Merge remote-tracking branch 'qatar/master'

* qatar/master:
  rtpdec: Use our own SSRC in the SDES field when sending RRs
  Finalize changelog for 0.8 Release
  Prepare for 0.8 Release
  threads: change the default for threads back to 1
  threads: update slice_count and slice_offset from user context
  aviocat: Remove useless includes
  doc/APIChanges: fill in missing dates and hashes
  Revert "avserver: fix build after the next bump."
  mpegaudiodec: switch error detection check to AV_EF_BUFFER
  lavf: rename fer option and document resulting (f_)err_detect options
  lavc: rename err_filter option to err_detect and document it
  mpegvideo: fix invalid memory access for small video dimensions
  movenc: Reorder entries in the MOVIentry struct, for tigheter packing
  rtsp: Remove extern declarations for variables that don't exist
  aviocat: Flush the output before closing

Conflicts:
	Changelog
	RELEASE
	libavcodec/mpegaudiodec.c
	libavcodec/pthread.c
	libavformat/options.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 10 insertions, 12 deletions

diff --git a/ffserver.c b/ffserver.c
index 79463c0e64..c225a94e3b 100644
--- a/ffserver.c
+++ b/ffserver.c
@@ -30,16 +30,13 @@
 #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"
@@ -877,7 +874,7 @@ static void close_connection(HTTPContext *c)
         }
         h = c->rtp_handles[i];
         if (h)
-            ffurl_close(h);
+            url_close(h);
     }
 
     ctx = &c->fmt_ctx;
@@ -2258,6 +2255,7 @@ 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) {
@@ -2376,7 +2374,7 @@ 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 = c->rtp_handles[c->packet_stream_index]->max_packet_size;
+                            max_packet_size = url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]);
                         ret = ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size);
                     } else {
                         ret = avio_open_dyn_buf(&ctx->pb);
@@ -2529,8 +2527,8 @@ static int http_send_data(HTTPContext *c)
                 } else {
                     /* send RTP packet directly in UDP */
                     c->buffer_ptr += 4;
-                    ffurl_write(c->rtp_handles[c->packet_stream_index],
-                                c->buffer_ptr, len);
+                    url_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 */
                 }
@@ -3413,10 +3411,10 @@ static int rtp_new_av_stream(HTTPContext *c,
                      "rtp://%s:%d", ipaddr, ntohs(dest_addr->sin_port));
         }
 
-        if (ffurl_open(&h, ctx->filename, AVIO_FLAG_WRITE, NULL, NULL) < 0)
+        if (url_open(&h, ctx->filename, AVIO_FLAG_WRITE) < 0)
             goto fail;
         c->rtp_handles[stream_index] = h;
-        max_packet_size = h->max_packet_size;
+        max_packet_size = url_get_max_packet_size(h);
         break;
     case RTSP_LOWER_TRANSPORT_TCP:
         /* RTP/TCP case */
@@ -3439,7 +3437,7 @@ static int rtp_new_av_stream(HTTPContext *c,
     if (avformat_write_header(ctx, NULL) < 0) {
     fail:
         if (h)
-            ffurl_close(h);
+            url_close(h);
         av_free(ctx);
         return -1;
     }
@@ -3476,7 +3474,7 @@ static AVStream *add_av_stream1(FFStream *stream, AVCodecContext *codec, int cop
     }
     fst->priv_data = av_mallocz(sizeof(FeedData));
     fst->index = stream->nb_streams;
-    avpriv_set_pts_info(fst, 33, 1, 90000);
+    av_set_pts_info(fst, 33, 1, 90000);
     fst->sample_aspect_ratio = codec->sample_aspect_ratio;
     stream->streams[stream->nb_streams++] = fst;
     return fst;