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gst/rtsp/gstrtspsrc.*: Reorganize stream parsing and creation.

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Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (find_stream_by_pt),
(gst_rtspsrc_create_stream), (gst_rtspsrc_free_stream),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
(gst_rtspsrc_push_event), (gst_rtspsrc_loop), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
Reorganize stream parsing and creation.
Detect container formats in interleaved mode.
Keep more state about the streams.
Assume a server also supports PLAY if it does not say.
Add unicast and interleaved properties to TCP transport requests to make
some servers happy (WMServer).
* gst/rtsp/sdpmessage.h:
Add some defines for the standard Bandwidth types.
This commit is contained in:
Wim Taymans 2006-09-19 17:25:15 +00:00
parent 34fd3a416d
commit a7d7309e18
4 changed files with 193 additions and 85 deletions
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20
ChangeLog
View file

@ -1,3 +1,23 @@
2006-09-19 Wim Taymans <wim@fluendo.com>
* gst/rtsp/gstrtspsrc.c: (find_stream_by_pt),
(gst_rtspsrc_create_stream), (gst_rtspsrc_free_stream),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
(gst_rtspsrc_push_event), (gst_rtspsrc_loop), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
Reorganize stream parsing and creation.
Detect container formats in interleaved mode.
Keep more state about the streams.
Assume a server also supports PLAY if it does not say.
Add unicast and interleaved properties to TCP transport requests to make
some servers happy (WMServer).
* gst/rtsp/sdpmessage.h:
Add some defines for the standard Bandwidth types.
2006-09-19 Wim Taymans <wim@fluendo.com>
* gst/rtsp/test.c: (main):

