mirror of
https://github.com/FFmpeg/FFmpeg.git
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6df9d9b55d
The ones left using av_gettime are NTP timestamps (for RTCP, which is specified to send the actual current realtime clock in RTCP SR packets), and the NUT muxer timestamper, which is documented as using wallclock time. Signed-off-by: Martin Storsjö <martin@martin.st>
954 lines
32 KiB
C
954 lines
32 KiB
C
/*
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* RTSP demuxer
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* Copyright (c) 2002 Fabrice Bellard
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/avstring.h"
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#include "libavutil/intreadwrite.h"
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#include "libavutil/mathematics.h"
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#include "libavutil/random_seed.h"
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#include "libavutil/time.h"
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#include "avformat.h"
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#include "internal.h"
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#include "network.h"
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#include "os_support.h"
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#include "rtpproto.h"
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#include "rtsp.h"
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#include "rdt.h"
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#include "url.h"
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static const struct RTSPStatusMessage {
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enum RTSPStatusCode code;
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const char *message;
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} status_messages[] = {
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{ RTSP_STATUS_OK, "OK" },
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{ RTSP_STATUS_METHOD, "Method Not Allowed" },
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{ RTSP_STATUS_BANDWIDTH, "Not Enough Bandwidth" },
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{ RTSP_STATUS_SESSION, "Session Not Found" },
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{ RTSP_STATUS_STATE, "Method Not Valid in This State" },
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{ RTSP_STATUS_AGGREGATE, "Aggregate operation not allowed" },
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{ RTSP_STATUS_ONLY_AGGREGATE, "Only aggregate operation allowed" },
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{ RTSP_STATUS_TRANSPORT, "Unsupported transport" },
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{ RTSP_STATUS_INTERNAL, "Internal Server Error" },
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{ RTSP_STATUS_SERVICE, "Service Unavailable" },
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{ RTSP_STATUS_VERSION, "RTSP Version not supported" },
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{ 0, "NULL" }
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};
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static int rtsp_read_close(AVFormatContext *s)
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{
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RTSPState *rt = s->priv_data;
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if (!(rt->rtsp_flags & RTSP_FLAG_LISTEN))
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ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
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ff_rtsp_close_streams(s);
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ff_rtsp_close_connections(s);
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ff_network_close();
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rt->real_setup = NULL;
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av_freep(&rt->real_setup_cache);
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return 0;
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}
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static inline int read_line(AVFormatContext *s, char *rbuf, const int rbufsize,
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int *rbuflen)
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{
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RTSPState *rt = s->priv_data;
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int idx = 0;
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int ret = 0;
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*rbuflen = 0;
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do {
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ret = ffurl_read_complete(rt->rtsp_hd, rbuf + idx, 1);
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if (ret <= 0)
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return ret ? ret : AVERROR_EOF;
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if (rbuf[idx] == '\r') {
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/* Ignore */
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} else if (rbuf[idx] == '\n') {
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rbuf[idx] = '\0';
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*rbuflen = idx;
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return 0;
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} else
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idx++;
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} while (idx < rbufsize);
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av_log(s, AV_LOG_ERROR, "Message too long\n");
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return AVERROR(EIO);
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}
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static int rtsp_send_reply(AVFormatContext *s, enum RTSPStatusCode code,
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const char *extracontent, uint16_t seq)
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{
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RTSPState *rt = s->priv_data;
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char message[4096];
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int index = 0;
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while (status_messages[index].code) {
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if (status_messages[index].code == code) {
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snprintf(message, sizeof(message), "RTSP/1.0 %d %s\r\n",
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code, status_messages[index].message);
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break;
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}
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index++;
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}
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if (!status_messages[index].code)
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return AVERROR(EINVAL);
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av_strlcatf(message, sizeof(message), "CSeq: %d\r\n", seq);
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av_strlcatf(message, sizeof(message), "Server: %s\r\n", LIBAVFORMAT_IDENT);
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if (extracontent)
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av_strlcat(message, extracontent, sizeof(message));
