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authorGiancarlo Formicuccia <ilsensine@inwind.it>2003-07-11 22:30:12 +0000
committerMichael Niedermayer <michaelni@gmx.at>2003-07-11 22:30:12 +0000
commit1bc1cfdddf7ab8ef50d0fc888808d6b609eb5d8d (patch)
tree76a842f76f221a2343ba29ac6aa623d49d465d54
parent6bc114b2fb026ff0c521caf206ad08fdb5910383 (diff)
downloadffmpeg-1bc1cfdddf7ab8ef50d0fc888808d6b609eb5d8d.tar.gz
- Gracefully handle the case where not all the streams are requested/wanted
from the client. Simply ignore the unwanted/unasked streams. - Don't need to pool() for every input character! (the socket is nonblocking, so the loop is ok). - Partially resurrect compute_send_delay for avoiding udp flood. Without a similar patch, udp transmission is seriously unreliable. (note that we don't link to a specific input reference stream, it's not needed as the pts values should be coherent anyway. Also, non-monotonic pts progression is unimportant in the long term). - rtsp_cmd_pause must reset the time reference patch by (Giancarlo Formicuccia <ilsensine at inwind dot it>) Originally committed as revision 2034 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r--ffserver.c25
1 files changed, 19 insertions, 6 deletions
diff --git a/ffserver.c b/ffserver.c
index 6cebfb4942..a0ded91e6d 100644
--- a/ffserver.c
+++ b/ffserver.c
@@ -113,6 +113,7 @@ typedef struct HTTPContext {
AVFormatContext *fmt_in;
long start_time; /* In milliseconds - this wraps fairly often */
int64_t first_pts; /* initial pts value */
+ int64_t cur_pts; /* current pts value */
int pts_stream_index; /* stream we choose as clock reference */
/* output format handling */
struct FFStream *stream;
@@ -786,6 +787,7 @@ static int handle_connection(HTTPContext *c)
if (!(c->poll_entry->revents & POLLIN))
return 0;
/* read the data */
+ read_loop:
len = read(c->fd, c->buffer_ptr, 1);
if (len < 0) {
if (errno != EAGAIN && errno != EINTR)
@@ -810,7 +812,7 @@ static int handle_connection(HTTPContext *c)
} else if (ptr >= c->buffer_end) {
/* request too long: cannot do anything */
return -1;
- }
+ } else goto read_loop;
}
break;
@@ -2078,11 +2080,20 @@ static int av_read_frame(AVFormatContext *s, AVPacket *pkt)
static int compute_send_delay(HTTPContext *c)
{
int datarate = 8 * get_longterm_datarate(&c->datarate, c->data_count);
+ int64_t delta_pts;
+ int64_t time_pts;
+ int m_delay;
if (datarate > c->stream->bandwidth * 2000) {
return 1000;
}
- return 0;
+ if(!c->stream->feed && c->first_pts!=AV_NOPTS_VALUE) {
+ time_pts = ((int64_t)(cur_time - c->start_time) * c->fmt_in->pts_den) /
+ ((int64_t) c->fmt_in->pts_num*1000);
+ delta_pts = c->cur_pts - time_pts;
+ m_delay = (delta_pts * 1000 * c->fmt_in->pts_num) / c->fmt_in->pts_den;
+ return m_delay>0 ? m_delay : 0;
+ } else return 0;
}
#endif
@@ -2189,9 +2200,11 @@ static int http_prepare_data(HTTPContext *c)
}
} else {
/* update first pts if needed */
- if (c->first_pts == AV_NOPTS_VALUE)
+ if (c->first_pts == AV_NOPTS_VALUE) {
c->first_pts = pkt.pts;
-
+ c->start_time = cur_time;
+ }
+ c->cur_pts = pkt.pts;
/* send it to the appropriate stream */
if (c->stream->feed) {
/* if coming from a feed, select the right stream */
@@ -2239,7 +2252,7 @@ static int http_prepare_data(HTTPContext *c)
ctx = c->rtp_ctx[c->packet_stream_index];
if(!ctx) {
av_free_packet(&pkt);
- return -1;
+ break;
}
codec = &ctx->streams[0]->codec;
/* only one stream per RTP connection */
@@ -3044,7 +3057,7 @@ static void rtsp_cmd_pause(HTTPContext *c, const char *url, RTSPHeader *h)
}
rtp_c->state = HTTPSTATE_READY;
-
+ rtp_c->first_pts = AV_NOPTS_VALUE;
/* now everything is OK, so we can send the connection parameters */
rtsp_reply_header(c, RTSP_STATUS_OK);
/* session ID */