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rtsp: Start implementing support for RTSP 2.0

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- Handle version negotation:

  Added a `default-version` property so that the user can configure
  what to use in case the server does not support version negotation
  (which actually exist)

- Handle pipelined requests, which allow avoiding full round trip to
  setup the RTP streams (request are sent in a raw, and response are
  handled as they arrive).

- Handle the new Media-Properties header

- Handle the new Seek-Style header

- Handle the new Accept-Ranges header

Handling of IPV6 should already be OK.

We are still missing (at least) the following features (which do not
seem really mandatory as they require a "persistent connection between
server and client"):

  - Server to Client TEARDOWN command (Not so usefull fmpov)
  - PLAY_NOTIFY (not needed for our server yet)
  - Support for the new REDIRECT features

and probably some more protocol changes might not be handled yet.

https://bugzilla.gnome.org/show_bug.cgi?id=781446
This commit is contained in:
Thibault Saunier 2017-04-20 17:45:39 -03:00
parent 5faad79049
commit 8121752887
2 changed files with 416 additions and 178 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

587
gst/rtsp/gstrtspsrc.c
View file

@ -233,6 +233,7 @@ gst_rtsp_src_ntp_time_source_get_type (void)
#define DEFAULT_RFC7273_SYNC FALSE
#define DEFAULT_MAX_TS_OFFSET_ADJUSTMENT 0
#define DEFAULT_MAX_TS_OFFSET 3000000000
#define DEFAULT_VERSION GST_RTSP_VERSION_1_0
enum
{
@ -274,7 +275,8 @@ enum
PROP_MAX_RTCP_RTP_TIME_DIFF,
PROP_RFC7273_SYNC,
PROP_MAX_TS_OFFSET_ADJUSTMENT,
PROP_MAX_TS_OFFSET
PROP_MAX_TS_OFFSET,
PROP_DEFAULT_VERSION,
};
#define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
@ -333,7 +335,7 @@ static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
gboolean async);
gboolean async, const gchar * seek_style);
static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async);
static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
gboolean only_close);
@ -755,6 +757,13 @@ gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
"(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_DEFAULT_VERSION,
g_param_spec_enum ("default-rtsp-version",
"The RTSP version to try first",
"The RTSP version that should be tried first when negotiating version.",
GST_TYPE_RTSP_VERSION, DEFAULT_VERSION,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPSrc:max-ts-offset-adjustment:
*
@ -936,6 +945,8 @@ gst_rtspsrc_init (GstRTSPSrc * src)
src->max_ts_offset_adjustment = DEFAULT_MAX_TS_OFFSET_ADJUSTMENT;
src->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
src->max_ts_offset_is_set = FALSE;
src->default_version = DEFAULT_VERSION;
src->version = GST_RTSP_VERSION_INVALID;
/* get a list of all extensions */
src->extensions = gst_rtsp_ext_list_get ();
@ -1242,6 +1253,9 @@ gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
rtspsrc->max_ts_offset = g_value_get_int64 (value);
rtspsrc->max_ts_offset_is_set = TRUE;
break;
case PROP_DEFAULT_VERSION:
rtspsrc->default_version = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@ -1397,6 +1411,9 @@ gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
case PROP_MAX_TS_OFFSET:
g_value_set_int64 (value, rtspsrc->max_ts_offset);
break;
case PROP_DEFAULT_VERSION:
g_value_set_enum (value, rtspsrc->default_version);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@ -2227,6 +2244,7 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
gboolean playing;
GstSegment seeksegment = { 0, };
GList *walk;
gchar *seek_style = NULL;
if (event) {
GST_DEBUG_OBJECT (src, "doing seek with event");
@ -2341,8 +2359,21 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
&& GST_STATE (src) == GST_STATE_PLAYING)
|| (GST_STATE_PENDING (src) == GST_STATE_PLAYING);
GST_OBJECT_UNLOCK (src);
if (src->version >= GST_RTSP_VERSION_2_0) {
if (flags & GST_SEEK_FLAG_ACCURATE)
seek_style = "RAP";
else if (flags & GST_SEEK_FLAG_KEY_UNIT)
seek_style = "CoRAP";
else if (flags & GST_SEEK_FLAG_KEY_UNIT