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av_strlcat(message, "\r\n", sizeof(message));
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av_dlog(s, "Sending response:\n%s", message);
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ffurl_write(rt->rtsp_hd, message, strlen(message));
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return 0;
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}
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static inline int check_sessionid(AVFormatContext *s,
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RTSPMessageHeader *request)
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{
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RTSPState *rt = s->priv_data;
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unsigned char *session_id = rt->session_id;
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if (!session_id[0]) {
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av_log(s, AV_LOG_WARNING, "There is no session-id at the moment\n");
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return 0;
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}
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if (strcmp(session_id, request->session_id)) {
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av_log(s, AV_LOG_ERROR, "Unexpected session-id %s\n",
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request->session_id);
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rtsp_send_reply(s, RTSP_STATUS_SESSION, NULL, request->seq);
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return AVERROR_STREAM_NOT_FOUND;
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}
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return 0;
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}
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static inline int rtsp_read_request(AVFormatContext *s,
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RTSPMessageHeader *request,
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const char *method)
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{
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RTSPState *rt = s->priv_data;
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char rbuf[1024];
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int rbuflen, ret;
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do {
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ret = read_line(s, rbuf, sizeof(rbuf), &rbuflen);
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if (ret)
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return ret;
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if (rbuflen > 1) {
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av_dlog(s, "Parsing[%d]: %s\n", rbuflen, rbuf);
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ff_rtsp_parse_line(request, rbuf, rt, method);
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}
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} while (rbuflen > 0);
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if (request->seq != rt->seq + 1) {
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av_log(s, AV_LOG_ERROR, "Unexpected Sequence number %d\n",
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request->seq);
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return AVERROR(EINVAL);
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}
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if (rt->session_id[0] && strcmp(method, "OPTIONS")) {
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ret = check_sessionid(s, request);
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if (ret)
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return ret;
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}
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return 0;
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}
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static int rtsp_read_announce(AVFormatContext *s)
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{
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RTSPState *rt = s->priv_data;
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RTSPMessageHeader request = { 0 };
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char sdp[4096];
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int ret;
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ret = rtsp_read_request(s, &request, "ANNOUNCE");
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if (ret)
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return ret;
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rt->seq++;
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if (strcmp(request.content_type, "application/sdp")) {
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av_log(s, AV_LOG_ERROR, "Unexpected content type %s\n",
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request.content_type);
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rtsp_send_reply(s, RTSP_STATUS_SERVICE, NULL, request.seq);
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return AVERROR_OPTION_NOT_FOUND;
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}
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if (request.content_length && request.content_length < sizeof(sdp) - 1) {
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/* Read SDP */
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if (ffurl_read_complete(rt->rtsp_hd, sdp, request.content_length)
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< request.content_length) {
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av_log(s, AV_LOG_ERROR,
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"Unable to get complete SDP Description in ANNOUNCE\n");
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rtsp_send_reply(s, RTSP_STATUS_INTERNAL, NULL, request.seq);
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return AVERROR(EIO);
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}
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sdp[request.content_length] = '\0';
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av_log(s, AV_LOG_VERBOSE, "SDP: %s\n", sdp);
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ret = ff_sdp_parse(s, sdp);
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if (ret)
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return ret;
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rtsp_send_reply(s, RTSP_STATUS_OK, NULL, request.seq);
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return 0;
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}
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av_log(s, AV_LOG_ERROR,
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"Content-Length header value exceeds sdp allocated buffer (4KB)\n");
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rtsp_send_reply(s, RTSP_STATUS_INTERNAL,
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"Content-Length exceeds buffer size", request.seq);
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return AVERROR(EIO);
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}
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static int rtsp_read_options(AVFormatContext *s)
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{
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RTSPState *rt = s->priv_data;
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RTSPMessageHeader request = { 0 };
