avserver: fix build after the next bump.

Now that 0.8 is out we can reapply this commit. It breaks shared
avserver builds due to avserver using internal libavformat symbols,
which are now hidden, so this commit also disables avserver with
--enable-shared.

avserver.c

@@ -26,13 +26,16 @@
 #include <string.h>
 #include <stdlib.h>
 #include "libavformat/avformat.h"
+// FIXME those are internal headers, avserver _really_ shouldn't use them
 #include "libavformat/ffm.h"
 #include "libavformat/network.h"
 #include "libavformat/os_support.h"
 #include "libavformat/rtpdec.h"
 #include "libavformat/rtsp.h"
-// XXX for ffio_open_dyn_packet_buffer, to be removed
 #include "libavformat/avio_internal.h"
+#include "libavformat/internal.h"
+#include "libavformat/url.h"
+
 #include "libavutil/avstring.h"
 #include "libavutil/lfg.h"
 #include "libavutil/dict.h"
@@ -867,7 +870,7 @@ static void close_connection(HTTPContext *c)
         }
         h = c->rtp_handles[i];
         if (h)
-            url_close(h);
+            ffurl_close(h);
     }
 
     ctx = &c->fmt_ctx;
@@ -2248,7 +2251,6 @@ static int http_prepare_data(HTTPContext *c)
          * Default value from Libav
          * Try to set it use configuration option
          */
-        c->fmt_ctx.preload   = (int)(0.5*AV_TIME_BASE);
         c->fmt_ctx.max_delay = (int)(0.7*AV_TIME_BASE);
 
         if (avformat_write_header(&c->fmt_ctx, NULL) < 0) {
@@ -2367,7 +2369,7 @@ static int http_prepare_data(HTTPContext *c)
                         if (c->rtp_protocol == RTSP_LOWER_TRANSPORT_TCP)
                             max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
                         else
-                            max_packet_size = url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]);
+                            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);
@@ -2520,8 +2522,8 @@ static int http_send_data(HTTPContext *c)
                 } else {
                     /* send RTP packet directly in UDP */
                     c->buffer_ptr += 4;
-                    url_write(c->rtp_handles[c->packet_stream_index],
-                              c->buffer_ptr, len);
+                    ffurl_write(c->rtp_handles[c->packet_stream_index],
+                                c->buffer_ptr, len);
                     c->buffer_ptr += len;
                     /* here we continue as we can send several packets per 10 ms slot */
                 }
@@ -3404,10 +3406,10 @@ static int rtp_new_av_stream(HTTPContext *c,
                      "rtp://%s:%d", ipaddr, ntohs(dest_addr->sin_port));
         }
 
-        if (url_open(&h, ctx->filename, AVIO_FLAG_WRITE) < 0)
+        if (ffurl_open(&h, ctx->filename, AVIO_FLAG_WRITE, NULL, NULL) < 0)
             goto fail;
         c->rtp_handles[stream_index] = h;
-        max_packet_size = url_get_max_packet_size(h);
+        max_packet_size = h->max_packet_size;
         break;
     case RTSP_LOWER_TRANSPORT_TCP:
         /* RTP/TCP case */
@@ -3430,7 +3432,7 @@ static int rtp_new_av_stream(HTTPContext *c,
     if (avformat_write_header(ctx, NULL) < 0) {
     fail:
         if (h)
-            url_close(h);
+            ffurl_close(h);
         av_free(ctx);
         return -1;
     }
@@ -3467,7 +3469,7 @@ static AVStream *add_av_stream1(FFStream *stream, AVCodecContext *codec, int cop
     }
     fst->priv_data = av_mallocz(sizeof(FeedData));
     fst->index = stream->nb_streams;
-    av_set_pts_info(fst, 33, 1, 90000);
+    avpriv_set_pts_info(fst, 33, 1, 90000);
     fst->sample_aspect_ratio = codec->sample_aspect_ratio;
     stream->streams[stream->nb_streams++] = fst;
     return fst;

configure

@@ -1534,7 +1534,7 @@ avconv_deps="avcodec avformat swscale"
 avplay_deps="avcodec avformat swscale sdl"
 avplay_select="rdft"
 avprobe_deps="avcodec avformat"
-avserver_deps="avformat ffm_muxer fork rtp_protocol rtsp_demuxer"
+avserver_deps="avformat ffm_muxer fork rtp_protocol rtsp_demuxer !shared"
 avserver_extralibs='$ldl'
 ffmpeg_deps="avcodec avformat swscale"