+/********************************************************************/
+/* RTSP handling */
+
+static void rtsp_reply_header(HTTPContext *c, enum RTSPStatusCode error_number)
+{
+ const char *str;
+ time_t ti;
+ char *p;
+ char buf2[32];
+
+ switch(error_number) {
+#define DEF(n, c, s) case c: str = s; break;
+#include "rtspcodes.h"
+#undef DEF
+ default:
+ str = "Unknown Error";
+ break;
+ }
+
+ url_fprintf(c->pb, "RTSP/1.0 %d %s\r\n", error_number, str);
+ url_fprintf(c->pb, "CSeq: %d\r\n", c->seq);
+
+ /* output GMT time */
+ ti = time(NULL);
+ p = ctime(&ti);
+ strcpy(buf2, p);
+ p = buf2 + strlen(p) - 1;
+ if (*p == '\n')
+ *p = '\0';
+ url_fprintf(c->pb, "Date: %s GMT\r\n", buf2);
+}
+
+static void rtsp_reply_error(HTTPContext *c, enum RTSPStatusCode error_number)
+{
+ rtsp_reply_header(c, error_number);
+ url_fprintf(c->pb, "\r\n");
+}
+
+static int rtsp_parse_request(HTTPContext *c)
+{
+ const char *p, *p1, *p2;
+ char cmd[32];
+ char url[1024];
+ char protocol[32];
+ char line[1024];
+ ByteIOContext pb1;
+ int len;
+ RTSPHeader header1, *header = &header1;
+
+ c->buffer_ptr[0] = '\0';
+ p = c->buffer;
+
+ get_word(cmd, sizeof(cmd), &p);
+ get_word(url, sizeof(url), &p);
+ get_word(protocol, sizeof(protocol), &p);
+
+ pstrcpy(c->method, sizeof(c->method), cmd);
+ pstrcpy(c->url, sizeof(c->url), url);
+ pstrcpy(c->protocol, sizeof(c->protocol), protocol);
+
+ c->pb = &pb1;
+ if (url_open_dyn_buf(c->pb) < 0) {
+ /* XXX: cannot do more */
+ c->pb = NULL; /* safety */
+ return -1;
+ }
+
+ /* check version name */
+ if (strcmp(protocol, "RTSP/1.0") != 0) {
+ rtsp_reply_error(c, RTSP_STATUS_VERSION);
+ goto the_end;
+ }
+
+ /* parse each header line */
+ memset(header, 0, sizeof(RTSPHeader));
+ /* skip to next line */
+ while (*p != '\n' && *p != '\0')
+ p++;
+ if (*p == '\n')
+ p++;
+ while (*p != '\0') {
+ p1 = strchr(p, '\n');
+ if (!p1)
+ break;
+ p2 = p1;
+ if (p2 > p && p2[-1] == '\r')
+ p2--;
+ /* skip empty line */
+ if (p2 == p)
+ break;
+ len = p2 - p;
+ if (len > sizeof(line) - 1)
+ len = sizeof(line) - 1;
+ memcpy(line, p, len);
+ line[len] = '\0';
+ rtsp_parse_line(header, line);
+ p = p1 + 1;
+ }
+
+ /* handle sequence number */
+ c->seq = header->seq;
+
+ if (!strcmp(cmd, "DESCRIBE")) {
+ rtsp_cmd_describe(c, url);
+ } else if (!strcmp(cmd, "OPTIONS")) {
+ rtsp_cmd_options(c, url);
+ } else if (!strcmp(cmd, "SETUP")) {
+ rtsp_cmd_setup(c, url, header);
+ } else if (!strcmp(cmd, "PLAY")) {
+ rtsp_cmd_play(c, url, header);
+ } else if (!strcmp(cmd, "PAUSE")) {
+ rtsp_cmd_pause(c, url, header);
+ } else if (!strcmp(cmd, "TEARDOWN")) {
+ rtsp_cmd_teardown(c, url, header);
+ } else {
+ rtsp_reply_error(c, RTSP_STATUS_METHOD);
+ }
+ the_end:
+ len = url_close_dyn_buf(c->pb, &c->pb_buffer);
+ c->pb = NULL; /* safety */
+ if (len < 0) {
+ /* XXX: cannot do more */
+ return -1;
+ }
+ c->buffer_ptr = c->pb_buffer;
+ c->buffer_end = c->pb_buffer + len;
+ c->state = RTSPSTATE_SEND_REPLY;
+ return 0;
+}
+
+/* XXX: move that to rtsp.c, but would need to replace FFStream by
