]> git.sesse.net Git - ffmpeg/blobdiff - ffserver.c
Merge remote-tracking branch 'qatar/master'
[ffmpeg] / ffserver.c
index c225a94e3b2bdbf47131674c57b5d441af668d14..79463c0e64981927c5a6025ea7751bde6fa77f88 100644 (file)
 #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"
@@ -874,7 +877,7 @@ static void close_connection(HTTPContext *c)
         }
         h = c->rtp_handles[i];
         if (h)
-            url_close(h);
+            ffurl_close(h);
     }
 
     ctx = &c->fmt_ctx;
@@ -2255,7 +2258,6 @@ static int http_prepare_data(HTTPContext *c)
          * Default value from FFmpeg
          * 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) {
@@ -2374,7 +2376,7 @@ 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);
@@ -2527,8 +2529,8 @@ 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 */
                 }
@@ -3411,10 +3413,10 @@ 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 */
@@ -3437,7 +3439,7 @@ 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;
     }
@@ -3474,7 +3476,7 @@ 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;