223
gst/rtsp/gstrtspsrc.c
View file

@ -132,6 +132,7 @@ static void gst_rtspsrc_finalize (GObject * object);
static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
gpointer iface_data);
static GstCaps *gst_rtspsrc_media_to_caps (gint pt, SDPMedia * media);
static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
GstStateChange transition);
@ -294,18 +295,74 @@ gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
}
}
static GstRTSPStream *
gst_rtspsrc_create_stream (GstRTSPSrc * src)
static gint
find_stream_by_pt (GstRTSPStream * stream, gconstpointer a)
{
GstRTSPStream *s;
gint pt = GPOINTER_TO_INT (a);
s = g_new0 (GstRTSPStream, 1);
s->parent = src;
s->id = src->numstreams++;
if (stream->pt == pt)
return 0;
src->streams = g_list_append (src->streams, s);
return -1;
}
return s;
static GstRTSPStream *
gst_rtspsrc_create_stream (GstRTSPSrc * src, SDPMedia * media)
{
GstRTSPStream *stream;
gchar *control_url;
gchar *payload;
stream = g_new0 (GstRTSPStream, 1);
stream->parent = src;
/* we mark the pad as not linked, we will mark it as OK when we add the pad to
* the element. */
stream->last_ret = GST_FLOW_NOT_LINKED;
stream->id = src->numstreams++;
/* we must have a payload. No payload means we cannot create caps */
/* FIXME, handle multiple formats. */
if ((payload = sdp_media_get_format (media, 0))) {
stream->pt = atoi (payload);
/* convert caps */
stream->caps = gst_rtspsrc_media_to_caps (stream->pt, media);
if (stream->pt >= 96) {
/* If we have a dynamic payload type, see if we have a stream with the
* same payload number. If there is one, they are part of the same
* container and we only need to add one pad. */
if (g_list_find_custom (src->streams, GINT_TO_POINTER (stream->pt),
(GCompareFunc) find_stream_by_pt)) {
stream->container = TRUE;
}
}
}
/* get control url to construct the setup url. The setup url is used to
* configure the transport of the stream and is used to identity the stream in
* the RTP-Info header field returned from PLAY. */
control_url = sdp_media_get_attribute_val (media, "control");
GST_DEBUG_OBJECT (src, "stream %d", stream->id);
GST_DEBUG_OBJECT (src, " pt: %d", stream->pt);
GST_DEBUG_OBJECT (src, " container: %d", stream->container);
GST_DEBUG_OBJECT (src, " caps: %" GST_PTR_FORMAT, stream->caps);
GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
if (control_url != NULL) {
/* FIXME, what if the control_url starts with a '/' or a non rtsp: protocol? */
/* check absolute/relative URL */
if (g_str_has_prefix (control_url, "rtsp://"))
stream->setup_url = g_strdup (control_url);
else
stream->setup_url = g_strdup_printf ("%s/%s", src->location, control_url);
}
GST_DEBUG_OBJECT (src, " setup: %s", GST_STR_NULL (stream->setup_url));
/* we keep track of all streams */
src->streams = g_list_append (src->streams, stream);
return stream;
}
#if 0
@ -314,8 +371,8 @@ gst_rtspsrc_free_stream (GstRTSPSrc * src, GstRTSPStream * stream)
{
if (stream->caps) {
gst_caps_unref (stream->caps);
stream->caps = NULL;
}
g_free (stream->setup_url);
src->streams = g_list_remove (src->streams, stream);
src->numstreams--;
@ -404,24 +461,16 @@ gst_rtspsrc_parse_rtpmap (gchar * rtpmap, gint * payload, gchar ** name,
* a=fmtp:<payload> <param>[=<value>];...
*/
static GstCaps *
gst_rtspsrc_media_to_caps (SDPMedia * media)
gst_rtspsrc_media_to_caps (gint pt, SDPMedia * media)
{
GstCaps *caps;
gchar *payload;
gchar *rtpmap;
gchar *fmtp;
gint pt;
gchar *name = NULL;
gint rate = -1;
gchar *params = NULL;
GstStructure *s;
payload = sdp_media_get_format (media, 0);
if (payload == NULL)
goto no_payload;
pt = atoi (payload);
/* dynamic payloads need rtpmap */
if (pt >= 96) {
gint payload = 0;
@ -499,11 +548,6 @@ gst_rtspsrc_media_to_caps (SDPMedia * media)
return caps;
/* ERRORS */
no_payload:
{
g_warning ("payload type not given");
return NULL;
}
no_rtpmap:
{
g_warning ("rtpmap type not given for dynamic payload %d", pt);
@ -512,7 +556,7 @@ no_rtpmap:
}
static gboolean
gst_rtspsrc_stream_setup_rtp (GstRTSPStream * stream, SDPMedia * media,
gst_rtspsrc_stream_setup_rtp (GstRTSPStream * stream,
gint * rtpport, gint * rtcpport)
{
GstStateChangeReturn ret;
@ -653,10 +697,10 @@ cleanup:
static gboolean
gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
SDPMedia * media, RTSPTransport * transport)
RTSPTransport * transport)
{
GstRTSPSrc *src;
GstPad *pad, *gpad;
GstPad *pad;
GstPadTemplate *template;
GstStateChangeReturn ret;
gchar *name;
@ -665,6 +709,7 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
GST_DEBUG ("configuring RTP transport for stream %p", stream);
/* FIXME, the session manager needs to be shared with all the streams */
if (!(stream->rtpdec = gst_element_factory_make ("rtpdec", NULL)))
goto no_element;
@ -686,12 +731,8 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
stream->rtcpchannel = transport->interleaved.max;
GST_DEBUG ("stream %p on channels %d-%d", stream,
stream->rtpchannel, stream->rtcpchannel);
/* also store the caps in the stream, we need this when setting caps on
* outgoing buffers */
stream->caps = gst_rtspsrc_media_to_caps (media);
} else {
/* multicast was selected, create UDP sources and connect to the multicast
/* multicast was selected, create UDP sources and join the multicast
* group. */
if (transport->multicast) {
gchar *uri;
@ -714,7 +755,6 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
if (stream->rtcpsrc == NULL)
goto no_element;
/* change state */
gst_element_set_state (stream->rtpsrc, GST_STATE_PAUSED);
gst_element_set_state (stream->rtcpsrc, GST_STATE_PAUSED);
@ -724,8 +764,7 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
gst_bin_add (GST_BIN_CAST (src), stream->rtcpsrc);
}
/* configure caps on the RTP source element */
stream->caps = gst_rtspsrc_media_to_caps (media);
/* set caps */
g_object_set (G_OBJECT (stream->rtpsrc), "caps", stream->caps, NULL);
/* configure for UDP delivery, we need to connect the UDP pads to
@ -748,13 +787,16 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
/* create ghostpad */
name = g_strdup_printf ("rtp_stream%d", stream->id);
template = gst_static_pad_template_get (&rtptemplate);
gpad = gst_ghost_pad_new_from_template (name, pad, template);
stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
gst_object_unref (template);
g_free (name);
gst_object_unref (pad);
gst_element_add_pad (GST_ELEMENT_CAST (src), gpad);
/* mark pad as ok */
stream->last_ret = GST_FLOW_OK;
/* and add */
gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
return TRUE;