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int ret = 0;
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/* Parsing headers */
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ret = rtsp_read_request(s, &request, "OPTIONS");
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if (ret)
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return ret;
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rt->seq++;
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/* Send Reply */
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rtsp_send_reply(s, RTSP_STATUS_OK,
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"Public: ANNOUNCE, PAUSE, SETUP, TEARDOWN, RECORD\r\n",
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request.seq);
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return 0;
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}
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static int rtsp_read_setup(AVFormatContext *s, char* host, char *controlurl)
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{
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RTSPState *rt = s->priv_data;
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RTSPMessageHeader request = { 0 };
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int ret = 0;
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char url[1024];
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RTSPStream *rtsp_st;
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char responseheaders[1024];
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int localport = -1;
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int transportidx = 0;
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int streamid = 0;
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ret = rtsp_read_request(s, &request, "SETUP");
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if (ret)
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return ret;
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rt->seq++;
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if (!request.nb_transports) {
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av_log(s, AV_LOG_ERROR, "No transport defined in SETUP\n");
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return AVERROR_INVALIDDATA;
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}
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for (transportidx = 0; transportidx < request.nb_transports;
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transportidx++) {
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if (!request.transports[transportidx].mode_record ||
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(request.transports[transportidx].lower_transport !=
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RTSP_LOWER_TRANSPORT_UDP &&
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request.transports[transportidx].lower_transport !=
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RTSP_LOWER_TRANSPORT_TCP)) {
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av_log(s, AV_LOG_ERROR, "mode=record/receive not set or transport"
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" protocol not supported (yet)\n");
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return AVERROR_INVALIDDATA;
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}
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}
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if (request.nb_transports > 1)
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av_log(s, AV_LOG_WARNING, "More than one transport not supported, "
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"using first of all\n");
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for (streamid = 0; streamid < rt->nb_rtsp_streams; streamid++) {
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if (!strcmp(rt->rtsp_streams[streamid]->control_url,
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controlurl))
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break;
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}
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if (streamid == rt->nb_rtsp_streams) {
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av_log(s, AV_LOG_ERROR, "Unable to find requested track\n");
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return AVERROR_STREAM_NOT_FOUND;
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}
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rtsp_st = rt->rtsp_streams[streamid];
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localport = rt->rtp_port_min;
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if (request.transports[0].lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
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rt->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
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if ((ret = ff_rtsp_open_transport_ctx(s, rtsp_st))) {
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rtsp_send_reply(s, RTSP_STATUS_TRANSPORT, NULL, request.seq);
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return ret;
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}
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rtsp_st->interleaved_min = request.transports[0].interleaved_min;
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rtsp_st->interleaved_max = request.transports[0].interleaved_max;
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snprintf(responseheaders, sizeof(responseheaders), "Transport: "
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"RTP/AVP/TCP;unicast;mode=receive;interleaved=%d-%d"
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"\r\n", request.transports[0].interleaved_min,
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request.transports[0].interleaved_max);
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} else {
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do {
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ff_url_join(url, sizeof(url), "rtp", NULL, host, localport, NULL);
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av_dlog(s, "Opening: %s", url);
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ret = ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
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&s->interrupt_callback, NULL);
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if (ret)
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localport += 2;
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} while (ret || localport > rt->rtp_port_max);
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if (localport > rt->rtp_port_max) {
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rtsp_send_reply(s, RTSP_STATUS_TRANSPORT, NULL, request.seq);
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return ret;
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}
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av_dlog(s, "Listening on: %d",
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ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle));
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if ((ret = ff_rtsp_open_transport_ctx(s, rtsp_st))) {
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rtsp_send_reply(s, RTSP_STATUS_TRANSPORT, NULL, request.seq);
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return ret;
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}
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localport = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
