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https://gitee.com/openharmony/third_party_ffmpeg
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b7e6da988b
Mixing these with the rtp depacketizer functions in rtpdec.h is no good. Signed-off-by: Martin Storsjö <martin@martin.st>
939 lines
32 KiB
C
939 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 - 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 == NULL) {
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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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}
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return 0;
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}
|
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|
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int ff_rtsp_parse_streaming_commands(AVFormatContext *s)
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|
{
|
|
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 host[128], path[512], auth[128];
|
|
char uri[500];
|
|
int port;
|
|
char tcpname[500];
|
|
unsigned char rbuf[4096];
|
|
unsigned char method[10];
|
|
int rbuflen = 0;
|
|
int ret;
|
|
enum RTSPMethod methodcode;
|
|
|
|
/* extract hostname and port */
|
|
av_url_split(NULL, 0, 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), "rtsp", NULL, host,
|
|
port, "%s", path);
|
|
|
|
if (port < 0)
|
|
port = RTSP_DEFAULT_PORT;
|
|
|
|
/* Create TCP connection */
|
|
ff_url_join(tcpname, sizeof(tcpname), "tcp", 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 (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);
|
|
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() - 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", "*", 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,
|
|
};
|