mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-28 06:42:16 +00:00
rtsp: Start implementing support for RTSP 2.0
- 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
parent
5faad79049
commit
8121752887
2 changed files with 416 additions and 178 deletions
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@ -233,6 +233,7 @@ gst_rtsp_src_ntp_time_source_get_type (void)
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#define DEFAULT_RFC7273_SYNC FALSE
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#define DEFAULT_MAX_TS_OFFSET_ADJUSTMENT 0
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#define DEFAULT_MAX_TS_OFFSET 3000000000
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#define DEFAULT_VERSION GST_RTSP_VERSION_1_0
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enum
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{
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@ -274,7 +275,8 @@ enum
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PROP_MAX_RTCP_RTP_TIME_DIFF,
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PROP_RFC7273_SYNC,
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PROP_MAX_TS_OFFSET_ADJUSTMENT,
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PROP_MAX_TS_OFFSET
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PROP_MAX_TS_OFFSET,
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PROP_DEFAULT_VERSION,
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};
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#define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
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@ -333,7 +335,7 @@ static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
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static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
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static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
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gboolean async);
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gboolean async, const gchar * seek_style);
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static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async);
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static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
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gboolean only_close);
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@ -755,6 +757,13 @@ gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
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"(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_DEFAULT_VERSION,
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g_param_spec_enum ("default-rtsp-version",
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"The RTSP version to try first",
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"The RTSP version that should be tried first when negotiating version.",
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GST_TYPE_RTSP_VERSION, DEFAULT_VERSION,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTSPSrc:max-ts-offset-adjustment:
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*
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@ -936,6 +945,8 @@ gst_rtspsrc_init (GstRTSPSrc * src)
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src->max_ts_offset_adjustment = DEFAULT_MAX_TS_OFFSET_ADJUSTMENT;
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src->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
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src->max_ts_offset_is_set = FALSE;
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src->default_version = DEFAULT_VERSION;
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src->version = GST_RTSP_VERSION_INVALID;
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/* get a list of all extensions */
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src->extensions = gst_rtsp_ext_list_get ();
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@ -1242,6 +1253,9 @@ gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
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rtspsrc->max_ts_offset = g_value_get_int64 (value);
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rtspsrc->max_ts_offset_is_set = TRUE;
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break;
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case PROP_DEFAULT_VERSION:
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rtspsrc->default_version = g_value_get_enum (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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@ -1397,6 +1411,9 @@ gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
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case PROP_MAX_TS_OFFSET:
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g_value_set_int64 (value, rtspsrc->max_ts_offset);
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break;
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case PROP_DEFAULT_VERSION:
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g_value_set_enum (value, rtspsrc->default_version);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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@ -2227,6 +2244,7 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
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gboolean playing;
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GstSegment seeksegment = { 0, };
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GList *walk;
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gchar *seek_style = NULL;
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if (event) {
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GST_DEBUG_OBJECT (src, "doing seek with event");
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@ -2341,8 +2359,21 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
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&& GST_STATE (src) == GST_STATE_PLAYING)
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|| (GST_STATE_PENDING (src) == GST_STATE_PLAYING);
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GST_OBJECT_UNLOCK (src);
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if (src->version >= GST_RTSP_VERSION_2_0) {
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if (flags & GST_SEEK_FLAG_ACCURATE)
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seek_style = "RAP";
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else if (flags & GST_SEEK_FLAG_KEY_UNIT)
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seek_style = "CoRAP";
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else if (flags & GST_SEEK_FLAG_KEY_UNIT
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&& flags & GST_SEEK_FLAG_SNAP_BEFORE)
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seek_style = "First-Prior";
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else if (flags & GST_SEEK_FLAG_KEY_UNIT && flags & GST_SEEK_FLAG_SNAP_AFTER)
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seek_style = "Next";
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}
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if (playing)
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gst_rtspsrc_play (src, &seeksegment, FALSE);
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gst_rtspsrc_play (src, &seeksegment, FALSE, seek_style);
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GST_RTSP_STREAM_UNLOCK (src);
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@ -2540,13 +2571,24 @@ gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
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if (format == GST_FORMAT_TIME) {
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gboolean seekable =
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src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
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GstClockTime start, duration = src->segment.duration;
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/* seeking without duration is unlikely */
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seekable = seekable && src->seekable && src->segment.duration &&
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seekable = seekable && src->seekable >= 0.0 && src->segment.duration &&
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GST_CLOCK_TIME_IS_VALID (src->segment.duration);
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gst_query_set_seeking (query, GST_FORMAT_TIME, seekable, 0,
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src->segment.duration);
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if (seekable) {
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if (src->seekable > 0.0) {
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start = src->last_pos - src->seekable * GST_SECOND;
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} else {
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/* src->seekable == 0 means that we can only seek to 0 */
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start = 0;
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duration = 0;
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}
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}
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gst_query_set_seeking (query, GST_FORMAT_TIME, seekable, start,
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duration);
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res = TRUE;
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}