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snprintf(responseheaders, sizeof(responseheaders), "Transport: "
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"RTP/AVP/UDP;unicast;mode=receive;source=%s;"
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"client_port=%d-%d;server_port=%d-%d\r\n",
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host, request.transports[0].client_port_min,
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request.transports[0].client_port_max, localport,
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localport + 1);
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}
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/* Establish sessionid if not previously set */
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/* Put this in a function? */
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/* RFC 2326: session id must be at least 8 digits */
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while (strlen(rt->session_id) < 8)
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av_strlcatf(rt->session_id, 512, "%u", av_get_random_seed());
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av_strlcatf(responseheaders, sizeof(responseheaders), "Session: %s\r\n",
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rt->session_id);
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/* Send Reply */
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rtsp_send_reply(s, RTSP_STATUS_OK, responseheaders, request.seq);
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rt->state = RTSP_STATE_PAUSED;
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return 0;
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}
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static int rtsp_read_record(AVFormatContext *s)
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{
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RTSPState *rt = s->priv_data;
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RTSPMessageHeader request = { 0 };
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int ret = 0;
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char responseheaders[1024];
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ret = rtsp_read_request(s, &request, "RECORD");
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if (ret)
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return ret;
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ret = check_sessionid(s, &request);
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if (ret)
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return ret;
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rt->seq++;
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snprintf(responseheaders, sizeof(responseheaders), "Session: %s\r\n",
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rt->session_id);
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rtsp_send_reply(s, RTSP_STATUS_OK, responseheaders, request.seq);
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rt->state = RTSP_STATE_STREAMING;
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return 0;
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}
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static inline int parse_command_line(AVFormatContext *s, const char *line,
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int linelen, char *uri, int urisize,
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char *method, int methodsize,
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enum RTSPMethod *methodcode)
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{
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RTSPState *rt = s->priv_data;
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const char *linept, *searchlinept;
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linept = strchr(line, ' ');
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if (!linept)
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return AVERROR_INVALIDDATA;
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if (linept - line > methodsize - 1) {
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av_log(s, AV_LOG_ERROR, "Method string too long\n");
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return AVERROR(EIO);
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}
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memcpy(method, line, linept - line);
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method[linept - line] = '\0';
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linept++;
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if (!strcmp(method, "ANNOUNCE"))
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*methodcode = ANNOUNCE;
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else if (!strcmp(method, "OPTIONS"))
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*methodcode = OPTIONS;
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else if (!strcmp(method, "RECORD"))
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*methodcode = RECORD;
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else if (!strcmp(method, "SETUP"))
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*methodcode = SETUP;
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else if (!strcmp(method, "PAUSE"))
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*methodcode = PAUSE;
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else if (!strcmp(method, "TEARDOWN"))
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*methodcode = TEARDOWN;
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else
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*methodcode = UNKNOWN;
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/* Check method with the state */
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if (rt->state == RTSP_STATE_IDLE) {
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if ((*methodcode != ANNOUNCE) && (*methodcode != OPTIONS)) {
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av_log(s, AV_LOG_ERROR, "Unexpected command in Idle State %s\n",
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line);
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return AVERROR_PROTOCOL_NOT_FOUND;
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}
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} else if (rt->state == RTSP_STATE_PAUSED) {
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if ((*methodcode != OPTIONS) && (*methodcode != RECORD)
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&& (*methodcode != SETUP)) {
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av_log(s, AV_LOG_ERROR, "Unexpected command in Paused State %s\n",
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line);
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return AVERROR_PROTOCOL_NOT_FOUND;
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}
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} else if (rt->state == RTSP_STATE_STREAMING) {
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if ((*methodcode != PAUSE) && (*methodcode != OPTIONS)
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&& (*methodcode != TEARDOWN)) {
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av_log(s, AV_LOG_ERROR, "Unexpected command in Streaming State"
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" %s\n", line);
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return AVERROR_PROTOCOL_NOT_FOUND;
