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13 avril 2011, par ,
Mis à jour : Février 2012
Langue : français
Type : Video
Autres articles (97)
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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 (...) -
Multilang : améliorer l’interface pour les blocs multilingues
18 février 2011, parMultilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela. -
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 (7007)
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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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avfilter/vf_colormatrix : add bt.2020 colorspace
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8 juin 2011, par cedric -Oui typiquement dans ce cas "analyser_backend" est une fonction de portée générique qui mérite d’être dans le core. Il faut gérer ces problèmes au cas par cas.
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