in->streams[source_index];
/* specific handling for RTP: we use several
* output streams (one for each RTP connection).
* XXX: need more abstract handling */
if (c->is_packetized) {
/* compute send time and duration */
if (pkt.dts != AV_NOPTS_VALUE) {
c->cur_pts = av_rescale_q(pkt.dts, ist->time_base, AV_TIME_BASE_Q);
c->cur_pts -= c->first_pts;
}
c->cur_frame_duration = av_rescale_q(pkt.duration, ist->time_base, AV_TIME_BASE_Q);
/* find RTP context */
c->packet_stream_index = pkt.stream_index;
ctx = c->rtp_ctx[c->packet_stream_index];
if(!ctx) {
av_packet_unref(&pkt);
break;
}
/* only one stream per RTP connection */
pkt.stream_index = 0;
} else {
ctx = c->pfmt_ctx;
/* Fudge here */
}
if (c->is_packetized) {
int max_packet_size;
if (c->rtp_protocol == RTSP_LOWER_TRANSPORT_TCP)
max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
else
max_packet_size = c->rtp_handles[c->packet_stream_index]->max_packet_size;
ret = ffio_open_dyn_packet_buf(&ctx->pb,
max_packet_size);
} else
ret = avio_open_dyn_buf(&ctx->pb);
if (ret < 0) {
/* XXX: potential leak */
return -1;
}
ost = ctx->streams[pkt.stream_index];
ctx->pb->seekable = 0;
if (pkt.dts != AV_NOPTS_VALUE)
pkt.dts = av_rescale_q(pkt.dts, ist->time_base,
ost->time_base);
if (pkt.pts != AV_NOPTS_VALUE)
pkt.pts = av_rescale_q(pkt.pts, ist->time_base,
ost->time_base);
pkt.duration = av_rescale_q(pkt.duration, ist->time_base,
ost->time_base);
if ((ret = av_write_frame(ctx, &pkt)) < 0) {
http_log("Error writing frame to output for stream '%s': %s\n",
c->stream->filename, av_err2str(ret));
c->state = HTTPSTATE_SEND_DATA_TRAILER;
}
av_freep(&c->pb_buffer);
len = avio_close_dyn_buf(ctx->pb, &c->pb_buffer);
ctx->pb = NULL;
c->cur_frame_bytes = len;
c->buffer_ptr = c->pb_buffer;
c->buffer_end = c->pb_buffer + len;
if (len == 0) {
av_packet_unref(&pkt);
goto redo;
}
}
av_packet_unref(&pkt);
}
}
break;
default:
case HTTPSTATE_SEND_DATA_TRAILER:
/* last packet test ? */
if (c->last_packet_sent || c->is_packetized)
return -1;
ctx = c->pfmt_ctx;
/* prepare header */
if (avio_open_dyn_buf(&ctx->pb) < 0) {
/* XXX: potential leak */
return -1;
}
c->pfmt_ctx->pb->seekable = 0;
av_write_trailer(ctx);
len = avio_close_dyn_buf(ctx->pb, &c->pb_buffer);
c->buffer_ptr = c->pb_buffer;
c->buffer_end = c->pb_buffer + len;
c->last_packet_sent = 1;
break;
}
return 0;
}if you see here there is API named setAVPacket(), through which I am passing my H264 based encoded packet to RTSPServer. And same AVPacket used by Other function named http_prepare_data(), which will be called when PLAY request is coming.
After implementing above code I can do handshake of RTSP and server can send RTP packet to client, but no player(tried VLC, ffplayer) can play video.
Can you help me on this ?
Very much thanks you in advance.
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