blob: bfbb01d586a66e25f432c69bd98e5d4e29a8cb01 [file] [log] [blame]
/*
* RTSP demuxer
* Copyright (c) 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/avstring.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/mathematics.h"
#include "libavutil/random_seed.h"
#include "libavutil/time.h"
#include "avformat.h"
#include "internal.h"
#include "network.h"
#include "os_support.h"
#include "rtpproto.h"
#include "rtsp.h"
#include "rdt.h"
#include "tls.h"
#include "url.h"
static const struct RTSPStatusMessage {
enum RTSPStatusCode code;
const char *message;
} status_messages[] = {
{ RTSP_STATUS_OK, "OK" },
{ RTSP_STATUS_METHOD, "Method Not Allowed" },
{ RTSP_STATUS_BANDWIDTH, "Not Enough Bandwidth" },
{ RTSP_STATUS_SESSION, "Session Not Found" },
{ RTSP_STATUS_STATE, "Method Not Valid in This State" },
{ RTSP_STATUS_AGGREGATE, "Aggregate operation not allowed" },
{ RTSP_STATUS_ONLY_AGGREGATE, "Only aggregate operation allowed" },
{ RTSP_STATUS_TRANSPORT, "Unsupported transport" },
{ RTSP_STATUS_INTERNAL, "Internal Server Error" },
{ RTSP_STATUS_SERVICE, "Service Unavailable" },
{ RTSP_STATUS_VERSION, "RTSP Version not supported" },
{ 0, "NULL" }
};
static int rtsp_read_close(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
if (!(rt->rtsp_flags & RTSP_FLAG_LISTEN))
ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
ff_rtsp_close_streams(s);
ff_rtsp_close_connections(s);
ff_network_close();
rt->real_setup = NULL;
av_freep(&rt->real_setup_cache);
return 0;
}
static inline int read_line(AVFormatContext *s, char *rbuf, const int rbufsize,
int *rbuflen)
{
RTSPState *rt = s->priv_data;
int idx = 0;
int ret = 0;
*rbuflen = 0;
do {
ret = ffurl_read_complete(rt->rtsp_hd, rbuf + idx, 1);
if (ret <= 0)
return ret ? ret : AVERROR_EOF;
if (rbuf[idx] == '\r') {
/* Ignore */
} else if (rbuf[idx] == '\n') {
rbuf[idx] = '\0';
*rbuflen = idx;
return 0;
} else
idx++;
} while (idx < rbufsize);
av_log(s, AV_LOG_ERROR, "Message too long\n");
return AVERROR(EIO);
}
static int rtsp_send_reply(AVFormatContext *s, enum RTSPStatusCode code,
const char *extracontent, uint16_t seq)
{
RTSPState *rt = s->priv_data;
char message[MAX_URL_SIZE];
int index = 0;
while (status_messages[index].code) {
if (status_messages[index].code == code) {
snprintf(message, sizeof(message), "RTSP/1.0 %d %s\r\n",
code, status_messages[index].message);
break;
}
index++;
}
if (!status_messages[index].code)
return AVERROR(EINVAL);
av_strlcatf(message, sizeof(message), "CSeq: %d\r\n", seq);
av_strlcatf(message, sizeof(message), "Server: %s\r\n", LIBAVFORMAT_IDENT);
if (extracontent)
av_strlcat(message, extracontent, sizeof(message));
av_strlcat(message, "\r\n", sizeof(message));
av_log(s, AV_LOG_TRACE, "Sending response:\n%s", message);
ffurl_write(rt->rtsp_hd_out, message, strlen(message));
return 0;
}
static inline int check_sessionid(AVFormatContext *s,
RTSPMessageHeader *request)
{
RTSPState *rt = s->priv_data;
unsigned char *session_id = rt->session_id;
if (!session_id[0]) {
av_log(s, AV_LOG_WARNING, "There is no session-id at the moment\n");
return 0;
}
if (strcmp(session_id, request->session_id)) {
av_log(s, AV_LOG_ERROR, "Unexpected session-id %s\n",
request->session_id);
rtsp_send_reply(s, RTSP_STATUS_SESSION, NULL, request->seq);
return AVERROR_STREAM_NOT_FOUND;
}
return 0;
}
static inline int rtsp_read_request(AVFormatContext *s,
RTSPMessageHeader *request,
const char *method)
{
RTSPState *rt = s->priv_data;
char rbuf[MAX_URL_SIZE];
