X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtpdec.c;h=a910cf08bb0bcb73530c30a2de0b0b306844ff24;hb=57b4a3dd2b358b2122736af861c1538acd1eed1a;hp=3d03b815c2a2e64c1a30b4fa9f0730145f5751dc;hpb=b92c5452822f9f58d33daf933a0d2a5516866bc1;p=ffmpeg diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c index 3d03b815c2a..a910cf08bb0 100644 --- a/libavformat/rtpdec.c +++ b/libavformat/rtpdec.c @@ -19,12 +19,10 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -/* needed for gethostname() */ -#define _XOPEN_SOURCE 600 - #include "libavcodec/get_bits.h" #include "avformat.h" #include "mpegts.h" +#include "url.h" #include #include @@ -41,7 +39,7 @@ buffer to 'rtp_write_packet' contains all the packets for ONE frame. Each packet should have a four byte header containing the length in big endian format (same trick as - 'url_open_dyn_packet_buf') + 'ffio_open_dyn_packet_buf') */ static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = { @@ -321,12 +319,12 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) } avio_flush(pb); - len = url_close_dyn_buf(pb, &buf); + len = avio_close_dyn_buf(pb, &buf); if ((len > 0) && buf) { - int result; + int av_unused result; av_dlog(s->ic, "sending %d bytes of RR\n", len); - result= url_write(s->rtp_ctx, buf, len); - av_dlog(s->ic, "result from url_write: %d\n", result); + result= ffurl_write(s->rtp_ctx, buf, len); + av_dlog(s->ic, "result from ffurl_write: %d\n", result); av_free(buf); } return 0; @@ -349,9 +347,9 @@ void rtp_send_punch_packets(URLContext* rtp_handle) avio_wb32(pb, 0); /* SSRC */ avio_flush(pb); - len = url_close_dyn_buf(pb, &buf); + len = avio_close_dyn_buf(pb, &buf); if ((len > 0) && buf) - url_write(rtp_handle, buf, len); + ffurl_write(rtp_handle, buf, len); av_free(buf); /* Send a minimal RTCP RR */ @@ -364,9 +362,9 @@ void rtp_send_punch_packets(URLContext* rtp_handle) avio_wb32(pb, 0); /* our own SSRC */ avio_flush(pb); - len = url_close_dyn_buf(pb, &buf); + len = avio_close_dyn_buf(pb, &buf); if ((len > 0) && buf) - url_write(rtp_handle, buf, len); + ffurl_write(rtp_handle, buf, len); av_free(buf); }