From 90abbdba1e7f9df2c32bdb0a9b7c68297cd135c8 Mon Sep 17 00:00:00 2001 From: "Ronald S. Bultje" Date: Tue, 30 Sep 2008 13:18:41 +0000 Subject: Rename RTSPProtocol to RTSPLowerTransport, so that its name properly tells us that it only describes the lower-level transport (TCP vs. UDP) and not the actual data layout (e.g. RDT vs. RTP). See discussion in "Realmedia patch" thread on ML. Originally committed as revision 15481 to svn://svn.ffmpeg.org/ffmpeg/trunk --- ffserver.c | 40 ++++++++++++++++++++-------------------- 1 file changed, 20 insertions(+), 20 deletions(-) (limited to 'ffserver.c') diff --git a/ffserver.c b/ffserver.c index 89836c6f2a..cdfb277e5c 100644 --- a/ffserver.c +++ b/ffserver.c @@ -157,7 +157,7 @@ typedef struct HTTPContext { int seq; /* RTSP sequence number */ /* RTP state specific */ - enum RTSPProtocol rtp_protocol; + enum RTSPLowerTransport rtp_protocol; char session_id[32]; /* session id */ AVFormatContext *rtp_ctx[MAX_STREAMS]; @@ -278,7 +278,7 @@ static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer, /* RTP handling */ static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr, FFStream *stream, const char *session_id, - enum RTSPProtocol rtp_protocol); + enum RTSPLowerTransport rtp_protocol); static int rtp_new_av_stream(HTTPContext *c, int stream_index, struct sockaddr_in *dest_addr, HTTPContext *rtsp_c); @@ -509,7 +509,7 @@ static void start_multicast(void) dest_addr.sin_port = htons(stream->multicast_port); rtp_c = rtp_new_connection(&dest_addr, stream, session_id, - RTSP_PROTOCOL_RTP_UDP_MULTICAST); + RTSP_LOWER_TRANSPORT_UDP_MULTICAST); if (!rtp_c) continue; @@ -2202,7 +2202,7 @@ static int http_prepare_data(HTTPContext *c) if (c->is_packetized) { int max_packet_size; - if (c->rtp_protocol == RTSP_PROTOCOL_RTP_TCP) + if (c->rtp_protocol == RTSP_LOWER_TRANSPORT_TCP) max_packet_size = RTSP_TCP_MAX_PACKET_SIZE; else max_packet_size = url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]); @@ -2306,7 +2306,7 @@ static int http_send_data(HTTPContext *c) if (c->stream) c->stream->bytes_served += len; - if (c->rtp_protocol == RTSP_PROTOCOL_RTP_TCP) { + if (c->rtp_protocol == RTSP_LOWER_TRANSPORT_TCP) { /* RTP packets are sent inside the RTSP TCP connection */ ByteIOContext *pb; int interleaved_index, size; @@ -2798,14 +2798,14 @@ static HTTPContext *find_rtp_session(const char *session_id) return NULL; } -static RTSPTransportField *find_transport(RTSPHeader *h, enum RTSPProtocol protocol) +static RTSPTransportField *find_transport(RTSPHeader *h, enum RTSPLowerTransport lower_transport) { RTSPTransportField *th; int i; for(i=0;inb_transports;i++) { th = &h->transports[i]; - if (th->protocol == protocol) + if (th->lower_transport == lower_transport) return th; } return NULL; @@ -2867,9 +2867,9 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url, rtp_c = find_rtp_session(h->session_id); if (!rtp_c) { /* always prefer UDP */ - th = find_transport(h, RTSP_PROTOCOL_RTP_UDP); + th = find_transport(h, RTSP_LOWER_TRANSPORT_UDP); if (!th) { - th = find_transport(h, RTSP_PROTOCOL_RTP_TCP); + th = find_transport(h, RTSP_LOWER_TRANSPORT_TCP); if (!th) { rtsp_reply_error(c, RTSP_STATUS_TRANSPORT); return; @@ -2877,7 +2877,7 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url, } rtp_c = rtp_new_connection(&c->from_addr, stream, h->session_id, - th->protocol); + th->lower_transport); if (!rtp_c) { rtsp_reply_error(c, RTSP_STATUS_BANDWIDTH); return; @@ -2905,7 +2905,7 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url, /* check transport */ th = find_transport(h, rtp_c->rtp_protocol); - if (!th || (th->protocol == RTSP_PROTOCOL_RTP_UDP && + if (!th || (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP && th->client_port_min <= 0)) { rtsp_reply_error(c, RTSP_STATUS_TRANSPORT); return; @@ -2928,14 +2928,14 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url, url_fprintf(c->pb, "Session: %s\r\n", rtp_c->session_id); switch(rtp_c->rtp_protocol) { - case RTSP_PROTOCOL_RTP_UDP: + case RTSP_LOWER_TRANSPORT_UDP: port = rtp_get_local_port(rtp_c->rtp_handles[stream_index]); url_fprintf(c->pb, "Transport: RTP/AVP/UDP;unicast;" "client_port=%d-%d;server_port=%d-%d", th->client_port_min, th->client_port_min + 1, port, port + 1); break; - case RTSP_PROTOCOL_RTP_TCP: + case RTSP_LOWER_TRANSPORT_TCP: url_fprintf(c->pb, "Transport: RTP/AVP/TCP;interleaved=%d-%d", stream_index * 2, stream_index * 2 + 1); break; @@ -3071,7 +3071,7 @@ static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPHeader *h) static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr, FFStream *stream, const char *session_id, - enum RTSPProtocol rtp_protocol) + enum RTSPLowerTransport rtp_protocol) { HTTPContext *c = NULL; const char *proto_str; @@ -3102,13 +3102,13 @@ static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr, /* protocol is shown in statistics */ switch(c->rtp_protocol) { - case RTSP_PROTOCOL_RTP_UDP_MULTICAST: + case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: proto_str = "MCAST"; break; - case RTSP_PROTOCOL_RTP_UDP: + case RTSP_LOWER_TRANSPORT_UDP: proto_str = "UDP"; break; - case RTSP_PROTOCOL_RTP_TCP: + case RTSP_LOWER_TRANSPORT_TCP: proto_str = "TCP"; break; default: @@ -3172,8 +3172,8 @@ static int rtp_new_av_stream(HTTPContext *c, ipaddr = inet_ntoa(dest_addr->sin_addr); switch(c->rtp_protocol) { - case RTSP_PROTOCOL_RTP_UDP: - case RTSP_PROTOCOL_RTP_UDP_MULTICAST: + case RTSP_LOWER_TRANSPORT_UDP: + case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: /* RTP/UDP case */ /* XXX: also pass as parameter to function ? */ @@ -3195,7 +3195,7 @@ static int rtp_new_av_stream(HTTPContext *c, c->rtp_handles[stream_index] = h; max_packet_size = url_get_max_packet_size(h); break; - case RTSP_PROTOCOL_RTP_TCP: + case RTSP_LOWER_TRANSPORT_TCP: /* RTP/TCP case */ c->rtsp_c = rtsp_c; max_packet_size = RTSP_TCP_MAX_PACKET_SIZE; -- cgit v1.2.3