@ -772,7 +814,7 @@ start_rtpdec_failure:
}
static gint
find_stream (GstRTSPStream * stream, gconstpointer a)
find_stream_by_channel (GstRTSPStream * stream, gconstpointer a)
{
gint channel = GPOINTER_TO_INT (a);
@ -824,6 +866,10 @@ gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
for (streams = src->streams; streams; streams = g_list_next (streams)) {
GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
/* only pads that have a connection to the outside world */
if (ostream->srcpad == NULL)
continue;
gst_event_ref (event);
gst_pad_push_event (ostream->rtpdecrtp, event);
gst_event_ref (event);
@ -857,7 +903,7 @@ gst_rtspsrc_loop (GstRTSPSrc * src)
channel = response.type_data.data.channel;
lstream = g_list_find_custom (src->streams, GINT_TO_POINTER (channel),
(GCompareFunc) find_stream);
(GCompareFunc) find_stream_by_channel);
if (!lstream)
goto unknown_stream;
@ -1062,9 +1108,9 @@ gst_rtspsrc_parse_methods (GstRTSPSrc * src, RTSPMessage * response)
rtsp_message_get_header (response, RTSP_HDR_PUBLIC, &respoptions);
if (!respoptions) {
/* this field is not required, assume the server supports
* DESCRIBE and SETUP*/
* DESCRIBE, SETUP and PLAY */
GST_DEBUG_OBJECT (src, "could not get OPTIONS");
src->methods = RTSP_DESCRIBE | RTSP_SETUP;
src->methods = RTSP_DESCRIBE | RTSP_SETUP | RTSP_PLAY;
goto done;
}
@ -1075,8 +1121,7 @@ gst_rtspsrc_parse_methods (GstRTSPSrc * src, RTSPMessage * response)
*/
options = g_strsplit (respoptions, ",", 0);
i = 0;
while (options[i]) {
for (i = 0; options[i]; i++) {
gchar *stripped;
gint method;
@ -1086,7 +1131,6 @@ gst_rtspsrc_parse_methods (GstRTSPSrc * src, RTSPMessage * response)
/* keep bitfield of supported methods */
if (method != -1)
src->methods |= method;
i++;
}
g_strfreev (options);
@ -1196,37 +1240,25 @@ gst_rtspsrc_open (GstRTSPSrc * src)
n_streams = sdp_message_medias_len (&sdp);
for (i = 0; i < n_streams; i++) {
SDPMedia *media;
gchar *setup_url;
gchar *control_url;
gchar *transports;
media = sdp_message_get_media (&sdp, i);
GST_DEBUG_OBJECT (src, "setup media %d", i);
control_url = sdp_media_get_attribute_val (media, "control");
if (control_url == NULL) {
GST_DEBUG_OBJECT (src, "no control url found, skipping stream %d", i);
/* create stream from the media */
stream = gst_rtspsrc_create_stream (src, media);
/* skip setup if we have no URL for it */
if (stream->setup_url == NULL)
continue;
}
/* check absolute/relative URL */
/* FIXME, what if the control_url starts with a '/' or a non rtsp: protocol? */
if (g_str_has_prefix (control_url, "rtsp://")) {
setup_url = g_strdup (control_url);
} else {
setup_url = g_strdup_printf ("%s/%s", src->location, control_url);
}
GST_DEBUG_OBJECT (src, "setup %s", setup_url);
GST_DEBUG_OBJECT (src, "doing setup of stream %d with %s", i,
stream->setup_url);
/* create SETUP request */
res = rtsp_message_init_request (&request, RTSP_SETUP, setup_url);
g_free (setup_url);
res = rtsp_message_init_request (&request, RTSP_SETUP, stream->setup_url);
if (res < 0)
goto create_request_failed;
stream = gst_rtspsrc_create_stream (src);
transports = g_strdup ("");
if (protocols & GST_RTSP_PROTO_UDP_UNICAST) {
gchar *new;
@ -1234,7 +1266,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
gchar *trxparams;
/* allocate two UDP ports */
if (!gst_rtspsrc_stream_setup_rtp (stream, media, &rtpport, &rtcpport))
if (!gst_rtspsrc_stream_setup_rtp (stream, &rtpport, &rtcpport))
goto setup_rtp_failed;
GST_DEBUG_OBJECT (src, "setting up RTP ports %d-%d", rtpport, rtcpport);
@ -1260,13 +1292,18 @@ gst_rtspsrc_open (GstRTSPSrc * src)
transports = new;
}
if (protocols & GST_RTSP_PROTO_TCP) {
gchar *new;
gchar *new, *interleaved;
gint channel;
GST_DEBUG_OBJECT (src, "setting up TCP");
/* the channels for this stream is by default the next available number */
channel = i * 2;
interleaved = g_strdup_printf ("interleaved=%d-%d", channel, channel + 1);
new =
g_strconcat (transports, transports[0] ? "," : "", "RTP/AVP/TCP",
NULL);
g_strconcat (transports, transports[0] ? "," : "",
"RTP/AVP/TCP;unicast;", interleaved, NULL);
g_free (interleaved);
g_free (transports);
transports = new;
}
@ -1308,11 +1345,15 @@ gst_rtspsrc_open (GstRTSPSrc * src)
protocols = GST_RTSP_PROTO_UDP_UNICAST;
}
}
/* now configure the stream with the transport */
if (!gst_rtspsrc_stream_configure_transport (stream, media, &transport)) {
GST_DEBUG_OBJECT (src,
"could not configure stream %d transport, skipping stream", i);
if (!stream->container || !src->interleaved) {
/* now configure the stream with the transport */
if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
GST_DEBUG_OBJECT (src,
"could not configure stream %d transport, skipping stream", i);
}
}
/* clean up our transport struct */
rtsp_transport_init (&transport);
}
@ -1450,12 +1491,37 @@ close_failed:
}
}
/* RTP-Info is of the format:
*
* url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
*/
static gboolean
gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
{
gchar **infos;
gint i;
infos = g_strsplit (rtpinfo, ",", 0);
for (i = 0; infos[i]; i++) {
/* FIXME, do something here:
* parse url, find stream for url.
* parse seq and rtptime. The seq number should be configured in the rtp
* depayloader or session manager to detect gaps. Same for the rtptime, it
* should be used to create an initial time newsegment.
*/
}
g_strfreev (infos);
return TRUE;
}
static gboolean
gst_rtspsrc_play (GstRTSPSrc * src)
{
RTSPMessage request = { 0 };
RTSPMessage response = { 0 };
RTSPResult res;
gchar *rtpinfo;
if (!(src->methods & RTSP_PLAY))
return TRUE;
@ -1473,14 +1539,23 @@ gst_rtspsrc_play (GstRTSPSrc * src)
goto send_error;
rtsp_message_unset (&request);
/* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
* for the RTP packets. If this is not present, we assume all starts from 0...
* FIXME, this is info for the RTP session manager ideally. */
rtsp_message_get_header (&response, RTSP_HDR_RTP_INFO, &rtpinfo);
if (rtpinfo)
gst_rtspsrc_parse_rtpinfo (src, rtpinfo);
rtsp_message_unset (&response);
/* for interleaved transport, we receive the data on the RTSP connection
* instead of UDP. We start a task to select and read from that connection. */
if (src->interleaved) {
src->task = gst_task_create ((GstTaskFunction) gst_rtspsrc_loop, src);
gst_task_set_lock (src->task, src->stream_rec_lock);
if (src->task == NULL) {
src->task = gst_task_create ((GstTaskFunction) gst_rtspsrc_loop, src);
gst_task_set_lock (src->task, src->stream_rec_lock);
}
src->running = TRUE;
gst_task_start (src->task);
}

