rtpdec: Add ff_ prefix to all nonstatic symbols

Signed-off-by: Martin Storsjö <martin@martin.st>

4 files changed, 37 insertions, 37 deletions

diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 0f6ed27..92535ec 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -218,7 +218,7 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
     return 1;
 }
 
-int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
+int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
 {
     AVIOContext *pb;
     uint8_t *buf;
@@ -315,7 +315,7 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
     return 0;
 }
 
-void rtp_send_punch_packets(URLContext* rtp_handle)
+void ff_rtp_send_punch_packets(URLContext* rtp_handle)
 {
     AVIOContext *pb;
     uint8_t *buf;
@@ -359,7 +359,7 @@ void rtp_send_punch_packets(URLContext* rtp_handle)
  * MPEG2TS streams to indicate that they should be demuxed inside the
  * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
  */
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
+RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
 {
     RTPDemuxContext *s;
 
@@ -407,8 +407,8 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r
 }
 
 void
-rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
-                               RTPDynamicProtocolHandler *handler)
+ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
+                                  RTPDynamicProtocolHandler *handler)
 {
     s->dynamic_protocol_context = ctx;
     s->parse_packet = handler->parse_packet;
@@ -722,8 +722,8 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
  * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
  * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
  */
-int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
-                     uint8_t **bufptr, int len)
+int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
+                        uint8_t **bufptr, int len)
 {
     int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
     s->prev_ret = rv;
@@ -732,7 +732,7 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
     return rv ? rv : has_next_packet(s);
 }
 
-void rtp_parse_close(RTPDemuxContext *s)
+void ff_rtp_parse_close(RTPDemuxContext *s)
 {
     ff_rtp_reset_packet_queue(s);
     if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
diff --git a/libavformat/rtpdec.h b/libavformat/rtpdec.h
index a4d21aa..d58eddd 100644
--- a/libavformat/rtpdec.h
+++ b/libavformat/rtpdec.h
@@ -38,18 +38,18 @@ typedef struct RTPDynamicProtocolHandler_s RTPDynamicProtocolHandler;
 #define RTP_NOTS_VALUE ((uint32_t)-1)
 
 typedef struct RTPDemuxContext RTPDemuxContext;
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size);
-void rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
-                                    RTPDynamicProtocolHandler *handler);
-int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
-                     uint8_t **buf, int len);
-void rtp_parse_close(RTPDemuxContext *s);
+RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size);
+void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
+                                       RTPDynamicProtocolHandler *handler);
+int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
+                        uint8_t **buf, int len);
+void ff_rtp_parse_close(RTPDemuxContext *s);
 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s);
 void ff_rtp_reset_packet_queue(RTPDemuxContext *s);
-int rtp_get_local_rtp_port(URLContext *h);
-int rtp_get_local_rtcp_port(URLContext *h);
+int ff_rtp_get_local_rtp_port(URLContext *h);
+int ff_rtp_get_local_rtcp_port(URLContext *h);
 
-int rtp_set_remote_url(URLContext *h, const char *uri);
+int ff_rtp_set_remote_url(URLContext *h, const char *uri);
 
 /**
  * Send a dummy packet on both port pairs to set up the connection
@@ -62,19 +62,19 @@ int rtp_set_remote_url(URLContext *h, const char *uri);
  * The same routine is used with RDT too, even if RDT doesn't use normal
  * RTP packets otherwise.
  */
-void rtp_send_punch_packets(URLContext* rtp_handle);
+void ff_rtp_send_punch_packets(URLContext* rtp_handle);
 
 /**
  * some rtp servers assume client is dead if they don't hear from them...
  * so we send a Receiver Report to the provided ByteIO context
  * (we don't have access to the rtcp handle from here)
  */
-int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count);
+int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count);
 
 /**
  * Get the file handle for the RTCP socket.
  */
-int rtp_get_rtcp_file_handle(URLContext *h);
+int ff_rtp_get_rtcp_file_handle(URLContext *h);
 
 // these statistics are used for rtcp receiver reports...
 typedef struct {
diff --git a/libavformat/rtpproto.c b/libavformat/rtpproto.c
index 0367198..9a18157 100644
--- a/libavformat/rtpproto.c
+++ b/libavformat/rtpproto.c
@@ -60,7 +60,7 @@ typedef struct RTPContext {
  * @return zero if no error.
  */
 