+ AVFormatContext */
+static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer,
+ struct in_addr my_ip)
+{
+ ByteIOContext pb1, *pb = &pb1;
+ int i, payload_type, port, private_payload_type, j;
+ const char *ipstr, *title, *mediatype;
+ AVStream *st;
+
+ if (url_open_dyn_buf(pb) < 0)
+ return -1;
+
+ /* general media info */
+
+ url_fprintf(pb, "v=0\n");
+ ipstr = inet_ntoa(my_ip);
+ url_fprintf(pb, "o=- 0 0 IN IP4 %s\n", ipstr);
+ title = stream->title;
+ if (title[0] == '\0')
+ title = "No Title";
+ url_fprintf(pb, "s=%s\n", title);
+ if (stream->comment[0] != '\0')
+ url_fprintf(pb, "i=%s\n", stream->comment);
+ if (stream->is_multicast) {
+ url_fprintf(pb, "c=IN IP4 %s\n", inet_ntoa(stream->multicast_ip));
+ }
+ /* for each stream, we output the necessary info */
+ private_payload_type = RTP_PT_PRIVATE;
+ for(i = 0; i < stream->nb_streams; i++) {
+ st = stream->streams[i];
+ if (st->codec.codec_id == CODEC_ID_MPEG2TS) {
+ mediatype = "video";
+ } else {
+ switch(st->codec.codec_type) {
+ case CODEC_TYPE_AUDIO:
+ mediatype = "audio";
+ break;
+ case CODEC_TYPE_VIDEO:
+ mediatype = "video";
+ break;
+ default:
+ mediatype = "application";
+ break;
+ }
+ }
+ /* NOTE: the port indication is not correct in case of
+ unicast. It is not an issue because RTSP gives it */
+ payload_type = rtp_get_payload_type(&st->codec);
+ if (payload_type < 0)
+ payload_type = private_payload_type++;
+ if (stream->is_multicast) {
+ port = stream->multicast_port + 2 * i;
+ } else {
+ port = 0;
+ }
+ url_fprintf(pb, "m=%s %d RTP/AVP %d\n",
+ mediatype, port, payload_type);
+ if (payload_type >= RTP_PT_PRIVATE) {
+ /* for private payload type, we need to give more info */
+ switch(st->codec.codec_id) {
+ case CODEC_ID_MPEG4:
+ {
+ uint8_t *data;
+ url_fprintf(pb, "a=rtpmap:%d MP4V-ES/%d\n",
+ payload_type, 90000);
+ /* we must also add the mpeg4 header */
+ data = st->codec.extradata;
+ if (data) {
+ url_fprintf(pb, "a=fmtp:%d config=", payload_type);
+ for(j=0;j<st->codec.extradata_size;j++) {
+ url_fprintf(pb, "%02x", data[j]);
+ }
+ url_fprintf(pb, "\n");
+ }
+ }
+ break;
+ default:
+ /* XXX: add other codecs ? */
+ goto fail;
+ }
+ }
+ url_fprintf(pb, "a=control:streamid=%d\n", i);
+ }
+ return url_close_dyn_buf(pb, pbuffer);
+ fail:
+ url_close_dyn_buf(pb, pbuffer);
+ av_free(*pbuffer);
+ return -1;
+}
+
+static void rtsp_cmd_options(HTTPContext *c, const char *url)
+{
+// rtsp_reply_header(c, RTSP_STATUS_OK);
+ url_fprintf(c->pb, "RTSP/1.0 %d %s\r\n", RTSP_STATUS_OK, "OK");
+ url_fprintf(c->pb, "CSeq: %d\r\n", c->seq);
+ url_fprintf(c->pb, "Public: %s\r\n", "OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE");
+ url_fprintf(c->pb, "\r\n");
+}
+
+static void rtsp_cmd_describe(HTTPContext *c, const char *url)
+{
+ FFStream *stream;
+ char path1[1024];
+ const char *path;
+ uint8_t *content;
+ int content_length, len;
+ struct sockaddr_in my_addr;
+
+ /* find which url is asked */
+ url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path1, sizeof(path1), url);
