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];
/* 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);
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;
return -1;
}
- http_log("ffserver started.\n");
+ http_log("FFserver started.\n");
start_children(first_feed);
ctx = &c->fmt_ctx;
- if (!c->last_packet_sent) {
+ if (!c->last_packet_sent && c->state == HTTPSTATE_SEND_DATA_TRAILER) {
if (ctx->oformat) {
/* prepare header */
if (url_open_dyn_buf(&ctx->pb) >= 0) {
/* If this is WMP, get the rate information */
if (extract_rates(ratebuf, sizeof(ratebuf), c->buffer)) {
if (modify_current_stream(c, ratebuf)) {
- for (i = 0; i < sizeof(c->feed_streams) / sizeof(c->feed_streams[0]); i++) {
+ for (i = 0; i < FF_ARRAY_ELEMS(c->feed_streams); i++) {
if (c->switch_feed_streams[i] >= 0)
do_switch_stream(c, i);
}
"\r\n"
"<html><head><title>Too busy</title></head><body>\r\n"
"<p>The server is too busy to serve your request at this time.</p>\r\n"
- "<p>The bandwidth being served (including your stream) is %lldkbit/sec, "
- "and this exceeds the limit of %lldkbit/sec.</p>\r\n"
+ "<p>The bandwidth being served (including your stream) is %"PRIu64"kbit/sec, "
+ "and this exceeds the limit of %"PRIu64"kbit/sec.</p>\r\n"
"</body></html>\r\n", current_bandwidth, max_bandwidth);
/* prepare output buffer */
c->buffer_ptr = c->buffer;
url_fprintf(pb, "Number of connections: %d / %d<BR>\n",
nb_connections, nb_max_connections);
- url_fprintf(pb, "Bandwidth in use: %lldk / %lldk<BR>\n",
+ url_fprintf(pb, "Bandwidth in use: %"PRIu64"k / %"PRIu64"k<BR>\n",
current_bandwidth, max_bandwidth);
url_fprintf(pb, "<TABLE>\n");
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]);
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;
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;
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;
}
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;
/* 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;
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;
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;
/* 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:
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 ? */
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;
memcpy(st->codec, av, sizeof(AVCodecContext));
}
-static int opt_audio_codec(const char *arg)
+static enum CodecID opt_audio_codec(const char *arg)
{
AVCodec *p= avcodec_find_encoder_by_name(arg);
return p->id;
}
-static int opt_video_codec(const char *arg)
+static enum CodecID opt_video_codec(const char *arg)
{
AVCodec *p= avcodec_find_encoder_by_name(arg);
}
#endif
-static int opt_default(const char *opt, const char *arg,
+static int ffserver_opt_default(const char *opt, const char *arg,
AVCodecContext *avctx, int type)
{
- const AVOption *o = NULL;
- const AVOption *o2 = av_find_opt(avctx, opt, NULL, type, type);
- if(o2)
- o = av_set_string2(avctx, opt, arg, 1);
- if(!o)
- return -1;
- return 0;
+ int ret = 0;
+ const AVOption *o = av_find_opt(avctx, opt, NULL, type, type);
+ if(o)
+ ret = av_set_string3(avctx, opt, arg, 1, NULL);
+ return ret;
}
static int parse_ffconfig(const char *filename)
FFStream **last_stream, *stream, *redirect;
FFStream **last_feed, *feed;
AVCodecContext audio_enc, video_enc;
- int audio_id, video_id;
+ enum CodecID audio_id, video_id;
f = fopen(filename, "r");
if (!f) {
}
} else if (!strcasecmp(cmd, "InputFormat")) {
get_arg(arg, sizeof(arg), &p);
- stream->ifmt = av_find_input_format(arg);
- if (!stream->ifmt) {
- fprintf(stderr, "%s:%d: Unknown input format: %s\n",
- filename, line_num, arg);
+ if (stream) {
+ stream->ifmt = av_find_input_format(arg);
+ if (!stream->ifmt) {
+ fprintf(stderr, "%s:%d: Unknown input format: %s\n",
+ filename, line_num, arg);
+ }
}
} else if (!strcasecmp(cmd, "FaviconURL")) {
if (stream && stream->stream_type == STREAM_TYPE_STATUS) {
avctx = &audio_enc;
type = AV_OPT_FLAG_AUDIO_PARAM;
}
- if (opt_default(arg, arg2, avctx, type|AV_OPT_FLAG_ENCODING_PARAM)) {
+ if (ffserver_opt_default(arg, arg2, avctx, type|AV_OPT_FLAG_ENCODING_PARAM)) {
fprintf(stderr, "AVOption error: %s %s\n", arg, arg2);
errors++;
}
projects / ffmpeg / blobdiff