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Linshizhi
ffmpeg.wasm-core
Commits
a392f82a
Commit
a392f82a
authored
Sep 29, 2009
by
Diego Biurrun
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Remove disabled code cruft.
Originally committed as revision 20087 to
svn://svn.ffmpeg.org/ffmpeg/trunk
parent
0e829974
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52 deletions
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ffserver.c
ffserver.c
+0
-52
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ffserver.c
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a392f82a
...
@@ -1882,36 +1882,6 @@ static void compute_status(HTTPContext *c)
...
@@ -1882,36 +1882,6 @@ static void compute_status(HTTPContext *c)
stream
=
stream
->
next
;
stream
=
stream
->
next
;
}
}
#if 0
{
float avg;
AVCodecContext *enc;
char buf[1024];
/* feed status */
stream = first_feed;
while (stream != NULL) {
url_fprintf(pb, "<h1>Feed '%s'</h1>\n", stream->filename);
url_fprintf(pb, "<table>\n");
url_fprintf(pb, "<tr><td>Parameters<td>Frame count<td>Size<td>Avg bitrate (kbits/s)\n");
for(i=0;i<stream->nb_streams;i++) {
AVStream *st = stream->streams[i];
FeedData *fdata = st->priv_data;
enc = st->codec;
avcodec_string(buf, sizeof(buf), enc);
avg = fdata->avg_frame_size * (float)enc->rate * 8.0;
if (enc->codec->type == CODEC_TYPE_AUDIO && enc->frame_size > 0)
avg /= enc->frame_size;
url_fprintf(pb, "<tr><td>%s <td> %d <td> %"PRId64" <td> %0.1f\n",
buf, enc->frame_number, fdata->data_count, avg / 1000.0);
}
url_fprintf(pb, "</table>\n");
stream = stream->next_feed;
}
}
#endif
/* connection status */
/* connection status */
url_fprintf
(
pb
,
"<h2>Connection Status</h2>
\n
"
);
url_fprintf
(
pb
,
"<h2>Connection Status</h2>
\n
"
);
...
@@ -2021,12 +1991,6 @@ static int open_input_stream(HTTPContext *c, const char *info)
...
@@ -2021,12 +1991,6 @@ static int open_input_stream(HTTPContext *c, const char *info)
if
(
input_filename
[
0
]
==
'\0'
)
if
(
input_filename
[
0
]
==
'\0'
)
return
-
1
;
return
-
1
;
#if 0
{ time_t when = stream_pos / 1000000;
http_log("Stream pos = %"PRId64", time=%s", stream_pos, ctime(&when));
}
#endif
/* open stream */
/* open stream */
if
((
ret
=
av_open_input_file
(
&
s
,
input_filename
,
c
->
stream
->
ifmt
,
if
((
ret
=
av_open_input_file
(
&
s
,
input_filename
,
c
->
stream
->
ifmt
,
buf_size
,
c
->
stream
->
ap_in
))
<
0
)
{
buf_size
,
c
->
stream
->
ap_in
))
<
0
)
{
...
@@ -2234,14 +2198,6 @@ static int http_prepare_data(HTTPContext *c)
...
@@ -2234,14 +2198,6 @@ static int http_prepare_data(HTTPContext *c)
if
(
ist
->
start_time
!=
AV_NOPTS_VALUE
)
if
(
ist
->
start_time
!=
AV_NOPTS_VALUE
)
c
->
cur_pts
-=
av_rescale_q
(
ist
->
start_time
,
ist
->
time_base
,
AV_TIME_BASE_Q
);
c
->
cur_pts
-=
av_rescale_q
(
ist
->
start_time
,
ist
->
time_base
,
AV_TIME_BASE_Q
);
c
->
cur_frame_duration
=
av_rescale_q
(
pkt
.
duration
,
ist
->
time_base
,
AV_TIME_BASE_Q
);
c
->
cur_frame_duration
=
av_rescale_q
(
pkt
.
duration
,
ist
->
time_base
,
AV_TIME_BASE_Q
);
#if 0
printf("index=%d pts=%0.3f duration=%0.6f\n",
pkt.stream_index,
(double)c->cur_pts /
AV_TIME_BASE,
(double)c->cur_frame_duration /
AV_TIME_BASE);
#endif
/* find RTP context */
/* find RTP context */
c
->
packet_stream_index
=
pkt
.
stream_index
;
c
->
packet_stream_index
=
pkt
.
stream_index
;
ctx
=
c
->
rtp_ctx
[
c
->
packet_stream_index
];
ctx
=
c
->
rtp_ctx
[
c
->
packet_stream_index
];
...
@@ -3072,14 +3028,6 @@ static void rtsp_cmd_play(HTTPContext *c, const char *url, RTSPMessageHeader *h)
...
@@ -3072,14 +3028,6 @@ static void rtsp_cmd_play(HTTPContext *c, const char *url, RTSPMessageHeader *h)
return
;
return
;
}
}
#if 0
/* XXX: seek in stream */
if (h->range_start != AV_NOPTS_VALUE) {
printf("range_start=%0.3f\n", (double)h->range_start / AV_TIME_BASE);
av_seek_frame(rtp_c->fmt_in, -1, h->range_start);
}
#endif
rtp_c
->
state
=
HTTPSTATE_SEND_DATA
;
rtp_c
->
state
=
HTTPSTATE_SEND_DATA
;
/* now everything is OK, so we can send the connection parameters */
/* now everything is OK, so we can send the connection parameters */
...
...
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