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https://github.com/FFmpeg/FFmpeg.git
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rtpdec: Add ff_ prefix to all nonstatic symbols
Signed-off-by: Martin Storsjö <martin@martin.st>
This commit is contained in:
parent
318efbfc10
commit
bfc6db4477
@ -3169,8 +3169,8 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
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switch(rtp_c->rtp_protocol) {
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case RTSP_LOWER_TRANSPORT_UDP:
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rtp_port = rtp_get_local_rtp_port(rtp_c->rtp_handles[stream_index]);
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rtcp_port = rtp_get_local_rtcp_port(rtp_c->rtp_handles[stream_index]);
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rtp_port = ff_rtp_get_local_rtp_port(rtp_c->rtp_handles[stream_index]);
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rtcp_port = ff_rtp_get_local_rtcp_port(rtp_c->rtp_handles[stream_index]);
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avio_printf(c->pb, "Transport: RTP/AVP/UDP;unicast;"
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"client_port=%d-%d;server_port=%d-%d",
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th->client_port_min, th->client_port_max,
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@ -218,7 +218,7 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
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return 1;
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}
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int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
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int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
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{
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AVIOContext *pb;
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uint8_t *buf;
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@ -315,7 +315,7 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
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return 0;
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}
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void rtp_send_punch_packets(URLContext* rtp_handle)
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void ff_rtp_send_punch_packets(URLContext* rtp_handle)
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{
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AVIOContext *pb;
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uint8_t *buf;
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@ -359,7 +359,7 @@ void rtp_send_punch_packets(URLContext* rtp_handle)
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* MPEG2TS streams to indicate that they should be demuxed inside the
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* rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
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*/
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RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
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RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
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{
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RTPDemuxContext *s;
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@ -407,8 +407,8 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r
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}
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void
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rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
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RTPDynamicProtocolHandler *handler)
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ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
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RTPDynamicProtocolHandler *handler)
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{
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s->dynamic_protocol_context = ctx;
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s->parse_packet = handler->parse_packet;
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@ -722,8 +722,8 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
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* @return 0 if a packet is returned, 1 if a packet is returned and more can follow
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* (use buf as NULL to read the next). -1 if no packet (error or no more packet).
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*/
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int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
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uint8_t **bufptr, int len)
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int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
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uint8_t **bufptr, int len)
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{
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int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
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s->prev_ret = rv;
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@ -732,7 +732,7 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
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return rv ? rv : has_next_packet(s);
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}
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void rtp_parse_close(RTPDemuxContext *s)
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void ff_rtp_parse_close(RTPDemuxContext *s)
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{
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ff_rtp_reset_packet_queue(s);
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if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
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@ -38,18 +38,18 @@ typedef struct RTPDynamicProtocolHandler_s RTPDynamicProtocolHandler;
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#define RTP_NOTS_VALUE ((uint32_t)-1)
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typedef struct RTPDemuxContext RTPDemuxContext;
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RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size);
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void rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
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RTPDynamicProtocolHandler *handler);
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int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
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uint8_t **buf, int len);
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void rtp_parse_close(RTPDemuxContext *s);
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RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size);
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void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
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RTPDynamicProtocolHandler *handler);
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int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
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uint8_t **buf, int len);
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void ff_rtp_parse_close(RTPDemuxContext *s);
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int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s);
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void ff_rtp_reset_packet_queue(RTPDemuxContext *s);
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int rtp_get_local_rtp_port(URLContext *h);
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int rtp_get_local_rtcp_port(URLContext *h);
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int ff_rtp_get_local_rtp_port(URLContext *h);
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int ff_rtp_get_local_rtcp_port(URLContext *h);
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int rtp_set_remote_url(URLContext *h, const char *uri);
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int ff_rtp_set_remote_url(URLContext *h, const char *uri);
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/**
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* Send a dummy packet on both port pairs to set up the connection
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@ -62,19 +62,19 @@ int rtp_set_remote_url(URLContext *h, const char *uri);
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* The same routine is used with RDT too, even if RDT doesn't use normal
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* RTP packets otherwise.
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*/
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void rtp_send_punch_packets(URLContext* rtp_handle);
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void ff_rtp_send_punch_packets(URLContext* rtp_handle);
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/**
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* some rtp servers assume client is dead if they don't hear from them...
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* so we send a Receiver Report to the provided ByteIO context
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* (we don't have access to the rtcp handle from here)
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*/
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int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count);
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int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count);
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/**
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* Get the file handle for the RTCP socket.
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*/
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int rtp_get_rtcp_file_handle(URLContext *h);
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int ff_rtp_get_rtcp_file_handle(URLContext *h);
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// these statistics are used for rtcp receiver reports...
