Recherche avancée
Médias (91)
-
Head down (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
-
Echoplex (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
-
Discipline (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
-
Letting you (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
-
1 000 000 (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
-
999 999 (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
Autres articles (23)
-
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. -
Submit bugs and patches
13 avril 2011Unfortunately a software is never perfect.
If you think you have found a bug, report it using our ticket system. Please to help us to fix it by providing the following information : the browser you are using, including the exact version as precise an explanation as possible of the problem if possible, the steps taken resulting in the problem a link to the site / page in question
If you think you have solved the bug, fill in a ticket and attach to it a corrective patch.
You may also (...) -
Les statuts des instances de mutualisation
13 mars 2010, parPour des raisons de compatibilité générale du plugin de gestion de mutualisations avec les fonctions originales de SPIP, les statuts des instances sont les mêmes que pour tout autre objets (articles...), seuls leurs noms dans l’interface change quelque peu.
Les différents statuts possibles sont : prepa (demandé) qui correspond à une instance demandée par un utilisateur. Si le site a déjà été créé par le passé, il est passé en mode désactivé. publie (validé) qui correspond à une instance validée par un (...)
Sur d’autres sites (4133)
-
avfilter/vf_lut* : strip color volume metadata
26 avril 2024, par Niklas Haas -
lavd/opengl : select the GL context before drawing
12 juillet 2023, par Nicolas George -
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.
Syndiquer tout le site