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}
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} else {
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av_log(s, AV_LOG_ERROR, "Unexpected State [%d]\n", rt->state);
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return AVERROR_BUG;
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}
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searchlinept = strchr(linept, ' ');
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if (!searchlinept) {
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av_log(s, AV_LOG_ERROR, "Error parsing message URI\n");
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return AVERROR_INVALIDDATA;
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}
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if (searchlinept - linept > urisize - 1) {
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av_log(s, AV_LOG_ERROR, "uri string length exceeded buffer size\n");
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return AVERROR(EIO);
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}
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memcpy(uri, linept, searchlinept - linept);
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uri[searchlinept - linept] = '\0';
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if (strcmp(rt->control_uri, uri)) {
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char host[128], path[512], auth[128];
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int port;
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char ctl_host[128], ctl_path[512], ctl_auth[128];
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int ctl_port;
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av_url_split(NULL, 0, auth, sizeof(auth), host, sizeof(host), &port,
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path, sizeof(path), uri);
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av_url_split(NULL, 0, ctl_auth, sizeof(ctl_auth), ctl_host,
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sizeof(ctl_host), &ctl_port, ctl_path, sizeof(ctl_path),
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rt->control_uri);
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if (strcmp(host, ctl_host))
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av_log(s, AV_LOG_INFO, "Host %s differs from expected %s\n",
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host, ctl_host);
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if (strcmp(path, ctl_path) && *methodcode != SETUP)
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av_log(s, AV_LOG_WARNING, "WARNING: Path %s differs from expected"
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" %s\n", path, ctl_path);
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if (*methodcode == ANNOUNCE) {
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av_log(s, AV_LOG_INFO,
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"Updating control URI to %s\n", uri);
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av_strlcpy(rt->control_uri, uri, sizeof(rt->control_uri));
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}
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}
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linept = searchlinept + 1;
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if (!av_strstart(linept, "RTSP/1.0", NULL)) {
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av_log(s, AV_LOG_ERROR, "Error parsing protocol or version\n");
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return AVERROR_PROTOCOL_NOT_FOUND;
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}
|
|
return 0;
|
|
}
|
|
|
|
int ff_rtsp_parse_streaming_commands(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
unsigned char rbuf[4096];
|
|
unsigned char method[10];
|
|
char uri[500];
|
|
int ret;
|
|
int rbuflen = 0;
|
|
RTSPMessageHeader request = { 0 };
|
|
enum RTSPMethod methodcode;
|
|
|
|
ret = read_line(s, rbuf, sizeof(rbuf), &rbuflen);
|
|
if (ret < 0)
|
|
return ret;
|
|
ret = parse_command_line(s, rbuf, rbuflen, uri, sizeof(uri), method,
|
|
sizeof(method), &methodcode);
|
|
if (ret) {
|
|
av_log(s, AV_LOG_ERROR, "RTSP: Unexpected Command\n");
|
|
return ret;
|
|
}
|
|
|
|
ret = rtsp_read_request(s, &request, method);
|
|
if (ret)
|
|
return ret;
|
|
rt->seq++;
|
|
if (methodcode == PAUSE) {
|
|
rt->state = RTSP_STATE_PAUSED;
|
|
ret = rtsp_send_reply(s, RTSP_STATUS_OK, NULL , request.seq);
|
|
// TODO: Missing date header in response
|
|
} else if (methodcode == OPTIONS) {
|
|
ret = rtsp_send_reply(s, RTSP_STATUS_OK,
|
|
"Public: ANNOUNCE, PAUSE, SETUP, TEARDOWN, "
|
|
"RECORD\r\n", request.seq);
|
|
} else if (methodcode == TEARDOWN) {
|
|
rt->state = RTSP_STATE_IDLE;
|
|
ret = rtsp_send_reply(s, RTSP_STATUS_OK, NULL , request.seq);
|
|
return 0;
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
static int rtsp_read_play(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
RTSPMessageHeader reply1, *reply = &reply1;
|
|
int i;
|
|
char cmd[1024];
|
|
|
|
av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
|
|
rt->nb_byes = 0;
|
|
|
|
if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
|
|
if (rt->transport == RTSP_TRANSPORT_RTP) {
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
RTSPStream *rtsp_st = rt->rtsp_streams[i];
|
|
RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
|
|
if (!rtpctx)
|
|
continue;
|
|
ff_rtp_reset_packet_queue(rtpctx);
|
|
rtpctx->last_rtcp_ntp_time = AV_NOPTS_VALUE;
|
|
rtpctx->first_rtcp_ntp_time = AV_NOPTS_VALUE;
|
|
rtpctx->base_timestamp = 0;
|
|
rtpctx->timestamp = 0;
|
|
rtpctx->unwrapped_timestamp = 0;
|
|
rtpctx->rtcp_ts_offset = 0;
|
|
}
|
|
}
|
|
if (rt->state == RTSP_STATE_PAUSED) {
|
|
cmd[0] = 0;
|
|
} else {
|
|
snprintf(cmd, sizeof(cmd),
|
|
"Range: npt=%"PRId64".%03"PRId64"-\r\n",
|
|
rt->seek_timestamp / AV_TIME_BASE,
|
|
rt->seek_timestamp / (AV_TIME_BASE / 1000) % 1000);
|
|
}
|
|
ff_rtsp_send_cmd(s, "PLAY", rt->control_uri, cmd, reply, NULL);
|
|
if (reply->status_code != RTSP_STATUS_OK) {
|
|
return -1;
|
|
}
|
|
if (rt->transport == RTSP_TRANSPORT_RTP &&
|
|
reply->range_start != AV_NOPTS_VALUE) {
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
RTSPStream *rtsp_st = rt->rtsp_streams[i];
|
|
RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
|
|
AVStream *st = NULL;
|
|
if (!rtpctx || rtsp_st->stream_index < 0)
|
|
continue;
|
|
st = s->streams[rtsp_st->stream_index];
|
|
rtpctx->range_start_offset =
|
|
av_rescale_q(reply->range_start, AV_TIME_BASE_Q,
|
|
st->time_base);
|
|
}
|
|
}
|
|
}
|
|
rt->state = RTSP_STATE_STREAMING;
|
|
return 0;
|
|
}
|
|
|
|
/* pause the stream */
|
|
static int rtsp_read_pause(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
RTSPMessageHeader reply1, *reply = &reply1;
|
|
|
|
if (rt->state != RTSP_STATE_STREAMING)
|
|
return 0;
|
|
else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
|
|
ff_rtsp_send_cmd(s, "PAUSE", rt->control_uri, NULL, reply, NULL);
|
|
if (reply->status_code != RTSP_STATUS_OK) {
|
|
return -1;
|
|
}
|
|
}
|
|
rt->state = RTSP_STATE_PAUSED;
|
|
return 0;
|
|
}
|
|
|
|
int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
char cmd[1024];
|
|
unsigned char *content = NULL;
|
|
int ret;
|
|
|
|
/* describe the stream */
|
|
snprintf(cmd, sizeof(cmd),
|
|
"Accept: application/sdp\r\n");
|
|
if (rt->server_type == RTSP_SERVER_REAL) {
|
|
/**
|
|
* The Require: attribute is needed for proper streaming from
|
|
* Realmedia servers.
