[FFmpeg-cvslog] rtsp: Add listen mode
Jordi Ortiz
git at videolan.org
Thu Jul 12 00:03:06 CEST 2012
ffmpeg | branch: master | Jordi Ortiz <nenjordi at gmail.com> | Tue Jul 10 19:36:11 2012 +0200| [a8ad6ffafe89e3a83f343f69249338e8245816f7] | committer: Martin Storsjö
rtsp: Add listen mode
This makes the RTSP demuxer act as a server, listening for an
incoming connection.
Signed-off-by: Martin Storsjö <martin at martin.st>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=a8ad6ffafe89e3a83f343f69249338e8245816f7
---
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(-)
diff --git a/Changelog b/Changelog
index 39ad8a3..66994b4 100644
--- a/Changelog
+++ b/Changelog
@@ -31,6 +31,7 @@ version <next>:
- 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 e75f108..943287a 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.
+ at 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:
+
+ at example
+avconv -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
+ at 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 d4206a1..f98dc6b 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 41bf8bb..a738a3d 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
@@ -529,6 +535,12 @@ 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
* pollfd array used for UDP streams.
diff --git a/libavformat/rtspcodes.h b/libavformat/rtspcodes.h
index 63ceb66..31ab336 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 6226f41..a356555 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 0017698..9547bd0 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, \
More information about the ffmpeg-cvslog
mailing list