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Rename RTSPProtocol to RTSPLowerTransport, so that its name properly tells us

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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
This commit is contained in:
Ronald S. Bultje
2008-09-30 13:18:41 +00:00
parent e1a96237df
commit 90abbdba1e
3 changed files with 59 additions and 59 deletions
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40
ffserver.c
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@@ -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;i<h->nb_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;