Merge remote branch 'qatar/master'

[ffmpeg] / libavformat / rtspenc.c

diff --git a/libavformat/rtspenc.c b/libavformat/rtspenc.c
index 1f1fd32fc106a7547a93e53b1cc084a6df58a84e..1cbcb4975033d6e17372bb424162e5bddc4fcf20 100644 (file)
--- a/libavformat/rtspenc.c
+++ b/libavformat/rtspenc.c
@@ -2,20 +2,20 @@
  * RTSP muxer
  * Copyright (c) 2010 Martin Storsjo
  *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
  *
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -29,8 +29,10 @@
 #include "os_support.h"
 #include "rtsp.h"
 #include "internal.h"
+#include "avio_internal.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/avstring.h"
+#include "url.h"
 
 #define SDP_MAX_SIZE 16384
 
@@ -64,7 +66,7 @@ int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
     ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
                 "rtsp", NULL, addr, -1, NULL);
     ctx_array[0] = &sdp_ctx;
-    if (avf_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
+    if (av_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
         av_free(sdp);
         return AVERROR_INVALIDDATA;
     }
@@ -135,14 +137,14 @@ static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
     int size;
     uint8_t *interleave_header, *interleaved_packet;
 
-    size = url_close_dyn_buf(rtpctx->pb, &buf);
+    size = avio_close_dyn_buf(rtpctx->pb, &buf);
     ptr = buf;
     while (size > 4) {
         uint32_t packet_len = AV_RB32(ptr);
         int id;
         /* The interleaving header is exactly 4 bytes, which happens to be
          * the same size as the packet length header from
-         * url_open_dyn_packet_buf. So by writing the interleaving header
+         * ffio_open_dyn_packet_buf. So by writing the interleaving header
          * over these bytes, we get a consecutive interleaved packet
          * that can be written in one call. */
         interleaved_packet = interleave_header = ptr;
@@ -157,12 +159,12 @@ static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
         interleave_header[0] = '$';
         interleave_header[1] = id;
         AV_WB16(interleave_header + 2, packet_len);
-        url_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);
+        ffurl_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);
         ptr += packet_len;
         size -= packet_len;
     }
     av_free(buf);
-    url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
+    ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
     return 0;
 }
 
@@ -171,7 +173,7 @@ static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)
     RTSPState *rt = s->priv_data;
     RTSPStream *rtsp_st;
     int n;
-    struct pollfd p = {url_get_file_handle(rt->rtsp_hd), POLLIN, 0};
+    struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0};
     AVFormatContext *rtpctx;
     int ret;