&& flags & GST_SEEK_FLAG_SNAP_BEFORE)
seek_style = "First-Prior";
else if (flags & GST_SEEK_FLAG_KEY_UNIT && flags & GST_SEEK_FLAG_SNAP_AFTER)
seek_style = "Next";
}
if (playing)
gst_rtspsrc_play (src, &seeksegment, FALSE);
gst_rtspsrc_play (src, &seeksegment, FALSE, seek_style);
GST_RTSP_STREAM_UNLOCK (src);
@ -2540,13 +2571,24 @@ gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
if (format == GST_FORMAT_TIME) {
gboolean seekable =
src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
GstClockTime start, duration = src->segment.duration;
/* seeking without duration is unlikely */
seekable = seekable && src->seekable && src->segment.duration &&
seekable = seekable && src->seekable >= 0.0 && src->segment.duration &&
GST_CLOCK_TIME_IS_VALID (src->segment.duration);
gst_query_set_seeking (query, GST_FORMAT_TIME, seekable, 0,
src->segment.duration);
if (seekable) {
if (src->seekable > 0.0) {
start = src->last_pos - src->seekable * GST_SECOND;
} else {
/* src->seekable == 0 means that we can only seek to 0 */
start = 0;
duration = 0;
}
}
gst_query_set_seeking (query, GST_FORMAT_TIME, seekable, start,
duration);
res = TRUE;
}
break;
@ -4500,8 +4542,7 @@ gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
if (res < 0)
goto send_error;
if (src->debug)
gst_rtsp_message_dump (&request);
request.type_data.request.version = src->version;
res = gst_rtspsrc_connection_send (src, &src->conninfo, &request, NULL);
if (res < 0)
@ -5074,7 +5115,7 @@ gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
goto open_failed;
/* start playback */
if (gst_rtspsrc_play (src, &src->segment, async) < 0)
if (gst_rtspsrc_play (src, &src->segment, async, NULL) < 0)
goto play_failed;
done:
@ -5481,33 +5522,14 @@ no_user_pass:
}
static GstRTSPResult
gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
GstRTSPMessage * request, GstRTSPMessage * response,
GstRTSPStatusCode * code)
gst_rtsp_src_receive_response (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
GstRTSPMessage * response, GstRTSPStatusCode * code)
{
GstRTSPResult res;
GstRTSPStatusCode thecode;
gchar *content_base = NULL;
gint try = 0;
GstRTSPResult res = gst_rtspsrc_connection_receive (src, conninfo,
response, src->ptcp_timeout);
again:
if (!src->short_header)
gst_rtsp_ext_list_before_send (src->extensions, request);
GST_DEBUG_OBJECT (src, "sending message");
DEBUG_RTSP (src, request);
res = gst_rtspsrc_connection_send (src, conninfo, request, src->ptcp_timeout);
if (res < 0)
goto send_error;
gst_rtsp_connection_reset_timeout (conninfo->connection);
next:
res =
gst_rtspsrc_connection_receive (src, conninfo, response,
src->ptcp_timeout);
if (res < 0)
goto receive_error;
@ -5520,7 +5542,9 @@ next:
goto server_eof;
else if (res < 0)
goto handle_request_failed;
goto next;
/* Not a response, receive next message */
return gst_rtsp_src_receive_response (src, conninfo, response, code);
case GST_RTSP_MESSAGE_RESPONSE:
/* ok, a response is good */
GST_DEBUG_OBJECT (src, "received response message");
@ -5529,11 +5553,15 @@ next:
/* get next response */
GST_DEBUG_OBJECT (src, "handle data response message");
gst_rtspsrc_handle_data (src, response);
goto next;
/* Not a response, receive next message */
return gst_rtsp_src_receive_response (src, conninfo, response, code);
default:
GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
response->type);
goto next;
/* Not a response, receive next message */
return gst_rtsp_src_receive_response (src, conninfo, response, code);
}
thecode = response->type_data.response.code;
@ -5555,38 +5583,15 @@ next:
g_free (src->content_base);
src->content_base = g_strdup (content_base);
}
gst_rtsp_ext_list_after_send (src->extensions, request, response);
return GST_RTSP_OK;
/* ERRORS */
send_error:
{
gchar *str = gst_rtsp_strresult (res);
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
} else {
GST_WARNING_OBJECT (src, "send interrupted");
}
g_free (str);
return res;
}
receive_error:
{
switch (res) {
case GST_RTSP_EEOF:
GST_WARNING_OBJECT (src, "server closed connection");
if ((try == 0) && !src->interleaved && src->udp_reconnect) {
try++;