int rbuflen, ret;
do {
ret = read_line(s, rbuf, sizeof(rbuf), &rbuflen);
if (ret)
return ret;
if (rbuflen > 1) {
av_log(s, AV_LOG_TRACE, "Parsing[%d]: %s\n", rbuflen, rbuf);
ff_rtsp_parse_line(s, request, rbuf, rt, method);
}
} while (rbuflen > 0);
if (request->seq != rt->seq + 1) {
av_log(s, AV_LOG_ERROR, "Unexpected Sequence number %d\n",
request->seq);
return AVERROR(EINVAL);
}
if (rt->session_id[0] && strcmp(method, "OPTIONS")) {
ret = check_sessionid(s, request);
if (ret)
return ret;
}
return 0;
}
static int rtsp_read_announce(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
RTSPMessageHeader request = { 0 };
char sdp[SDP_MAX_SIZE];
int ret;
ret = rtsp_read_request(s, &request, "ANNOUNCE");
if (ret)
return ret;
rt->seq++;
if (strcmp(request.content_type, "application/sdp")) {
av_log(s, AV_LOG_ERROR, "Unexpected content type %s\n",
request.content_type);
rtsp_send_reply(s, RTSP_STATUS_SERVICE, NULL, request.seq);
return AVERROR_OPTION_NOT_FOUND;
}
if (request.content_length && request.content_length < sizeof(sdp) - 1) {
/* Read SDP */
if (ffurl_read_complete(rt->rtsp_hd, sdp, request.content_length)
< request.content_length) {
av_log(s, AV_LOG_ERROR,
"Unable to get complete SDP Description in ANNOUNCE\n");
rtsp_send_reply(s, RTSP_STATUS_INTERNAL, NULL, request.seq);
return AVERROR(EIO);
}
sdp[request.content_length] = '\0';
av_log(s, AV_LOG_VERBOSE, "SDP: %s\n", sdp);
ret = ff_sdp_parse(s, sdp);
if (ret)
return ret;
rtsp_send_reply(s, RTSP_STATUS_OK, NULL, request.seq);
return 0;
}
av_log(s, AV_LOG_ERROR,
"Content-Length header value exceeds sdp allocated buffer (4KB)\n");
rtsp_send_reply(s, RTSP_STATUS_INTERNAL,
"Content-Length exceeds buffer size", request.seq);
return AVERROR(EIO);
}
static int rtsp_read_options(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
RTSPMessageHeader request = { 0 };
int ret = 0;
/* Parsing headers */
ret = rtsp_read_request(s, &request, "OPTIONS");
if (ret)
return ret;
rt->seq++;
/* Send Reply */
rtsp_send_reply(s, RTSP_STATUS_OK,
"Public: ANNOUNCE, PAUSE, SETUP, TEARDOWN, RECORD\r\n",
request.seq);
return 0;
}
static int rtsp_read_setup(AVFormatContext *s, char* host, char *controlurl)
{
RTSPState *rt = s->priv_data;
RTSPMessageHeader request = { 0 };
int ret = 0;
char url[MAX_URL_SIZE];
RTSPStream *rtsp_st;
char responseheaders[MAX_URL_SIZE];
int localport = -1;
int transportidx = 0;
int streamid = 0;
ret = rtsp_read_request(s, &request, "SETUP");
if (ret)
return ret;
rt->seq++;
if (!request.nb_transports) {
av_log(s, AV_LOG_ERROR, "No transport defined in SETUP\n");
return AVERROR_INVALIDDATA;
}
for (transportidx = 0; transportidx < request.nb_transports;
transportidx++) {
if (!request.transports[transportidx].mode_record ||
(request.transports[transportidx].lower_transport !=
RTSP_LOWER_TRANSPORT_UDP &&
request.transports[transportidx].lower_transport !=
RTSP_LOWER_TRANSPORT_TCP)) {
av_log(s, AV_LOG_ERROR, "mode=record/receive not set or transport"
" protocol not supported (yet)\n");
return AVERROR_INVALIDDATA;
}
}
if (request.nb_transports > 1)
av_log(s, AV_LOG_WARNING, "More than one transport not supported, "
"using first of all\n");
for (streamid = 0; streamid < rt->nb_rtsp_streams; streamid++) {
if (!strcmp(rt->rtsp_streams[streamid]->control_url,
controlurl))
break;
}
if (streamid == rt->nb_rtsp_streams) {
av_log(s, AV_LOG_ERROR, "Unable to find requested track\n");
return AVERROR_STREAM_NOT_FOUND;
}
rtsp_st = rt->rtsp_streams[streamid];
localport = rt->rtp_port_min;
