[FFmpeg-cvslog] rtpdec: Add ff_ prefix to all nonstatic symbols
Martin Storsjö
git at videolan.org
Thu Oct 13 06:01:45 CEST 2011
ffmpeg | branch: master | Martin Storsjö <martin at martin.st> | Wed Oct 12 12:37:42 2011 +0300| [bfc6db4477cd1ca6c32ab533783238cf8381f177] | committer: Martin Storsjö
rtpdec: Add ff_ prefix to all nonstatic symbols
Signed-off-by: Martin Storsjö <martin at martin.st>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=bfc6db4477cd1ca6c32ab533783238cf8381f177
---
avserver.c | 4 ++--
libavformat/rtpdec.c | 16 ++++++++--------
libavformat/rtpdec.h | 24 ++++++++++++------------
libavformat/rtpproto.c | 8 ++++----
libavformat/rtsp.c | 26 +++++++++++++-------------
5 files changed, 39 insertions(+), 39 deletions(-)
diff --git a/avserver.c b/avserver.c
index 7b8bf13..ff7a15c 100644
--- a/avserver.c
+++ b/avserver.c
@@ -3169,8 +3169,8 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
switch(rtp_c->rtp_protocol) {
case RTSP_LOWER_TRANSPORT_UDP:
- rtp_port = rtp_get_local_rtp_port(rtp_c->rtp_handles[stream_index]);
- rtcp_port = rtp_get_local_rtcp_port(rtp_c->rtp_handles[stream_index]);
+ rtp_port = ff_rtp_get_local_rtp_port(rtp_c->rtp_handles[stream_index]);
+ rtcp_port = ff_rtp_get_local_rtcp_port(rtp_c->rtp_handles[stream_index]);
avio_printf(c->pb, "Transport: RTP/AVP/UDP;unicast;"
"client_port=%d-%d;server_port=%d-%d",
th->client_port_min, th->client_port_max,
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. */
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