/* if reconnect succeeds, try again */
if ((res =
gst_rtsp_conninfo_reconnect (src, &src->conninfo,
FALSE)) == 0)
goto again;
}
/* only try once after reconnect, then fallthrough and error out */
return GST_RTSP_EEOF;
default:
{
gchar *str = gst_rtsp_strresult (res);
@ -5619,13 +5624,70 @@ server_eof:
}
}
static GstRTSPResult
gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
GstRTSPMessage * request, GstRTSPMessage * response,
GstRTSPStatusCode * code)
{
GstRTSPResult res;
gint try = 0;
again:
if (!src->short_header)
gst_rtsp_ext_list_before_send (src->extensions, request);
GST_DEBUG_OBJECT (src, "sending message");
DEBUG_RTSP (src, request);
res = gst_rtspsrc_connection_send (src, conninfo, request, src->ptcp_timeout);
if (res < 0)
goto send_error;
gst_rtsp_connection_reset_timeout (conninfo->connection);
if (!response)
return res;
res = gst_rtsp_src_receive_response (src, conninfo, response, code);
if (res == GST_RTSP_EEOF) {
GST_WARNING_OBJECT (src, "server closed connection");
/* only try once after reconnect, then fallthrough and error out */
if ((try == 0) && !src->interleaved && src->udp_reconnect) {
try++;
/* if reconnect succeeds, try again */
if ((res = gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) == 0)
goto again;
}
}
gst_rtsp_ext_list_after_send (src->extensions, request, response);
return res;
send_error:
{
gchar *str = gst_rtsp_strresult (res);
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
} else {
GST_WARNING_OBJECT (src, "send interrupted");
}
g_free (str);
return res;
}
}
/**
* gst_rtspsrc_send:
* @src: the rtsp source
* @conn: the connection to send on
* @conninfo: the connection information to send on
* @request: must point to a valid request
* @response: must point to an empty #GstRTSPMessage
* @code: an optional code result
* @versions: List of versions to try, setting it back onto the @request message
* if not set, `src->version` will be used as RTSP version.
*
* send @request and retrieve the response in @response. optionally @code can be
* non-NULL in which case it will contain the status code of the response.
@ -5645,13 +5707,14 @@ server_eof:
static GstRTSPResult
gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
GstRTSPMessage * request, GstRTSPMessage * response,
GstRTSPStatusCode * code)
GstRTSPStatusCode * code, GstRTSPVersion * versions)
{
GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
GstRTSPResult res = GST_RTSP_ERROR;
gint count;
gboolean retry;
GstRTSPMethod method = GST_RTSP_INVALID;
gint version_retry = 0;
count = 0;
do {
@ -5664,6 +5727,9 @@ gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
/* save method so we can disable it when the server complains */
method = request->type_data.request.method;
if (!versions)
request->type_data.request.version = src->version;
if ((res =
gst_rtspsrc_try_send (src, conninfo, request, response,
&int_code)) < 0)
@ -5678,6 +5744,19 @@ gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
retry = TRUE;
}
break;
case GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED:
GST_INFO_OBJECT (src, "Version %s not supported by the server",
gst_rtsp_version_as_text (versions[version_retry]));
if (versions && versions[version_retry] != GST_RTSP_VERSION_INVALID) {
GST_INFO_OBJECT (src, "Unsupported version %s => trying %s",
gst_rtsp_version_as_text (request->type_data.request.version),
gst_rtsp_version_as_text (versions[version_retry]));
request->type_data.request.version = versions[version_retry];
retry = TRUE;
version_retry++;
break;
}
/* falltrough */
default:
break;
}
@ -5769,7 +5848,7 @@ static GstRTSPResult
gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
GstRTSPMessage * response, GstRTSPSrc * src)
{
return gst_rtspsrc_send (src, &src->conninfo, request, response, NULL);
return gst_rtspsrc_send (src, &src->conninfo, request, response, NULL, NULL);
}
@ -5792,11 +5871,19 @@ gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
while (TRUE) {
respoptions = NULL;
gst_rtsp_message_get_header (response, field, &respoptions, indx);
if (indx == 0 && !respoptions) {
/* if no Allow header was found then try the Public header... */
