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Exemple de boutons d’action pour une collection collaborative
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Autres articles (74)
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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. -
Participer à sa traduction
10 avril 2011Vous pouvez nous aider à améliorer les locutions utilisées dans le logiciel ou à traduire celui-ci dans n’importe qu’elle nouvelle langue permettant sa diffusion à de nouvelles communautés linguistiques.
Pour ce faire, on utilise l’interface de traduction de SPIP où l’ensemble des modules de langue de MediaSPIP sont à disposition. ll vous suffit de vous inscrire sur la liste de discussion des traducteurs pour demander plus d’informations.
Actuellement MediaSPIP n’est disponible qu’en français et (...) -
MediaSPIP v0.2
21 juin 2013, parMediaSPIP 0.2 est la première version de MediaSPIP stable.
Sa date de sortie officielle est le 21 juin 2013 et est annoncée ici.
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Comme pour la version précédente, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...)
Sur d’autres sites (14347)
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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.
-
Revision 6d15132742 : Change dx_time data type in vpxdec.c Change dx_time data type to int64_t to pre
22 février 2014, par James YuChanged Paths :
Modify /vpxdec.c
Change dx_time data type in vpxdec.cChange dx_time data type to int64_t to prevent
test time overflow when decoding long video.Change-Id : I3dd5e324a246843e07e635fd25c50e71e385ed70
Signed-off-by : James Yu <james.yu@linaro.org> -
dcaenc : Reverse data layout to prevent data copies during Huffman encoding introduction
6 janvier 2017, par Daniil Cherednik
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