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typedef struct {
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@ -60,7 +60,7 @@ typedef struct RTPContext {
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* @return zero if no error.
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*/
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int rtp_set_remote_url(URLContext *h, const char *uri)
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int ff_rtp_set_remote_url(URLContext *h, const char *uri)
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{
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RTPContext *s = h->priv_data;
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char hostname[256];
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@ -300,7 +300,7 @@ static int rtp_close(URLContext *h)
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* @return the local port number
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*/
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int rtp_get_local_rtp_port(URLContext *h)
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int ff_rtp_get_local_rtp_port(URLContext *h)
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{
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RTPContext *s = h->priv_data;
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return ff_udp_get_local_port(s->rtp_hd);
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@ -312,7 +312,7 @@ int rtp_get_local_rtp_port(URLContext *h)
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* @return the local port number
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*/
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int rtp_get_local_rtcp_port(URLContext *h)
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int ff_rtp_get_local_rtcp_port(URLContext *h)
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{
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RTPContext *s = h->priv_data;
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return ff_udp_get_local_port(s->rtcp_hd);
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@ -324,7 +324,7 @@ static int rtp_get_file_handle(URLContext *h)
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return s->rtp_fd;
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}
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int rtp_get_rtcp_file_handle(URLContext *h) {
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int ff_rtp_get_rtcp_file_handle(URLContext *h) {
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RTPContext *s = h->priv_data;
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return s->rtcp_fd;
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}
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@ -501,7 +501,7 @@ void ff_rtsp_undo_setup(AVFormatContext *s)
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} else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
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ff_rdt_parse_close(rtsp_st->transport_priv);
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else if (CONFIG_RTPDEC)
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rtp_parse_close(rtsp_st->transport_priv);
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ff_rtp_parse_close(rtsp_st->transport_priv);
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}
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rtsp_st->transport_priv = NULL;
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if (rtsp_st->rtp_handle)
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@ -558,7 +558,7 @@ static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
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rtsp_st->dynamic_protocol_context,
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rtsp_st->dynamic_handler);
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else if (CONFIG_RTPDEC)
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rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
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rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
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rtsp_st->sdp_payload_type,
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(rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
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? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
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@ -567,9 +567,9 @@ static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
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return AVERROR(ENOMEM);
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} else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
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if (rtsp_st->dynamic_handler) {
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rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
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rtsp_st->dynamic_protocol_context,
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rtsp_st->dynamic_handler);
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ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
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rtsp_st->dynamic_protocol_context,
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rtsp_st->dynamic_handler);
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}
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}
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@ -1121,7 +1121,7 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
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goto fail;
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rtp_opened:
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port = rtp_get_local_rtp_port(rtsp_st->rtp_handle);
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port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
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have_port:
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snprintf(transport, sizeof(transport) - 1,
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"%s/UDP;", trans_pref);
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@ -1225,7 +1225,7 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
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reply->transports[0].server_port_min, "%s", options);
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}
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if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
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rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
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ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
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err = AVERROR_INVALIDDATA;
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goto fail;
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}
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@ -1235,7 +1235,7 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
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*/
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if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
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CONFIG_RTPDEC)
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rtp_send_punch_packets(rtsp_st->rtp_handle);
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ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
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break;
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}
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case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
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@ -1569,7 +1569,7 @@ static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
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if (rtsp_st->rtp_handle) {
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p[max_p].fd = ffurl_get_file_handle(rtsp_st->rtp_handle);
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p[max_p++].events = POLLIN;
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p[max_p].fd = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
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p[max_p].fd = ff_rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
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p[max_p++].events = POLLIN;
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}
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}
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@ -1624,7 +1624,7 @@ int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
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if (rt->transport == RTSP_TRANSPORT_RDT) {
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ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
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} else
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ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
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ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
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if (ret == 0) {
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rt->cur_transport_priv = NULL;
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return 0;
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@ -1672,13 +1672,13 @@ int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
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case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
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len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
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if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
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rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
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ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
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break;
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}
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if (len == AVERROR(EAGAIN) && first_queue_st &&
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rt->transport == RTSP_TRANSPORT_RTP) {
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rtsp_st = first_queue_st;
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ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
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ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
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goto end;
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}
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if (len < 0)
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@ -1688,7 +1688,7 @@ int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
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if (rt->transport == RTSP_TRANSPORT_RDT) {
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ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
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} else {
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ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
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ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
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if (ret < 0) {
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/* Either bad packet, or a RTCP packet. Check if the
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* first_rtcp_ntp_time field was initialized. */
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