field = GST_RTSP_HDR_PUBLIC;
gst_rtsp_message_get_header (response, field, &respoptions, indx);
}
if (!respoptions)
break;
src->methods |= gst_rtsp_options_from_text (respoptions);
indx++;
}
indx = 0;
field = GST_RTSP_HDR_PUBLIC;
while (TRUE) {
respoptions = NULL;
gst_rtsp_message_get_header (response, field, &respoptions, indx);
if (!respoptions)
break;
@ -5816,7 +5903,7 @@ gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
* this */
src->methods |= GST_RTSP_PLAY;
/* also assume it will support Range */
src->seekable = TRUE;
src->seekable = G_MAXDOUBLE;
/* we need describe and setup */
if (!(src->methods & GST_RTSP_DESCRIBE))
@ -6086,6 +6173,170 @@ gst_rtspsrc_stream_make_keymgmt (GstRTSPSrc * src, GstRTSPStream * stream)
return result;
}
static GstRTSPResult
gst_rtsp_src_setup_stream_from_response (GstRTSPSrc * src,
GstRTSPStream * stream, GstRTSPMessage * response,
GstRTSPLowerTrans * protocols, gint retry, gint * rtpport, gint * rtcpport)
{
gchar *resptrans = NULL;
GstRTSPTransport transport = { 0 };
gst_rtsp_message_get_header (response, GST_RTSP_HDR_TRANSPORT, &resptrans, 0);
if (!resptrans) {
gst_rtspsrc_stream_free_udp (stream);
goto no_transport;
}
/* parse transport, go to next stream on parse error */
if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans);
return GST_RTSP_ELAST;
}
/* update allowed transports for other streams. once the transport of
* one stream has been determined, we make sure that all other streams
* are configured in the same way */
switch (transport.lower_transport) {
case GST_RTSP_LOWER_TRANS_TCP:
GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
if (protocols)
*protocols = GST_RTSP_LOWER_TRANS_TCP;
src->interleaved = TRUE;
/* update free channels */
src->free_channel = MAX (transport.interleaved.min, src->free_channel);
src->free_channel = MAX (transport.interleaved.max, src->free_channel);
src->free_channel++;
break;
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
/* only allow multicast for other streams */
GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
if (protocols)
*protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
/* if the server selected our ports, increment our counters so that
* we select a new port later */
if (src->next_port_num == transport.port.min &&
src->next_port_num + 1 == transport.port.max) {
src->next_port_num += 2;
}
break;
case GST_RTSP_LOWER_TRANS_UDP:
/* only allow unicast for other streams */
GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
if (protocols)
*protocols = GST_RTSP_LOWER_TRANS_UDP;
break;
default:
GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
transport.lower_transport);
break;
}
if (!src->interleaved || !retry) {
/* now configure the stream with the selected transport */
if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
GST_DEBUG_OBJECT (src,
"could not configure stream %p transport, skipping stream", stream);
goto done;
} else if (stream->udpsrc[0] && stream->udpsrc[1] && rtpport && rtcpport) {
/* retain the first allocated UDP port pair */
g_object_get (G_OBJECT (stream->udpsrc[0]), "port", rtpport, NULL);
g_object_get (G_OBJECT (stream->udpsrc[1]), "port", rtcpport, NULL);
}
}
/* we need to activate at least one stream when we detect activity */
src->need_activate = TRUE;
/* stream is setup now */
stream->setup = TRUE;
stream->waiting_setup_response = FALSE;
if (src->version >= GST_RTSP_VERSION_2_0) {
gchar *prop, *media_properties;
gchar **props;
gint i;
if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_MEDIA_PROPERTIES,
&media_properties, 0) != GST_RTSP_OK) {
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Error: No MEDIA_PROPERTY header in a SETUP request in RTSP 2.0"
" - this header is mandatory."));
gst_rtsp_message_unset (response);
return GST_RTSP_ERROR;
}
props = g_strsplit (media_properties, ",", -2);
for (i = 0; props[i]; i++) {
prop = props[i];
while (*prop == ' ')
prop++;
if (strstr (prop, "Random-Access")) {
gchar **random_seekable_val = g_strsplit (prop, "=", 2);
if (!random_seekable_val[1])
src->seekable = G_MAXDOUBLE;
else
