+ c->buffer_ptr += len;
+ c->data_count += len;
+ update_datarate(&c->datarate, c->data_count);
+ }
+ }
+
+ if (c->buffer_ptr - c->buffer >= 2 && c->data_count > FFM_PACKET_SIZE) {
+ if (c->buffer[0] != 'f' ||
+ c->buffer[1] != 'm') {
+ http_log("Feed stream has become desynchronized -- disconnecting\n");
+ goto fail;
+ }
+ }
+
+ if (c->buffer_ptr >= c->buffer_end) {
+ FFStream *feed = c->stream;
+ /* a packet has been received : write it in the store, except
+ if header */
+ if (c->data_count > FFM_PACKET_SIZE) {
+
+ // printf("writing pos=0x%Lx size=0x%Lx\n", feed->feed_write_index, feed->feed_size);
+ /* XXX: use llseek or url_seek */
+ lseek(c->feed_fd, feed->feed_write_index, SEEK_SET);
+ write(c->feed_fd, c->buffer, FFM_PACKET_SIZE);
+
+ feed->feed_write_index += FFM_PACKET_SIZE;
+ /* update file size */
+ if (feed->feed_write_index > c->stream->feed_size)
+ feed->feed_size = feed->feed_write_index;
+
+ /* handle wrap around if max file size reached */
+ if (c->stream->feed_max_size && feed->feed_write_index >= c->stream->feed_max_size)
+ feed->feed_write_index = FFM_PACKET_SIZE;
+
+ /* write index */
+ ffm_write_write_index(c->feed_fd, feed->feed_write_index);
+
+ /* wake up any waiting connections */
+ for(c1 = first_http_ctx; c1 != NULL; c1 = c1->next) {
+ if (c1->state == HTTPSTATE_WAIT_FEED &&
+ c1->stream->feed == c->stream->feed) {
+ c1->state = HTTPSTATE_SEND_DATA;
+ }
+ }
+ } else {
+ /* We have a header in our hands that contains useful data */
+ AVFormatContext s;
+ AVInputFormat *fmt_in;
+ ByteIOContext *pb = &s.pb;
+ int i;
+
+ memset(&s, 0, sizeof(s));
+
+ url_open_buf(pb, c->buffer, c->buffer_end - c->buffer, URL_RDONLY);
+ pb->buf_end = c->buffer_end; /* ?? */
+ pb->is_streamed = 1;
+
+ /* use feed output format name to find corresponding input format */
+ fmt_in = av_find_input_format(feed->fmt->name);
+ if (!fmt_in)
+ goto fail;
+
+ if (fmt_in->priv_data_size > 0) {
+ s.priv_data = av_mallocz(fmt_in->priv_data_size);
+ if (!s.priv_data)
+ goto fail;
+ } else
+ s.priv_data = NULL;
+
+ if (fmt_in->read_header(&s, 0) < 0) {
+ av_freep(&s.priv_data);
+ goto fail;
+ }
+
+ /* Now we have the actual streams */
+ if (s.nb_streams != feed->nb_streams) {
+ av_freep(&s.priv_data);
+ goto fail;
+ }
+ for (i = 0; i < s.nb_streams; i++) {
+ memcpy(feed->streams[i]->codec,
+ s.streams[i]->codec, sizeof(AVCodecContext));
+ }
+ av_freep(&s.priv_data);
+ }
+ c->buffer_ptr = c->buffer;
+ }
+
+ return 0;
+ fail:
+ c->stream->feed_opened = 0;
+ close(c->feed_fd);
+ return -1;
+}
+
+/********************************************************************/
+/* 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);
+ }