+ path = path1;
+ if (*path == '/')
+ path++;
+
+ for(stream = first_stream; stream != NULL; stream = stream->next) {
+ if (!stream->is_feed && stream->fmt == &rtp_mux &&
+ !strcmp(path, stream->filename)) {
+ goto found;
+ }
+ }
+ /* no stream found */
+ rtsp_reply_error(c, RTSP_STATUS_SERVICE); /* XXX: right error ? */
+ return;
+
+ found:
+ /* prepare the media description in sdp format */
+
+ /* get the host IP */
+ len = sizeof(my_addr);
+ getsockname(c->fd, (struct sockaddr *)&my_addr, &len);
+ content_length = prepare_sdp_description(stream, &content, my_addr.sin_addr);
+ if (content_length < 0) {
+ rtsp_reply_error(c, RTSP_STATUS_INTERNAL);
+ return;
+ }
+ rtsp_reply_header(c, RTSP_STATUS_OK);
+ url_fprintf(c->pb, "Content-Type: application/sdp\r\n");
+ url_fprintf(c->pb, "Content-Length: %d\r\n", content_length);
+ url_fprintf(c->pb, "\r\n");
+ put_buffer(c->pb, content, content_length);
+}
+
+static HTTPContext *find_rtp_session(const char *session_id)
+{
+ HTTPContext *c;
+
+ if (session_id[0] == '\0')
+ return NULL;
+
+ for(c = first_http_ctx; c != NULL; c = c->next) {
+ if (!strcmp(c->session_id, session_id))
+ return c;
+ }
+ return NULL;
+}
+
+static RTSPTransportField *find_transport(RTSPHeader *h, enum RTSPProtocol protocol)
+{
+ RTSPTransportField *th;
+ int i;
+
+ for(i=0;i<h->nb_transports;i++) {
+ th = &h->transports[i];
+ if (th->protocol == protocol)
+ return th;
+ }
+ return NULL;
+}
+
+static void rtsp_cmd_setup(HTTPContext *c, const char *url,
+ RTSPHeader *h)
+{
+ FFStream *stream;
+ int stream_index, port;
+ char buf[1024];
+ char path1[1024];
+ const char *path;
+ HTTPContext *rtp_c;
+ RTSPTransportField *th;
+ struct sockaddr_in dest_addr;
+ RTSPActionServerSetup setup;
+
+ /* find which url is asked */
+ url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path1, sizeof(path1), url);
+ path = path1;
+ if (*path == '/')
+ path++;
+
+ /* now check each stream */
+ for(stream = first_stream; stream != NULL; stream = stream->next) {
+ if (!stream->is_feed && stream->fmt == &rtp_mux) {
+ /* accept aggregate filenames only if single stream */
+ if (!strcmp(path, stream->filename)) {
+ if (stream->nb_streams != 1) {
+ rtsp_reply_error(c, RTSP_STATUS_AGGREGATE);
+ return;
+ }
+ stream_index = 0;
+ goto found;
+ }
+
+ for(stream_index = 0; stream_index < stream->nb_streams;
+ stream_index++) {
+ snprintf(buf, sizeof(buf), "%s/streamid=%d",
+ stream->filename, stream_index);
+ if (!strcmp(path, buf))
+ goto found;
+ }
+ }
+ }
+ /* no stream found */
+ rtsp_reply_error(c, RTSP_STATUS_SERVICE); /* XXX: right error ? */
+ return;
+ found:
+
+ /* generate session id if needed */
+ if (h->session_id[0] == '\0') {
+ snprintf(h->session_id, sizeof(h->session_id),
+ "%08x%08x", (int)random(), (int)random());
+ }
+
+ /* find rtp session, and create it if none found */
+ rtp_c = find_rtp_session(h->session_id);
+ if (!rtp_c) {
+ /* always prefer UDP */
+ th = find_transport(h, RTSP_PROTOCOL_RTP_UDP);