/* check if the stream has already been setup */
if (rtsp_st->transport_priv) {
if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
ff_rdt_parse_close(rtsp_st->transport_priv);
else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
ff_rtp_parse_close(rtsp_st->transport_priv);
rtsp_st->transport_priv = NULL;
}
if (rtsp_st->rtp_handle)
ffurl_closep(&rtsp_st->rtp_handle);
if (request.transports[0].lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
rt->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
if ((ret = ff_rtsp_open_transport_ctx(s, rtsp_st))) {
rtsp_send_reply(s, RTSP_STATUS_TRANSPORT, NULL, request.seq);
return ret;
}
rtsp_st->interleaved_min = request.transports[0].interleaved_min;
rtsp_st->interleaved_max = request.transports[0].interleaved_max;
snprintf(responseheaders, sizeof(responseheaders), "Transport: "
"RTP/AVP/TCP;unicast;mode=receive;interleaved=%d-%d"
"\r\n", request.transports[0].interleaved_min,
request.transports[0].interleaved_max);
} else {
do {
AVDictionary *opts = NULL;
av_dict_set_int(&opts, "buffer_size", rt->buffer_size, 0);
ff_url_join(url, sizeof(url), "rtp", NULL, host, localport, NULL);
av_log(s, AV_LOG_TRACE, "Opening: %s\n", url);
ret = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
&s->interrupt_callback, &opts,
s->protocol_whitelist, s->protocol_blacklist, NULL);
av_dict_free(&opts);
if (ret)
localport += 2;
} while (ret || localport > rt->rtp_port_max);
if (localport > rt->rtp_port_max) {
rtsp_send_reply(s, RTSP_STATUS_TRANSPORT, NULL, request.seq);
return ret;
}
av_log(s, AV_LOG_TRACE, "Listening on: %d\n",
ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle));
if ((ret = ff_rtsp_open_transport_ctx(s, rtsp_st))) {
rtsp_send_reply(s, RTSP_STATUS_TRANSPORT, NULL, request.seq);
return ret;
}
localport = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
snprintf(responseheaders, sizeof(responseheaders), "Transport: "
"RTP/AVP/UDP;unicast;mode=receive;source=%s;"
"client_port=%d-%d;server_port=%d-%d\r\n",
host, request.transports[0].client_port_min,
request.transports[0].client_port_max, localport,
localport + 1);
}
/* Establish sessionid if not previously set */
/* Put this in a function? */
/* RFC 2326: session id must be at least 8 digits */
while (strlen(rt->session_id) < 8)
av_strlcatf(rt->session_id, 512, "%u", av_get_random_seed());
av_strlcatf(responseheaders, sizeof(responseheaders), "Session: %s\r\n",
rt->session_id);
/* Send Reply */
rtsp_send_reply(s, RTSP_STATUS_OK, responseheaders, request.seq);
rt->state = RTSP_STATE_PAUSED;
return 0;
}
static int rtsp_read_record(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
RTSPMessageHeader request = { 0 };
int ret = 0;
char responseheaders[MAX_URL_SIZE];
ret = rtsp_read_request(s, &request, "RECORD");
if (ret)
return ret;
ret = check_sessionid(s, &request);
if (ret)
return ret;
rt->seq++;
snprintf(responseheaders, sizeof(responseheaders), "Session: %s\r\n",
rt->session_id);
rtsp_send_reply(s, RTSP_STATUS_OK, responseheaders, request.seq);
rt->state = RTSP_STATE_STREAMING;
return 0;
}
static inline int parse_command_line(AVFormatContext *s, const char *line,
int linelen, char *uri, int urisize,
char *method, int methodsize,
enum RTSPMethod *methodcode)
{
RTSPState *rt = s->priv_data;
const char *linept, *searchlinept;
linept = strchr(line, ' ');
if (!linept) {
av_log(s, AV_LOG_ERROR, "Error parsing method string\n");
return AVERROR_INVALIDDATA;
}
if (linept - line > methodsize - 1) {