src->seekable = g_ascii_strtod (random_seekable_val[1], NULL);
g_strfreev (random_seekable_val);
} else if (!g_strcmp0 (prop, "No-Seeking")) {
src->seekable = -1.0;
} else if (!g_strcmp0 (prop, "Beginning-Only")) {
src->seekable = 0.0;
}
}
g_strfreev (props);
}
done:
/* clean up our transport struct */
gst_rtsp_transport_init (&transport);
/* clean up used RTSP messages */
gst_rtsp_message_unset (response);
return GST_RTSP_OK;
no_transport:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Server did not select transport."));
gst_rtsp_message_unset (response);
return GST_RTSP_ERROR;
}
}
static GstRTSPResult
gst_rtspsrc_setup_streams_end (GstRTSPSrc * src, gboolean async)
{
GList *tmp;
GstRTSPConnInfo *conninfo;
g_assert (src->version >= GST_RTSP_VERSION_2_0);
conninfo = &src->conninfo;
for (tmp = src->streams; tmp; tmp = tmp->next) {
GstRTSPStream *stream = (GstRTSPStream *) tmp->data;
GstRTSPMessage response = { 0, };
if (!stream->waiting_setup_response)
continue;
if (!src->conninfo.connection)
conninfo = &((GstRTSPStream *) tmp->data)->conninfo;
gst_rtsp_src_receive_response (src, conninfo, &response, NULL);
gst_rtsp_src_setup_stream_from_response (src, stream,
&response, NULL, 1, NULL, NULL);
}
return GST_RTSP_OK;
}
/* Perform the SETUP request for all the streams.
*
* We ask the server for a specific transport, which initially includes all the
@ -6095,11 +6346,14 @@ gst_rtspsrc_stream_make_keymgmt (GstRTSPSrc * src, GstRTSPStream * stream)
* Once the server replied with a transport, we configure the other streams
* with the same transport.
*
* This function will also configure the stream for the selected transport,
* which basically means creating the pipeline.
* In case setup request are not pipelined, this function will also configure the
* stream for the selected transport, * which basically means creating the pipeline.
* Otherwise, the first stream is setup right away from the reply and a
* CMD_FINALIZE_SETUP command is set for the stream pipelines to happen on the
* remaining streams from the RTSP thread.
*/
static GstRTSPResult
gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
gst_rtspsrc_setup_streams_start (GstRTSPSrc * src, gboolean async)
{
GList *walk;
GstRTSPResult res = GST_RTSP_ERROR;
@ -6112,6 +6366,7 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
gint rtpport, rtcpport;
GstRTSPUrl *url;
gchar *hval;
gchar *pipelined_request_id = NULL;
if (src->conninfo.connection) {
url = gst_rtsp_connection_get_url (src->conninfo.connection);
@ -6260,6 +6515,17 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
goto create_request_failed;
}
if (src->version >= GST_RTSP_VERSION_2_0) {
if (!pipelined_request_id)
pipelined_request_id = g_strdup_printf ("%d",
g_random_int_range (0, G_MAXINT32));
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_PIPELINED_REQUESTS,
pipelined_request_id);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT_RANGES,
"npt, clock, smpte, clock");
}
/* select transport */
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
@ -6282,7 +6548,9 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
stream->id));
/* handle the code ourselves */
res = gst_rtspsrc_send (src, conninfo, &request, &response, &code);
res =
gst_rtspsrc_send (src, conninfo, &request,
pipelined_request_id ? NULL : &response, &code, NULL);
if (res < 0)
goto send_error;
@ -6322,106 +6590,50 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
goto response_error;
}
/* parse response transport */
{
gchar *resptrans = NULL;
GstRTSPTransport transport = { 0 };
gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
&resptrans, 0);
if (!resptrans) {
gst_rtspsrc_stream_free_udp (stream);
goto no_transport;
}
/* parse transport, go to next stream on parse error */
if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans);
goto next;
}
/* update allowed transports for other streams. once the transport of
* one stream has been determined, we make sure that all other streams
* are configured in the same way */
switch (transport.lower_transport) {
case GST_RTSP_LOWER_TRANS_TCP:
GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