31
gst/rtsp/gstrtspsrc.h
View file

@ -60,28 +60,37 @@ typedef enum
typedef struct _GstRTSPStream GstRTSPStream;
struct _GstRTSPStream {
gint id;
gint id;
GstRTSPSrc *parent;
GstRTSPSrc *parent;
/* pad we expose or NULL when it does not have an actual pad */
GstPad *srcpad;
GstFlowReturn last_ret;
/* for interleaved mode */
gint rtpchannel;
gint rtcpchannel;
GstCaps *caps;
gint rtpchannel;
gint rtcpchannel;
GstCaps *caps;
/* our udp sources for RTP */
GstElement *rtpsrc;
GstElement *rtcpsrc;
GstElement *rtpsrc;
GstElement *rtcpsrc;
/* our udp sink back to the server */
GstElement *rtcpsink;
GstElement *rtcpsink;
/* the RTP decoder */
GstElement *rtpdec;
GstPad *rtpdecrtp;
GstPad *rtpdecrtcp;
GstElement *rtpdec;
GstPad *rtpdecrtp;
GstPad *rtpdecrtcp;
/* state */
gint pt;
gboolean container;
gchar *setup_url;
guint32 ssrc;
guint32 seqbase;
};
struct _GstRTSPSrc {

4
gst/rtsp/sdpmessage.h
View file

@ -43,6 +43,10 @@ typedef struct {
gint addr_number;
} SDPConnection;
#define SDP_BWTYPE_CT "CT" /* conference total */
#define SDP_BWTYPE_AS "AS" /* application specific */
#define SDP_BWTYPE_EXT_PREFIX "X-" /* extension prefix */
typedef struct {
gchar *bwtype;
gint bandwidth;