Merge remote-tracking branch 'qatar/master'

[ffmpeg] / libavformat / rtspdec.c

diff --git a/libavformat/rtspdec.c b/libavformat/rtspdec.c
index 8fa419714ebec16922983077aa972a259cf5fdfd..cf9591511008fcf98a7730fe28a9dfb218a6feca 100644 (file)
--- a/libavformat/rtspdec.c
+++ b/libavformat/rtspdec.c
@@ -21,6 +21,7 @@
 
 #include "libavutil/avstring.h"
 #include "libavutil/intreadwrite.h"
+#include "libavutil/opt.h"
 #include "avformat.h"
 
 #include "internal.h"
@@ -28,9 +29,7 @@
 #include "os_support.h"
 #include "rtsp.h"
 #include "rdt.h"
-
-//#define DEBUG
-//#define DEBUG_RTP_TCP
+#include "url.h"
 
 static int rtsp_read_play(AVFormatContext *s)
 {
@@ -164,7 +163,12 @@ static int rtsp_read_header(AVFormatContext *s,
         return AVERROR(ENOMEM);
     rt->real_setup = rt->real_setup_cache + s->nb_streams;
 
-    if (ap->initial_pause) {
+#if FF_API_FORMAT_PARAMETERS
+    if (ap->initial_pause)
+        rt->initial_pause = ap->initial_pause;
+#endif
+
+    if (rt->initial_pause) {
          /* do not start immediately */
     } else {
          if (rtsp_read_play(s) < 0) {
@@ -184,9 +188,7 @@ int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
     int id, len, i, ret;
     RTSPStream *rtsp_st;
 
-#ifdef DEBUG_RTP_TCP
     av_dlog(s, "tcp_read_packet:\n");
-#endif
 redo:
     for (;;) {
         RTSPMessageHeader reply;
@@ -200,18 +202,16 @@ redo:
         if (rt->state != RTSP_STATE_STREAMING)
             return 0;
     }
-    ret = url_read_complete(rt->rtsp_hd, buf, 3);
+    ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
     if (ret != 3)
         return -1;
     id  = buf[0];
     len = AV_RB16(buf + 1);
-#ifdef DEBUG_RTP_TCP
     av_dlog(s, "id=%d len=%d\n", id, len);
-#endif
     if (len > buf_size || len < 12)
         goto redo;
     /* get the data */
-    ret = url_read_complete(rt->rtsp_hd, buf, len);
+    ret = ffurl_read_complete(rt->rtsp_hd, buf, len);
     if (ret != len)
         return -1;
     if (rt->transport == RTSP_TRANSPORT_RDT &&
@@ -340,7 +340,9 @@ retry:
 
     /* send dummy request to keep TCP connection alive */
     if ((av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) {
-        if (rt->server_type == RTSP_SERVER_WMS) {
+        if (rt->server_type == RTSP_SERVER_WMS ||
+           (rt->server_type != RTSP_SERVER_REAL &&
+            rt->get_parameter_supported)) {
             ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL);
         } else {
             ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL);
@@ -396,6 +398,18 @@ static int rtsp_read_close(AVFormatContext *s)
     return 0;
 }
 
+static const AVOption options[] = {
+    { "initial_pause",  "Don't start playing the stream immediately", offsetof(RTSPState, initial_pause),  FF_OPT_TYPE_INT, {.dbl = 0}, 0, 1, AV_OPT_FLAG_DECODING_PARAM },
+    { NULL },
+};
+
+const AVClass rtsp_demuxer_class = {
+    .class_name     = "RTSP demuxer",
+    .item_name      = av_default_item_name,
+    .option         = options,
+    .version        = LIBAVUTIL_VERSION_INT,
+};
+
 AVInputFormat ff_rtsp_demuxer = {
     "rtsp",
     NULL_IF_CONFIG_SMALL("RTSP input format"),
@@ -408,4 +422,5 @@ AVInputFormat ff_rtsp_demuxer = {
     .flags = AVFMT_NOFILE,
     .read_play = rtsp_read_play,
     .read_pause = rtsp_read_pause,
+    .priv_class = &rtsp_demuxer_class,
 };