protocols = GST_RTSP_LOWER_TRANS_TCP;
src->interleaved = TRUE;
/* update free channels */
src->free_channel =
MAX (transport.interleaved.min, src->free_channel);
src->free_channel =
MAX (transport.interleaved.max, src->free_channel);
src->free_channel++;
break;
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
/* only allow multicast for other streams */
GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
/* if the server selected our ports, increment our counters so that
* we select a new port later */
if (src->next_port_num == transport.port.min &&
src->next_port_num + 1 == transport.port.max) {
src->next_port_num += 2;
}
break;
case GST_RTSP_LOWER_TRANS_UDP:
/* only allow unicast for other streams */
GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
protocols = GST_RTSP_LOWER_TRANS_UDP;
break;
if (!pipelined_request_id) {
/* parse response transport */
res = gst_rtsp_src_setup_stream_from_response (src, stream,
&response, &protocols, retry, &rtpport, &rtcpport);
switch (res) {
case GST_RTSP_ERROR:
goto cleanup_error;
case GST_RTSP_ELAST:
goto retry;
default:
GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
transport.lower_transport);
break;
}
if (!src->interleaved || !retry) {
/* now configure the stream with the selected transport */
if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
GST_DEBUG_OBJECT (src,
"could not configure stream %p transport, skipping stream",
stream);
goto next;
} else if (stream->udpsrc[0] && stream->udpsrc[1]) {
/* retain the first allocated UDP port pair */
g_object_get (G_OBJECT (stream->udpsrc[0]), "port", &rtpport, NULL);
g_object_get (G_OBJECT (stream->udpsrc[1]), "port", &rtcpport, NULL);
}
}
/* we need to activate at least one streams when we detect activity */
} else {
stream->waiting_setup_response = TRUE;
/* we need to activate at least one stream when we detect activity */
src->need_activate = TRUE;
}
/* stream is setup now */
stream->setup = TRUE;
{
GList *skip = walk;
{
GList *skip = walk;
while (TRUE) {
GstRTSPStream *sskip;
while (TRUE) {
GstRTSPStream *sskip;
skip = g_list_next (skip);
if (skip == NULL)
break;
skip = g_list_next (skip);
if (skip == NULL)
break;
sskip = (GstRTSPStream *) skip->data;
sskip = (GstRTSPStream *) skip->data;
/* skip all streams with the same control url */
if (g_str_equal (stream->conninfo.location, sskip->conninfo.location)) {
GST_DEBUG_OBJECT (src, "found stream %p with same control %s",
sskip, sskip->conninfo.location);
sskip->skipped = TRUE;
}
/* skip all streams with the same control url */
if (g_str_equal (stream->conninfo.location, sskip->conninfo.location)) {
GST_DEBUG_OBJECT (src, "found stream %p with same control %s",
sskip, sskip->conninfo.location);
sskip->skipped = TRUE;
}
}
next:
/* clean up our transport struct */
gst_rtsp_transport_init (&transport);
/* clean up used RTSP messages */
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
}
gst_rtsp_message_unset (&request);
}
if (pipelined_request_id) {
gst_rtspsrc_setup_streams_end (src, TRUE);
}
/* store the transport protocol that was configured */
@ -6487,13 +6699,6 @@ send_error:
g_free (str);
goto cleanup_error;
}
no_transport:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Server did not select transport."));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
nothing_to_activate:
{
/* none of the available error codes is really right .. */
@ -6742,7 +6947,7 @@ gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
src->state = GST_RTSP_STATE_INIT;
/* setup streams */
if ((res = gst_rtspsrc_setup_streams (src, async)) < 0)
if ((res = gst_rtspsrc_setup_streams_start (src, async)) < 0)
goto setup_failed;
/* reset our state */
@ -6772,6 +6977,13 @@ gst_rtspsrc_retrieve_sdp (GstRTSPSrc * src, GstSDPMessage ** sdp,
guint8 *data;
guint size;
gchar *respcont = NULL;
GstRTSPVersion versions[] =
{ GST_RTSP_VERSION_2_0, GST_RTSP_VERSION_INVALID };
src->version = src->default_version;
if (src->default_version == GST_RTSP_VERSION_2_0) {
versions[0] = GST_RTSP_VERSION_1_0;
}
restart:
src->need_redirect = FALSE;
@ -6787,7 +6999,7 @@ restart:
goto connect_failed;