+ if (!th) {
+ th = find_transport(h, RTSP_PROTOCOL_RTP_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);
+ if (!rtp_c) {
+ rtsp_reply_error(c, RTSP_STATUS_BANDWIDTH);
+ return;
+ }
+
+ /* open input stream */
+ if (open_input_stream(rtp_c, "") < 0) {
+ rtsp_reply_error(c, RTSP_STATUS_INTERNAL);
+ return;
+ }
+ }
+
+ /* test if stream is OK (test needed because several SETUP needs
+ to be done for a given file) */
+ if (rtp_c->stream != stream) {
+ rtsp_reply_error(c, RTSP_STATUS_SERVICE);
+ return;
+ }
+
+ /* test if stream is already set up */
+ if (rtp_c->rtp_ctx[stream_index]) {
+ rtsp_reply_error(c, RTSP_STATUS_STATE);
+ return;
+ }
+
+ /* check transport */
+ th = find_transport(h, rtp_c->rtp_protocol);
+ if (!th || (th->protocol == RTSP_PROTOCOL_RTP_UDP &&
+ th->client_port_min <= 0)) {
+ rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
+ return;
+ }
+
+ /* setup default options */
+ setup.transport_option[0] = '\0';
+ dest_addr = rtp_c->from_addr;
+ dest_addr.sin_port = htons(th->client_port_min);
+
+ /* add transport option if needed */
+ if (ff_rtsp_callback) {
+ setup.ipaddr = ntohl(dest_addr.sin_addr.s_addr);
+ if (ff_rtsp_callback(RTSP_ACTION_SERVER_SETUP, rtp_c->session_id,
+ (char *)&setup, sizeof(setup),
+ stream->rtsp_option) < 0) {
+ rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
+ return;
+ }
+ dest_addr.sin_addr.s_addr = htonl(setup.ipaddr);
+ }
+
+ /* setup stream */
+ if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr, c) < 0) {
+ rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
+ return;
+ }
+
+ /* now everything is OK, so we can send the connection parameters */
+ rtsp_reply_header(c, RTSP_STATUS_OK);
+ /* session ID */
+ url_fprintf(c->pb, "Session: %s\r\n", rtp_c->session_id);
+
+ switch(rtp_c->rtp_protocol) {
+ case RTSP_PROTOCOL_RTP_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:
+ url_fprintf(c->pb, "Transport: RTP/AVP/TCP;interleaved=%d-%d",
+ stream_index * 2, stream_index * 2 + 1);
+ break;
+ default:
+ break;
+ }
+ if (setup.transport_option[0] != '\0') {
+ url_fprintf(c->pb, ";%s", setup.transport_option);
+ }
+ url_fprintf(c->pb, "\r\n");
+
+
+ url_fprintf(c->pb, "\r\n");
+}
+
+
+/* find an rtp connection by using the session ID. Check consistency
+ with filename */
+static HTTPContext *find_rtp_session_with_url(const char *url,
+ const char *session_id)
+{
+ HTTPContext *rtp_c;
+ char path1[1024];
+ const char *path;
+ char buf[1024];
+ int s;
+
+ rtp_c = find_rtp_session(session_id);
+ if (!rtp_c)
+ return NULL;
+
+ /* find which url is asked */
+ url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path1, sizeof(path1), url);
+ path = path1;
+ if (*path == '/')
+ path++;
+ if(!strcmp(path, rtp_c->stream->filename)) return rtp_c;
+ for(s=0; s<rtp_c->stream->nb_streams; ++s) {
+ snprintf(buf, sizeof(buf), "%s/streamid=%d",
+ rtp_c->stream->filename, s);
+ if(!strncmp(path, buf, sizeof(buf))) {
+ // XXX: Should we reply with RTSP_STATUS_ONLY_AGGREGATE if nb_streams>1?