av_log(s, AV_LOG_ERROR, "Method string too long\n");
return AVERROR(EIO);
}
memcpy(method, line, linept - line);
method[linept - line] = '\0';
linept++;
if (!strcmp(method, "ANNOUNCE"))
*methodcode = ANNOUNCE;
else if (!strcmp(method, "OPTIONS"))
*methodcode = OPTIONS;
else if (!strcmp(method, "RECORD"))
*methodcode = RECORD;
else if (!strcmp(method, "SETUP"))
*methodcode = SETUP;
else if (!strcmp(method, "PAUSE"))
*methodcode = PAUSE;
else if (!strcmp(method, "TEARDOWN"))
*methodcode = TEARDOWN;
else
*methodcode = UNKNOWN;
/* Check method with the state */
if (rt->state == RTSP_STATE_IDLE) {
if ((*methodcode != ANNOUNCE) && (*methodcode != OPTIONS)) {
av_log(s, AV_LOG_ERROR, "Unexpected command in Idle State %s\n",
line);
return AVERROR_PROTOCOL_NOT_FOUND;
}
} else if (rt->state == RTSP_STATE_PAUSED) {
if ((*methodcode != OPTIONS) && (*methodcode != RECORD)
&& (*methodcode != SETUP)) {
av_log(s, AV_LOG_ERROR, "Unexpected command in Paused State %s\n",
line);
return AVERROR_PROTOCOL_NOT_FOUND;
}
} else if (rt->state == RTSP_STATE_STREAMING) {
if ((*methodcode != PAUSE) && (*methodcode != OPTIONS)
&& (*methodcode != TEARDOWN)) {
av_log(s, AV_LOG_ERROR, "Unexpected command in Streaming State"
" %s\n", line);
return AVERROR_PROTOCOL_NOT_FOUND;
}
} else {
av_log(s, AV_LOG_ERROR, "Unexpected State [%d]\n", rt->state);
return AVERROR_BUG;
}
searchlinept = strchr(linept, ' ');
if (!searchlinept) {
av_log(s, AV_LOG_ERROR, "Error parsing message URI\n");
return AVERROR_INVALIDDATA;
}
if (searchlinept - linept > urisize - 1) {
av_log(s, AV_LOG_ERROR, "uri string length exceeded buffer size\n");
return AVERROR(EIO);
}
memcpy(uri, linept, searchlinept - linept);
uri[searchlinept - linept] = '\0';
if (strcmp(rt->control_uri, uri)) {
char host[128], path[512], auth[128];
int port;
char ctl_host[128], ctl_path[512], ctl_auth[128];
int ctl_port;
av_url_split(NULL, 0, auth, sizeof(auth), host, sizeof(host), &port,
path, sizeof(path), uri);
av_url_split(NULL, 0, ctl_auth, sizeof(ctl_auth), ctl_host,
sizeof(ctl_host), &ctl_port, ctl_path, sizeof(ctl_path),
rt->control_uri);
if (strcmp(host, ctl_host))
av_log(s, AV_LOG_INFO, "Host %s differs from expected %s\n",
host, ctl_host);
if (strcmp(path, ctl_path) && *methodcode != SETUP)
av_log(s, AV_LOG_WARNING, "WARNING: Path %s differs from expected"
" %s\n", path, ctl_path);
if (*methodcode == ANNOUNCE) {
av_log(s, AV_LOG_INFO,
"Updating control URI to %s\n", uri);
av_strlcpy(rt->control_uri, uri, sizeof(rt->control_uri));
}
}
linept = searchlinept + 1;
if (!av_strstart(linept, "RTSP/1.0", NULL)) {
av_log(s, AV_LOG_ERROR, "Error parsing protocol or version\n");
return AVERROR_PROTOCOL_NOT_FOUND;
}
return 0;
}
int ff_rtsp_parse_streaming_commands(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
unsigned char rbuf[MAX_URL_SIZE];
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;
av_log(s, AV_LOG_TRACE, "Parsing[%d]: %s\n", rbuflen, rbuf);
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 ret;
}
static int rtsp_read_play(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
RTSPMessageHeader reply1, *reply = &reply1;
int i;
char cmd[MAX_URL_SIZE];
av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
rt->nb_byes = 0;
if (rt->lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
for (i = 0; i < rt->nb_rtsp_streams; i++) {
RTSPStream *rtsp_st = rt->rtsp_streams[i];
/* Try to initialize the connection state in a
* potential NAT router by sending dummy packets.