/* create OPTIONS */
GST_DEBUG_OBJECT (src, "create options...");
GST_DEBUG_OBJECT (src, "create options... (%s)", async ? "async" : "sync");
res =
gst_rtspsrc_init_request (src, &request, GST_RTSP_OPTIONS,
src->conninfo.url_str);
@ -6795,6 +7007,7 @@ restart:
goto create_request_failed;
/* send OPTIONS */
request.type_data.request.version = src->version;
GST_DEBUG_OBJECT (src, "send options...");
if (async)
@ -6802,8 +7015,13 @@ restart:
if ((res =
gst_rtspsrc_send (src, &src->conninfo, &request, &response,
NULL)) < 0)
NULL, versions)) < 0) {
goto send_error;
}
src->version = request.type_data.request.version;
GST_INFO_OBJECT (src, "Now using version: %s",
gst_rtsp_version_as_text (src->version));
/* parse OPTIONS */
if (!gst_rtspsrc_parse_methods (src, &response))
@ -6829,7 +7047,7 @@ restart:
if ((res =
gst_rtspsrc_send (src, &src->conninfo, &request, &response,
NULL)) < 0)
NULL, NULL)) < 0)
goto send_error;
/* we only perform redirect for describe and play, currently */
@ -7051,7 +7269,8 @@ gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
if (async)
GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
if ((res = gst_rtspsrc_send (src, info, &request, &response, NULL)) < 0)
if ((res =
gst_rtspsrc_send (src, info, &request, &response, NULL, NULL)) < 0)
goto send_error;
/* FIXME, parse result? */
@ -7304,7 +7523,8 @@ done:
}
static GstRTSPResult
gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async,
const gchar * seek_style)
{
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
@ -7364,7 +7584,7 @@ restart:
if (res < 0)
goto create_request_failed;
if (src->need_range) {
if (src->need_range && src->seekable >= 0.0) {
hval = gen_range_header (src, segment);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
@ -7383,10 +7603,16 @@ restart:
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
}
if (seek_style)
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SEEK_STYLE,
seek_style);
if (async)
GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
if ((res = gst_rtspsrc_send (src, conninfo, &request, &response, NULL)) < 0)
if ((res =
gst_rtspsrc_send (src, conninfo, &request, &response, NULL, NULL))
< 0)
goto send_error;
if (src->need_redirect) {
@ -7404,9 +7630,14 @@ restart:
/* seek may have silently failed as it is not supported */
if (!(src->methods & GST_RTSP_PLAY)) {
GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");
if (src->version >= GST_RTSP_VERSION_2_0 && src->seekable >= 0.0) {
GST_WARNING_OBJECT (src, "Server declared stream as seekable but"
" playing with range failed... Ignoring information.");
}
/* obviously it is supported as we made it here */
src->methods |= GST_RTSP_PLAY;
src->seekable = FALSE;
src->seekable = -1.0;
/* but there is nothing to parse in the response,
* so convey we have no idea and not to expect anything particular */
clear_rtp_base (src, stream);
@ -7590,7 +7821,9 @@ gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
setup_url)) < 0)
goto create_request_failed;
if ((res = gst_rtspsrc_send (src, conninfo, &request, &response, NULL)) < 0)
if ((res =
gst_rtspsrc_send (src, conninfo, &request, &response, NULL,
NULL)) < 0)
goto send_error;
gst_rtsp_message_unset (&request);
@ -7760,7 +7993,7 @@ gst_rtspsrc_thread (GstRTSPSrc * src)
gst_rtspsrc_open (src, TRUE);
break;
case CMD_PLAY:
gst_rtspsrc_play (src, &src->segment, TRUE);
gst_rtspsrc_play (src, &src->segment, TRUE, NULL);
break;
case CMD_PAUSE:
gst_rtspsrc_pause (src, TRUE);

7
gst/rtsp/gstrtspsrc.h
View file

@ -107,6 +107,7 @@ struct _GstRTSPStream {
gboolean eos;
gboolean discont;
gboolean need_caps;
gboolean waiting_setup_response;
/* for interleaved mode */
guint8 channel[2];
@ -269,7 +270,8 @@ struct _GstRTSPSrc {
/* supported methods */
gint methods;
gboolean seekable;
/* Same semantic as described in gst_rtsp_media_seekable */
gfloat seekable;
GstClockTime last_pos;
/* session management */
@ -282,6 +284,9 @@ struct _GstRTSPSrc {
/* a list of RTSP extensions as GstElement */
GstRTSPExtensionList *extensions;
GstRTSPVersion default_version;
GstRTSPVersion version;
};
struct _GstRTSPSrcClass {