|
|
*/
|
|
av_strlcat(cmd,
|
|
"Require: com.real.retain-entity-for-setup\r\n",
|
|
sizeof(cmd));
|
|
}
|
|
ff_rtsp_send_cmd(s, "DESCRIBE", rt->control_uri, cmd, reply, &content);
|
|
if (!content)
|
|
return AVERROR_INVALIDDATA;
|
|
if (reply->status_code != RTSP_STATUS_OK) {
|
|
av_freep(&content);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", content);
|
|
/* now we got the SDP description, we parse it */
|
|
ret = ff_sdp_parse(s, (const char *)content);
|
|
av_freep(&content);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int rtsp_listen(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
char proto[128], host[128], path[512], auth[128];
|
|
char uri[500];
|
|
int port;
|
|
int default_port = RTSP_DEFAULT_PORT;
|
|
char tcpname[500];
|
|
const char *lower_proto = "tcp";
|
|
unsigned char rbuf[4096];
|
|
unsigned char method[10];
|
|
int rbuflen = 0;
|
|
int ret;
|
|
enum RTSPMethod methodcode;
|
|
|
|
/* extract hostname and port */
|
|
av_url_split(proto, sizeof(proto), auth, sizeof(auth), host, sizeof(host),
|
|
&port, path, sizeof(path), s->filename);
|
|
|
|
/* ff_url_join. No authorization by now (NULL) */
|
|
ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL, host,
|
|
port, "%s", path);
|
|
|
|
if (!strcmp(proto, "rtsps")) {
|
|
lower_proto = "tls";
|
|
default_port = RTSPS_DEFAULT_PORT;
|
|
}
|
|
|
|
if (port < 0)
|
|
port = default_port;
|
|
|
|
/* Create TCP connection */
|
|
ff_url_join(tcpname, sizeof(tcpname), lower_proto, NULL, host, port,
|
|
"?listen&listen_timeout=%d", rt->initial_timeout * 1000);
|
|
|
|
if (ret = ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
|
|
&s->interrupt_callback, NULL)) {
|
|
av_log(s, AV_LOG_ERROR, "Unable to open RTSP for listening\n");
|
|
return ret;
|
|
}
|
|
rt->state = RTSP_STATE_IDLE;
|
|
rt->rtsp_hd_out = rt->rtsp_hd;
|
|
for (;;) { /* Wait for incoming RTSP messages */
|
|
ret = read_line(s, rbuf, sizeof(rbuf), &rbuflen);
|
|
if (ret < 0)
|
|
return ret;
|
|
ret = parse_command_line(s, rbuf, rbuflen, uri, sizeof(uri), method,
|
|
sizeof(method), &methodcode);
|
|
if (ret) {
|
|
av_log(s, AV_LOG_ERROR, "RTSP: Unexpected Command\n");
|
|
return ret;
|
|
}
|
|
|
|
if (methodcode == ANNOUNCE) {
|
|
ret = rtsp_read_announce(s);
|
|
rt->state = RTSP_STATE_PAUSED;
|
|
} else if (methodcode == OPTIONS) {
|
|
ret = rtsp_read_options(s);
|
|
} else if (methodcode == RECORD) {
|
|
ret = rtsp_read_record(s);
|
|
if (!ret)
|
|
return 0; // We are ready for streaming
|
|
} else if (methodcode == SETUP)
|
|
ret = rtsp_read_setup(s, host, uri);
|
|
if (ret) {
|
|
ffurl_close(rt->rtsp_hd);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int rtsp_probe(AVProbeData *p)
|
|
{
|
|
if (
|
|
#if CONFIG_TLS_PROTOCOL
|
|
av_strstart(p->filename, "rtsps:", NULL) ||
|
|
#endif
|
|
av_strstart(p->filename, "rtsp:", NULL))
|
|
return AVPROBE_SCORE_MAX;
|
|
return 0;
|
|
}
|
|
|
|
static int rtsp_read_header(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int ret;
|
|
|
|
if (rt->initial_timeout > 0)
|
|
rt->rtsp_flags |= RTSP_FLAG_LISTEN;
|
|
|
|
if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
|
|
ret = rtsp_listen(s);
|
|
if (ret)
|
|
return ret;
|
|
} else {
|
|
ret = ff_rtsp_connect(s);
|
|
if (ret)
|
|
return ret;
|
|
|
|
rt->real_setup_cache = !s->nb_streams ? NULL :
|
|
av_mallocz(2 * s->nb_streams * sizeof(*rt->real_setup_cache));
|
|
if (!rt->real_setup_cache && s->nb_streams)
|
|
return AVERROR(ENOMEM);
|
|
rt->real_setup = rt->real_setup_cache + s->nb_streams;
|
|
|
|
if (rt->initial_pause) {
|
|
/* do not start immediately */
|
|
} else {
|
|
if (rtsp_read_play(s) < 0) {
|
|
ff_rtsp_close_streams(s);
|
|
ff_rtsp_close_connections(s);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
|
|
uint8_t *buf, int buf_size)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int id, len, i, ret;