* RTP/RTCP dummy packets are used for RDT, too.
*/
if (rtsp_st->rtp_handle &&
!(rt->server_type == RTSP_SERVER_WMS && i > 1))
ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
}
}
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 ff_rtsp_averror(reply->status_code, -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 ff_rtsp_averror(reply->status_code, -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[MAX_URL_SIZE];
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 (reply->status_code != RTSP_STATUS_OK) {
av_freep(&content);
return ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
}
if (!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[MAX_URL_SIZE];
unsigned char method[10];
int rbuflen = 0;
int ret;
enum RTSPMethod methodcode;
if (!ff_network_init())
return AVERROR(EIO);
/* extract hostname and port */
av_url_split(proto, sizeof(proto), auth, sizeof(auth), host, sizeof(host),
&port, path, sizeof(path), s->url);
/* 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_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
&s->interrupt_callback, NULL,
s->protocol_whitelist, s->protocol_blacklist, NULL)) {
av_log(s, AV_LOG_ERROR, "Unable to open RTSP for listening\n");
goto fail;
}
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)
goto fail;
av_log(s, AV_LOG_TRACE, "Parsing[%d]: %s\n", rbuflen, rbuf);
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");
goto fail;
}
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) {
ret = AVERROR_INVALIDDATA;
goto fail;
}
}
fail:
ff_rtsp_close_streams(s);
ff_rtsp_close_connections(s);
ff_network_close();
return ret;
}
static int rtsp_probe(const 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_array(s->nb_streams, 2 * sizeof(*rt->real_setup_cache));
if (!rt->real_setup_cache && s->nb_streams) {
ret = AVERROR(ENOMEM);
goto fail;
}
rt->real_setup = rt->real_setup_cache + s->nb_streams;
if (rt->initial_pause) {
/* do not start immediately */
} else {
ret = rtsp_read_play(s);
if (ret < 0)
goto fail;
}
}
return 0;
fail:
rtsp_read_close(s);
return ret;
}
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_log(s, AV_LOG_TRACE, "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 AVERROR(EIO);
id = buf[0];
len = AV_RB16(buf + 1);
av_log(s, AV_LOG_TRACE, "id=%d len=%d\n", id, len);
if (len > buf_size || len < 8)
goto redo;
/* get the data */
ret = ffurl_read_complete(rt->rtsp_hd, buf, len);
if (ret != len)
return AVERROR(EIO);
if (rt->transport == RTSP_TRANSPORT_RDT &&
(ret = ff_rdt_parse_header(buf, len, &id, NULL, NULL, NULL, NULL)) < 0)
return ret;
/* 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->url);
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[MAX_URL_SIZE];
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 ff_rtsp_averror(reply->status_code, 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 ff_rtsp_averror(reply->status_code, 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;
int ret;
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 ((ret = rtsp_read_pause(s)) != 0)
return ret;
rt->state = RTSP_STATE_SEEKING;
if ((ret = rtsp_read_play(s)) != 0)
return ret;
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,
};