+ return rtp_c;
+ }
+ }
+ return NULL;
+}
+
+static void rtsp_cmd_play(HTTPContext *c, const char *url, RTSPHeader *h)
+{
+ HTTPContext *rtp_c;
+
+ rtp_c = find_rtp_session_with_url(url, h->session_id);
+ if (!rtp_c) {
+ rtsp_reply_error(c, RTSP_STATUS_SESSION);
+ return;
+ }
+
+ if (rtp_c->state != HTTPSTATE_SEND_DATA &&
+ rtp_c->state != HTTPSTATE_WAIT_FEED &&
+ rtp_c->state != HTTPSTATE_READY) {
+ rtsp_reply_error(c, RTSP_STATUS_STATE);
+ return;
+ }
+
+#if 0
+ /* XXX: seek in stream */
+ if (h->range_start != AV_NOPTS_VALUE) {
+ printf("range_start=%0.3f\n", (double)h->range_start / AV_TIME_BASE);
+ av_seek_frame(rtp_c->fmt_in, -1, h->range_start);
+ }
+#endif
+
+ rtp_c->state = HTTPSTATE_SEND_DATA;
+
+ /* now everything is OK, so we can send the connection parameters */
+ rtsp_reply_header(c, RTSP_STATUS_OK);
+ /* session ID */
+ url_fprintf(c->pb, "Session: %s\r\n", rtp_c->session_id);
+ url_fprintf(c->pb, "\r\n");
+}
+
+static void rtsp_cmd_pause(HTTPContext *c, const char *url, RTSPHeader *h)
+{
+ HTTPContext *rtp_c;
+
+ rtp_c = find_rtp_session_with_url(url, h->session_id);
+ if (!rtp_c) {
+ rtsp_reply_error(c, RTSP_STATUS_SESSION);
+ return;
+ }
+
+ if (rtp_c->state != HTTPSTATE_SEND_DATA &&
+ rtp_c->state != HTTPSTATE_WAIT_FEED) {
+ rtsp_reply_error(c, RTSP_STATUS_STATE);
+ return;
+ }
+
+ rtp_c->state = HTTPSTATE_READY;
+ rtp_c->first_pts = AV_NOPTS_VALUE;
+ /* now everything is OK, so we can send the connection parameters */
+ rtsp_reply_header(c, RTSP_STATUS_OK);
+ /* session ID */
+ url_fprintf(c->pb, "Session: %s\r\n", rtp_c->session_id);
+ url_fprintf(c->pb, "\r\n");
+}
+
+static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPHeader *h)
+{
+ HTTPContext *rtp_c;
+
+ rtp_c = find_rtp_session_with_url(url, h->session_id);
+ if (!rtp_c) {
+ rtsp_reply_error(c, RTSP_STATUS_SESSION);
+ return;
+ }
+
+ /* abort the session */
+ close_connection(rtp_c);
+
+ if (ff_rtsp_callback) {
+ ff_rtsp_callback(RTSP_ACTION_SERVER_TEARDOWN, rtp_c->session_id,
+ NULL, 0,
+ rtp_c->stream->rtsp_option);
+ }
+
+ /* now everything is OK, so we can send the connection parameters */
+ rtsp_reply_header(c, RTSP_STATUS_OK);
+ /* session ID */
+ url_fprintf(c->pb, "Session: %s\r\n", rtp_c->session_id);
+ url_fprintf(c->pb, "\r\n");
+}
+
+
+/********************************************************************/
+/* RTP handling */
+
+static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
+ FFStream *stream, const char *session_id,
+ enum RTSPProtocol rtp_protocol)
+{
+ HTTPContext *c = NULL;
+ const char *proto_str;
+
+ /* XXX: should output a warning page when coming
+ close to the connection limit */
+ if (nb_connections >= nb_max_connections)
+ goto fail;
+
+ /* add a new connection */
+ c = av_mallocz(sizeof(HTTPContext));
+ if (!c)
+ goto fail;
+
+ c->fd = -1;
+ c->poll_entry = NULL;
+ c->from_addr = *from_addr;
+ c->buffer_size = IOBUFFER_INIT_SIZE;
+ c->buffer = av_malloc(c->buffer_size);
+ if (!c->buffer)
+ goto fail;
+ nb_connections++;
+ c->stream = stream;
+ pstrcpy(c->session_id, sizeof(c->session_id), session_id);
+ c->state = HTTPSTATE_READY;
+ c->is_packetized = 1;
+ c->rtp_protocol = rtp_protocol;
+
+ /* protocol is shown in statistics */
+ switch(c->rtp_protocol) {
+ case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
+ proto_str = "MCAST";
+ break;
+ case RTSP_PROTOCOL_RTP_UDP:
+ proto_str = "UDP";
+ break;
