From 45b068580b8c7d65a4422b47f2b98b258cf9587b Mon Sep 17 00:00:00 2001 From: Jordi Ortiz Date: Tue, 10 Jul 2012 19:21:58 +0200 Subject: rtsp: Parse the mode=receive/record parameter in transport lines MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit We need to support the nonstandard mode=receive, for compatibility with older libavformat clients. Signed-off-by: Martin Storsjö --- libavformat/rtsp.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'libavformat/rtsp.h') diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h index e512336ab0..55743b5d2d 100644 --- a/libavformat/rtsp.h +++ b/libavformat/rtsp.h @@ -102,6 +102,9 @@ typedef struct RTSPTransportField { * packets will be allowed to make before being discarded. */ int ttl; + /** transport set to record data */ + int mode_record; + struct sockaddr_storage destination; /**< destination IP address */ char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */ -- cgit v1.2.3 From 6e71c1202bbdca0a95680e07507b39c55bb04f12 Mon Sep 17 00:00:00 2001 From: Jordi Ortiz Date: Tue, 10 Jul 2012 19:25:04 +0200 Subject: rtsp: Make rtsp_open_transport_ctx() non-static MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This is required for the upcoming RTSP listen mode. Signed-off-by: Martin Storsjö --- libavformat/rtsp.c | 6 +++--- libavformat/rtsp.h | 5 +++++ 2 files changed, 8 insertions(+), 3 deletions(-) (limited to 'libavformat/rtsp.h') diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c index 551884ba70..d4206a155e 100644 --- a/libavformat/rtsp.c +++ b/libavformat/rtsp.c @@ -595,7 +595,7 @@ void ff_rtsp_close_streams(AVFormatContext *s) av_free(rt->recvbuf); } -static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st) +int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st) { RTSPState *rt = s->priv_data; AVStream *st = NULL; @@ -1402,7 +1402,7 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, } } - if ((err = rtsp_open_transport_ctx(s, rtsp_st))) + if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st))) goto fail; } @@ -1925,7 +1925,7 @@ static int sdp_read_header(AVFormatContext *s) err = AVERROR_INVALIDDATA; goto fail; } - if ((err = rtsp_open_transport_ctx(s, rtsp_st))) + if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st))) goto fail; } return 0; diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h index 55743b5d2d..41bf8bbb8a 100644 --- a/libavformat/rtsp.h +++ b/libavformat/rtsp.h @@ -561,6 +561,11 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, */ void ff_rtsp_undo_setup(AVFormatContext *s); +/** + * Open RTSP transport context. + */ +int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st); + extern const AVOption ff_rtsp_options[]; #endif /* AVFORMAT_RTSP_H */ -- cgit v1.2.3 From a8ad6ffafe89e3a83f343f69249338e8245816f7 Mon Sep 17 00:00:00 2001 From: Jordi Ortiz Date: Tue, 10 Jul 2012 19:36:11 +0200 Subject: rtsp: Add listen mode MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This makes the RTSP demuxer act as a server, listening for an incoming connection. Signed-off-by: Martin Storsjö --- Changelog | 1 + doc/protocols.texi | 8 + libavformat/rtsp.c | 30 ++- libavformat/rtsp.h | 12 + libavformat/rtspcodes.h | 14 ++ libavformat/rtspdec.c | 573 +++++++++++++++++++++++++++++++++++++++++++++--- libavformat/version.h | 2 +- 7 files changed, 601 insertions(+), 39 deletions(-) (limited to 'libavformat/rtsp.h') diff --git a/Changelog b/Changelog index 39ad8a3fec..66994b41d0 100644 --- a/Changelog +++ b/Changelog @@ -31,6 +31,7 @@ version : - join audio filter - audio channel mapping filter - Microsoft ATC Screen decoder +- RTSP listen mode version 0.8: diff --git a/doc/protocols.texi b/doc/protocols.texi index e75f10838a..943287aa9f 100644 --- a/doc/protocols.texi +++ b/doc/protocols.texi @@ -347,6 +347,8 @@ Flags for @code{rtsp_flags}: @table @option @item filter_src Accept packets only from negotiated peer address and port. +@item listen +Act as a server, listening for an incoming connection. @end table When receiving data over UDP, the demuxer tries to reorder received packets @@ -379,6 +381,12 @@ To send a stream in realtime to a RTSP server, for others to watch: avconv -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp @end example +To receive a stream in realtime: + +@example +avconv -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output} +@end example + @section sap Session Announcement Protocol (RFC 2974). This is not technically a diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c index d4206a155e..f98dc6b4fd 100644 --- a/libavformat/rtsp.c +++ b/libavformat/rtsp.c @@ -63,7 +63,8 @@ #define RTSP_FLAG_OPTS(name, longname) \ { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \ - { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" } + { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }, \ + { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" } #define RTSP_MEDIATYPE_OPTS(name, longname) \ { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \ @@ -83,6 +84,7 @@ const AVOption ff_rtsp_options[] = { RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"), { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC }, { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC }, + { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {-1}, INT_MIN, INT_MAX, DEC }, { NULL }, }; @@ -1714,14 +1716,24 @@ static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, } #if CONFIG_RTSP_DEMUXER if (tcp_fd != -1 && p[0].revents & POLLIN) { - RTSPMessageHeader reply; - - ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL); - if (ret < 0) - return ret; - /* XXX: parse message */ - if (rt->state != RTSP_STATE_STREAMING) - return 0; + if (rt->rtsp_flags & RTSP_FLAG_LISTEN) { + if (rt->state == RTSP_STATE_STREAMING) { + if (!ff_rtsp_parse_streaming_commands(s)) + return AVERROR_EOF; + else + av_log(s, AV_LOG_WARNING, + "Unable to answer to TEARDOWN\n"); + } else + return 0; + } else { + RTSPMessageHeader reply; + ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL); + if (ret < 0) + return ret; + /* XXX: parse message */ + if (rt->state != RTSP_STATE_STREAMING) + return 0; + } } #endif } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) { diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h index 41bf8bbb8a..a738a3d434 100644 --- a/libavformat/rtsp.h +++ b/libavformat/rtsp.h @@ -372,11 +372,17 @@ typedef struct RTSPState { * Minimum and maximum local UDP ports. */ int rtp_port_min, rtp_port_max; + + /** + * Timeout to wait for incoming connections. + */ + int initial_timeout; } RTSPState; #define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets - receive packets only from the right source address and port. */ +#define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */ /** * Describe a single stream, as identified by a single m= line block in the @@ -528,6 +534,12 @@ int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply); */ int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr); +/** + * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in + * listen mode. + */ +int ff_rtsp_parse_streaming_commands(AVFormatContext *s); + /** * Parse an SDP description of streams by populating an RTSPState struct * within the AVFormatContext; also allocate the RTP streams and the diff --git a/libavformat/rtspcodes.h b/libavformat/rtspcodes.h index 63ceb66cfe..31ab33699c 100644 --- a/libavformat/rtspcodes.h +++ b/libavformat/rtspcodes.h @@ -37,4 +37,18 @@ RTSP_STATUS_SERVICE =503, /**< Service Unavailable */ RTSP_STATUS_VERSION =505, /**< RTSP Version not supported */ }; +enum RTSPMethod { + DESCRIBE, + ANNOUNCE, + OPTIONS, + SETUP, + PLAY, + PAUSE, + TEARDOWN, + GET_PARAMETER, + SET_PARAMETER, + REDIRECT, + RECORD, + UNKNOWN = -1, +}; #endif /* AVFORMAT_RTSPCODES_H */ diff --git a/libavformat/rtspdec.c b/libavformat/rtspdec.c index 6226f41660..a3565557d4 100644 --- a/libavformat/rtspdec.c +++ b/libavformat/rtspdec.c @@ -22,6 +22,7 @@ #include "libavutil/avstring.h" #include "libavutil/intreadwrite.h" #include "libavutil/mathematics.h" +#include "libavutil/random_seed.h" #include "avformat.h" #include "internal.h" @@ -31,11 +32,30 @@ #include "rdt.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; - ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL); + 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); @@ -45,6 +65,429 @@ static int rtsp_read_close(AVFormatContext *s) 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; + 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[4096]; + 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_dlog(s, "Sending response:\n%s", message); + ffurl_write(rt->rtsp_hd, 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[1024]; + int rbuflen, ret; + do { + ret = read_line(s, rbuf, sizeof(rbuf), &rbuflen); + if (ret) + return ret; + if (rbuflen > 1) { + av_dlog(s, "Parsing[%d]: %s\n", rbuflen, rbuf); + ff_rtsp_parse_line(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[4096]; + 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[1024]; + RTSPStream *rtsp_st; + char responseheaders[1024]; + 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; + + 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 { + ff_url_join(url, sizeof(url), "rtp", NULL, host, localport, NULL); + av_dlog(s, "Opening: %s", url); + ret = ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE, + &s->interrupt_callback, NULL); + 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_dlog(s, "Listening on: %d", + 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[1024]; + + 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 - 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 == NULL) { + 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); + strcpy(rt->control_uri, 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[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; @@ -157,6 +600,67 @@ int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply) 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); + /* 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)) @@ -169,23 +673,32 @@ static int rtsp_read_header(AVFormatContext *s) RTSPState *rt = s->priv_data; int ret; - 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_timeout > 0) + rt->rtsp_flags |= RTSP_FLAG_LISTEN; - if (rt->initial_pause) { - /* do not start immediately */ + if (rt->rtsp_flags & RTSP_FLAG_LISTEN) { + ret = rtsp_listen(s); + if (ret) + return ret; } else { - if (rtsp_read_play(s) < 0) { - ff_rtsp_close_streams(s); - ff_rtsp_close_connections(s); - return AVERROR_INVALIDDATA; + 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; + } } } @@ -349,20 +862,22 @@ retry: } rt->packets++; - /* 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); + 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; } - /* 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; diff --git a/libavformat/version.h b/libavformat/version.h index 0017698cce..9547bd089e 100644 --- a/libavformat/version.h +++ b/libavformat/version.h @@ -30,7 +30,7 @@ #include "libavutil/avutil.h" #define LIBAVFORMAT_VERSION_MAJOR 54 -#define LIBAVFORMAT_VERSION_MINOR 6 +#define LIBAVFORMAT_VERSION_MINOR 7 #define LIBAVFORMAT_VERSION_MICRO 0 #define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \ -- cgit v1.2.3