-int rtp_set_remote_url(URLContext *h, const char *uri)
+int ff_rtp_set_remote_url(URLContext *h, const char *uri)
 {
     RTPContext *s = h->priv_data;
     char hostname[256];
@@ -300,7 +300,7 @@ static int rtp_close(URLContext *h)
  * @return the local port number
  */
 
-int rtp_get_local_rtp_port(URLContext *h)
+int ff_rtp_get_local_rtp_port(URLContext *h)
 {
     RTPContext *s = h->priv_data;
     return ff_udp_get_local_port(s->rtp_hd);
@@ -312,7 +312,7 @@ int rtp_get_local_rtp_port(URLContext *h)
  * @return the local port number
  */
 
-int rtp_get_local_rtcp_port(URLContext *h)
+int ff_rtp_get_local_rtcp_port(URLContext *h)
 {
     RTPContext *s = h->priv_data;
     return ff_udp_get_local_port(s->rtcp_hd);
@@ -324,7 +324,7 @@ static int rtp_get_file_handle(URLContext *h)
     return s->rtp_fd;
 }
 
-int rtp_get_rtcp_file_handle(URLContext *h) {
+int ff_rtp_get_rtcp_file_handle(URLContext *h) {
     RTPContext *s = h->priv_data;
     return s->rtcp_fd;
 }
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 8b70c8b..ff4d16a 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -501,7 +501,7 @@ void ff_rtsp_undo_setup(AVFormatContext *s)
             } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
                 ff_rdt_parse_close(rtsp_st->transport_priv);
             else if (CONFIG_RTPDEC)
-                rtp_parse_close(rtsp_st->transport_priv);
+                ff_rtp_parse_close(rtsp_st->transport_priv);
         }
         rtsp_st->transport_priv = NULL;
         if (rtsp_st->rtp_handle)
@@ -558,7 +558,7 @@ static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
                                             rtsp_st->dynamic_protocol_context,
                                             rtsp_st->dynamic_handler);
     else if (CONFIG_RTPDEC)
-        rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
+        rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
                                          rtsp_st->sdp_payload_type,
             (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
             ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
@@ -567,9 +567,9 @@ static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
          return AVERROR(ENOMEM);
     } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
         if (rtsp_st->dynamic_handler) {
-            rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
-                                           rtsp_st->dynamic_protocol_context,
-                                           rtsp_st->dynamic_handler);
+            ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
+                                              rtsp_st->dynamic_protocol_context,
+                                              rtsp_st->dynamic_handler);
         }
     }
 
@@ -1121,7 +1121,7 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
             goto fail;
 
         rtp_opened:
-            port = rtp_get_local_rtp_port(rtsp_st->rtp_handle);
+            port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
         have_port:
             snprintf(transport, sizeof(transport) - 1,
                      "%s/UDP;", trans_pref);
@@ -1225,7 +1225,7 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
                             reply->transports[0].server_port_min, "%s", options);
             }
             if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
-                rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
+                ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
                 err = AVERROR_INVALIDDATA;
                 goto fail;
             }
@@ -1235,7 +1235,7 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
              */
             if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
                 CONFIG_RTPDEC)
-                rtp_send_punch_packets(rtsp_st->rtp_handle);
+                ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
             break;
         }
         case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
@@ -1569,7 +1569,7 @@ static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
             if (rtsp_st->rtp_handle) {
                 p[max_p].fd = ffurl_get_file_handle(rtsp_st->rtp_handle);
                 p[max_p++].events = POLLIN;
-                p[max_p].fd = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
+                p[max_p].fd = ff_rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
                 p[max_p++].events = POLLIN;
             }
         }
@@ -1624,7 +1624,7 @@ int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
         if (rt->transport == RTSP_TRANSPORT_RDT) {
             ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
         } else
-            ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
+            ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
         if (ret == 0) {
             rt->cur_transport_priv = NULL;
             return 0;
@@ -1672,13 +1672,13 @@ int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
     case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
         len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
         if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
-            rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
+            ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
         break;
     }
     if (len == AVERROR(EAGAIN) && first_queue_st &&
         rt->transport == RTSP_TRANSPORT_RTP) {
         rtsp_st = first_queue_st;
-        ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
+        ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
         goto end;
     }
     if (len < 0)
@@ -1688,7 +1688,7 @@ int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
     if (rt->transport == RTSP_TRANSPORT_RDT) {
         ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
     } else {
-        ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
+        ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
         if (ret < 0) {
             /* Either bad packet, or a RTCP packet. Check if the
              * first_rtcp_ntp_time field was initialized. */