+ case RTSP_PROTOCOL_RTP_TCP:
+ proto_str = "TCP";
+ break;
+ default:
+ proto_str = "???";
+ break;
+ }
+ pstrcpy(c->protocol, sizeof(c->protocol), "RTP/");
+ pstrcat(c->protocol, sizeof(c->protocol), proto_str);
+
+ current_bandwidth += stream->bandwidth;
+
+ c->next = first_http_ctx;
+ first_http_ctx = c;
+ return c;
+
+ fail:
+ if (c) {
+ av_free(c->buffer);
+ av_free(c);
+ }
+ return NULL;
+}
+
+/* add a new RTP stream in an RTP connection (used in RTSP SETUP
+ command). If RTP/TCP protocol is used, TCP connection 'rtsp_c' is
+ used. */
+static int rtp_new_av_stream(HTTPContext *c,
+ int stream_index, struct sockaddr_in *dest_addr,
+ HTTPContext *rtsp_c)
+{
+ AVFormatContext *ctx;
+ AVStream *st;
+ char *ipaddr;
+ URLContext *h;
+ uint8_t *dummy_buf;
+ char buf2[32];
+ int max_packet_size;
+
+ /* now we can open the relevant output stream */
+ ctx = av_alloc_format_context();
+ if (!ctx)
+ return -1;
+ ctx->oformat = &rtp_mux;
+
+ st = av_mallocz(sizeof(AVStream));
+ if (!st)
+ goto fail;
+ ctx->nb_streams = 1;
+ ctx->streams[0] = st;
+
+ if (!c->stream->feed ||
+ c->stream->feed == c->stream) {
+ memcpy(st, c->stream->streams[stream_index], sizeof(AVStream));
+ } else {
+ memcpy(st,
+ c->stream->feed->streams[c->stream->feed_streams[stream_index]],
+ sizeof(AVStream));
+ }
+
+ /* build destination RTP address */
+ ipaddr = inet_ntoa(dest_addr->sin_addr);
+
+ switch(c->rtp_protocol) {
+ case RTSP_PROTOCOL_RTP_UDP:
+ case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
+ /* RTP/UDP case */
+
+ /* XXX: also pass as parameter to function ? */
+ if (c->stream->is_multicast) {
+ int ttl;
+ ttl = c->stream->multicast_ttl;
+ if (!ttl)
+ ttl = 16;
+ snprintf(ctx->filename, sizeof(ctx->filename),
+ "rtp://%s:%d?multicast=1&ttl=%d",
+ ipaddr, ntohs(dest_addr->sin_port), ttl);
+ } else {
+ snprintf(ctx->filename, sizeof(ctx->filename),
+ "rtp://%s:%d", ipaddr, ntohs(dest_addr->sin_port));
+ }
+
+ if (url_open(&h, ctx->filename, URL_WRONLY) < 0)
+ goto fail;
+ c->rtp_handles[stream_index] = h;
+ max_packet_size = url_get_max_packet_size(h);
+ break;
+ case RTSP_PROTOCOL_RTP_TCP:
+ /* RTP/TCP case */
+ c->rtsp_c = rtsp_c;
+ max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
+ break;
+ default:
+ goto fail;
+ }
+
+ http_log("%s:%d - - [%s] \"PLAY %s/streamid=%d %s\"\n",
+ ipaddr, ntohs(dest_addr->sin_port),
+ ctime1(buf2),
+ c->stream->filename, stream_index, c->protocol);
+
+ /* normally, no packets should be output here, but the packet size may be checked */
+ if (url_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0) {
+ /* XXX: close stream */
+ goto fail;
+ }
+ av_set_parameters(ctx, NULL);
+ if (av_write_header(ctx) < 0) {
+ fail:
+ if (h)
+ url_close(h);
+ av_free(ctx);
+ return -1;
+ }
+ url_close_dyn_buf(&ctx->pb, &dummy_buf);
+ av_free(dummy_buf);
+
+ c->rtp_ctx[stream_index] = ctx;
+ return 0;
+}
+
+/********************************************************************/
+/* ffserver initialization */
+
+static AVStream *add_av_stream1(FFStream *stream, AVCodecContext *codec)
+{
+ AVStream *fst;
+
+ fst = av_mallocz(sizeof(AVStream));
+ if (!fst)
+ return NULL;
+ fst->priv_data = av_mallocz(sizeof(FeedData));
+ memcpy(&fst->codec, codec, sizeof(AVCodecContext));
+ fst->codec.coded_frame = &dummy_frame;
+ fst->index = stream->nb_streams;
+ av_set_pts_info(fst, 33, 1, 90000);
+ stream->streams[stream->nb_streams++] = fst;
+ return fst;
+}
+