|
|
RTSPStream *rtsp_st;
|
|
|
|
av_dlog(s, "tcp_read_packet:\n");
|
|
redo:
|
|
for (;;) {
|
|
RTSPMessageHeader reply;
|
|
|
|
ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL);
|
|
if (ret < 0)
|
|
return ret;
|
|
if (ret == 1) /* received '$' */
|
|
break;
|
|
/* XXX: parse message */
|
|
if (rt->state != RTSP_STATE_STREAMING)
|
|
return 0;
|
|
}
|
|
ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
|
|
if (ret != 3)
|
|
return -1;
|
|
id = buf[0];
|
|
len = AV_RB16(buf + 1);
|
|
av_dlog(s, "id=%d len=%d\n", id, len);
|
|
if (len > buf_size || len < 12)
|
|
goto redo;
|
|
/* get the data */
|
|
ret = ffurl_read_complete(rt->rtsp_hd, buf, len);
|
|
if (ret != len)
|
|
return -1;
|
|
if (rt->transport == RTSP_TRANSPORT_RDT &&
|
|
ff_rdt_parse_header(buf, len, &id, NULL, NULL, NULL, NULL) < 0)
|
|
return -1;
|
|
|
|
/* find the matching stream */
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
rtsp_st = rt->rtsp_streams[i];
|
|
if (id >= rtsp_st->interleaved_min &&
|
|
id <= rtsp_st->interleaved_max)
|
|
goto found;
|
|
}
|
|
goto redo;
|
|
found:
|
|
*prtsp_st = rtsp_st;
|
|
return len;
|
|
}
|
|
|
|
static int resetup_tcp(AVFormatContext *s)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
char host[1024];
|
|
int port;
|
|
|
|
av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port, NULL, 0,
|
|
s->filename);
|
|
ff_rtsp_undo_setup(s, 0);
|
|
return ff_rtsp_make_setup_request(s, host, port, RTSP_LOWER_TRANSPORT_TCP,
|
|
rt->real_challenge);
|
|
}
|
|
|
|
static int rtsp_read_packet(AVFormatContext *s, AVPacket *pkt)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
int ret;
|
|
RTSPMessageHeader reply1, *reply = &reply1;
|
|
char cmd[1024];
|
|
|
|
retry:
|
|
if (rt->server_type == RTSP_SERVER_REAL) {
|
|
int i;
|
|
|
|
for (i = 0; i < s->nb_streams; i++)
|
|
rt->real_setup[i] = s->streams[i]->discard;
|
|
|
|
if (!rt->need_subscription) {
|
|
if (memcmp (rt->real_setup, rt->real_setup_cache,
|
|
sizeof(enum AVDiscard) * s->nb_streams)) {
|
|
snprintf(cmd, sizeof(cmd),
|
|
"Unsubscribe: %s\r\n",
|
|
rt->last_subscription);
|
|
ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
|
|
cmd, reply, NULL);
|
|
if (reply->status_code != RTSP_STATUS_OK)
|
|
return AVERROR_INVALIDDATA;
|
|
rt->need_subscription = 1;
|
|
}
|
|
}
|
|
|
|
if (rt->need_subscription) {
|
|
int r, rule_nr, first = 1;
|
|
|
|
memcpy(rt->real_setup_cache, rt->real_setup,
|
|
sizeof(enum AVDiscard) * s->nb_streams);
|
|
rt->last_subscription[0] = 0;
|
|
|
|
snprintf(cmd, sizeof(cmd),
|
|
"Subscribe: ");
|
|
for (i = 0; i < rt->nb_rtsp_streams; i++) {
|
|
rule_nr = 0;
|
|
for (r = 0; r < s->nb_streams; r++) {
|
|
if (s->streams[r]->id == i) {
|
|
if (s->streams[r]->discard != AVDISCARD_ALL) {
|
|
if (!first)
|
|
av_strlcat(rt->last_subscription, ",",
|
|
sizeof(rt->last_subscription));
|
|
ff_rdt_subscribe_rule(
|
|
rt->last_subscription,
|
|
sizeof(rt->last_subscription), i, rule_nr);
|
|
first = 0;
|
|
}
|
|
rule_nr++;
|
|
}
|
|
}
|
|
}
|
|
av_strlcatf(cmd, sizeof(cmd), "%s\r\n", rt->last_subscription);
|
|
ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
|
|
cmd, reply, NULL);
|
|
if (reply->status_code != RTSP_STATUS_OK)
|
|
return AVERROR_INVALIDDATA;
|
|
rt->need_subscription = 0;
|
|
|
|
if (rt->state == RTSP_STATE_STREAMING)
|
|
rtsp_read_play (s);
|
|
}
|
|
}
|
|
|
|
ret = ff_rtsp_fetch_packet(s, pkt);
|
|
if (ret < 0) {
|
|
if (ret == AVERROR(ETIMEDOUT) && !rt->packets) {
|
|
if (rt->lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
|
|
rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP)) {
|
|
RTSPMessageHeader reply1, *reply = &reply1;
|
|
av_log(s, AV_LOG_WARNING, "UDP timeout, retrying with TCP\n");
|
|
if (rtsp_read_pause(s) != 0)
|
|
return -1;
|
|
// TEARDOWN is required on Real-RTSP, but might make
|
|
// other servers close the connection.
|
|
if (rt->server_type == RTSP_SERVER_REAL)
|
|
ff_rtsp_send_cmd(s, "TEARDOWN", rt->control_uri, NULL,
|
|
reply, NULL);
|
|
rt->session_id[0] = '\0';
|
|
if (resetup_tcp(s) == 0) {
|
|
rt->state = RTSP_STATE_IDLE;
|
|
rt->need_subscription = 1;
|
|
if (rtsp_read_play(s) != 0)
|
|
return -1;
|
|
goto retry;
|
|
}
|
|
}
|
|
}
|
|
return ret;
|
|
}
|
|
rt->packets++;
|
|
|
|
if (!(rt->rtsp_flags & RTSP_FLAG_LISTEN)) {
|
|
/* send dummy request to keep TCP connection alive */
|
|
if ((av_gettime_relative() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2 ||
|
|
rt->auth_state.stale) {
|
|
if (rt->server_type == RTSP_SERVER_WMS ||
|
|
(rt->server_type != RTSP_SERVER_REAL &&
|
|
rt->get_parameter_supported)) {
|
|
ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL);
|
|
} else {
|
|
ff_rtsp_send_cmd_async(s, "OPTIONS", rt->control_uri, NULL);
|
|
}
|
|
/* The stale flag should be reset when creating the auth response in
|
|
* ff_rtsp_send_cmd_async, but reset it here just in case we never
|
|
* called the auth code (if we didn't have any credentials set). */
|
|
rt->auth_state.stale = 0;
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int rtsp_read_seek(AVFormatContext *s, int stream_index,
|
|
int64_t timestamp, int flags)
|
|
{
|
|
RTSPState *rt = s->priv_data;
|
|
|
|
rt->seek_timestamp = av_rescale_q(timestamp,
|
|
s->streams[stream_index]->time_base,
|
|
AV_TIME_BASE_Q);
|
|
switch(rt->state) {
|
|
default:
|
|
case RTSP_STATE_IDLE:
|
|
break;
|
|
case RTSP_STATE_STREAMING:
|
|
if (rtsp_read_pause(s) != 0)
|
|
return -1;
|
|
rt->state = RTSP_STATE_SEEKING;
|
|
if (rtsp_read_play(s) != 0)
|
|
return -1;
|
|
break;
|
|
case RTSP_STATE_PAUSED:
|
|
rt->state = RTSP_STATE_IDLE;
|
|
break;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static const AVClass rtsp_demuxer_class = {
|
|
.class_name = "RTSP demuxer",
|
|
.item_name = av_default_item_name,
|
|
.option = ff_rtsp_options,
|
|
.version = LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
AVInputFormat ff_rtsp_demuxer = {
|
|
.name = "rtsp",
|
|
.long_name = NULL_IF_CONFIG_SMALL("RTSP input"),
|
|
.priv_data_size = sizeof(RTSPState),
|
|
.read_probe = rtsp_probe,
|
|
.read_header = rtsp_read_header,
|
|
.read_packet = rtsp_read_packet,
|
|
.read_close = rtsp_read_close,
|
|
.read_seek = rtsp_read_seek,
|
|
.flags = AVFMT_NOFILE,
|
|
.read_play = rtsp_read_play,
|
|
.read_pause = rtsp_read_pause,
|
|
.priv_class = &rtsp_demuxer_class,
|
|
};
|