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break;
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@ -4500,8 +4542,7 @@ gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
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if (res < 0)
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goto send_error;
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if (src->debug)
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gst_rtsp_message_dump (&request);
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request.type_data.request.version = src->version;
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res = gst_rtspsrc_connection_send (src, &src->conninfo, &request, NULL);
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if (res < 0)
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@ -5074,7 +5115,7 @@ gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
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goto open_failed;
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/* start playback */
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if (gst_rtspsrc_play (src, &src->segment, async) < 0)
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if (gst_rtspsrc_play (src, &src->segment, async, NULL) < 0)
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goto play_failed;
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done:
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@ -5481,33 +5522,14 @@ no_user_pass:
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}
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static GstRTSPResult
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gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
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GstRTSPMessage * request, GstRTSPMessage * response,
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GstRTSPStatusCode * code)
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gst_rtsp_src_receive_response (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
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GstRTSPMessage * response, GstRTSPStatusCode * code)
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{
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GstRTSPResult res;
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GstRTSPStatusCode thecode;
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gchar *content_base = NULL;
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gint try = 0;
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GstRTSPResult res = gst_rtspsrc_connection_receive (src, conninfo,
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response, src->ptcp_timeout);
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again:
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if (!src->short_header)
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gst_rtsp_ext_list_before_send (src->extensions, request);
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GST_DEBUG_OBJECT (src, "sending message");
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DEBUG_RTSP (src, request);
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res = gst_rtspsrc_connection_send (src, conninfo, request, src->ptcp_timeout);
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if (res < 0)
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goto send_error;
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gst_rtsp_connection_reset_timeout (conninfo->connection);
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next:
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res =
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gst_rtspsrc_connection_receive (src, conninfo, response,
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src->ptcp_timeout);
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if (res < 0)
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goto receive_error;
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@ -5520,7 +5542,9 @@ next:
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goto server_eof;
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else if (res < 0)
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goto handle_request_failed;
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goto next;
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/* Not a response, receive next message */
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return gst_rtsp_src_receive_response (src, conninfo, response, code);
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case GST_RTSP_MESSAGE_RESPONSE:
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/* ok, a response is good */
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GST_DEBUG_OBJECT (src, "received response message");
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@ -5529,11 +5553,15 @@ next:
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/* get next response */
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GST_DEBUG_OBJECT (src, "handle data response message");
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gst_rtspsrc_handle_data (src, response);
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goto next;
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/* Not a response, receive next message */
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return gst_rtsp_src_receive_response (src, conninfo, response, code);
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default:
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GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
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response->type);
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goto next;
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/* Not a response, receive next message */
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return gst_rtsp_src_receive_response (src, conninfo, response, code);
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}
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thecode = response->type_data.response.code;
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@ -5555,38 +5583,15 @@ next:
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g_free (src->content_base);
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src->content_base = g_strdup (content_base);
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}
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gst_rtsp_ext_list_after_send (src->extensions, request, response);
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return GST_RTSP_OK;
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/* ERRORS */
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send_error:
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{
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gchar *str = gst_rtsp_strresult (res);
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if (res != GST_RTSP_EINTR) {
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GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
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("Could not send message. (%s)", str));
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} else {
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GST_WARNING_OBJECT (src, "send interrupted");
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}
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g_free (str);
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return res;
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}
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receive_error:
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{
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switch (res) {
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case GST_RTSP_EEOF:
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GST_WARNING_OBJECT (src, "server closed connection");
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if ((try == 0) && !src->interleaved && src->udp_reconnect) {
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try++;
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/* if reconnect succeeds, try again */
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if ((res =
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gst_rtsp_conninfo_reconnect (src, &src->conninfo,
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FALSE)) == 0)
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goto again;
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}
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/* only try once after reconnect, then fallthrough and error out */
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return GST_RTSP_EEOF;
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default:
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{
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gchar *str = gst_rtsp_strresult (res);
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@ -5619,13 +5624,70 @@ server_eof:
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}
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}
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static GstRTSPResult
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gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
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GstRTSPMessage * request, GstRTSPMessage * response,
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GstRTSPStatusCode * code)
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{
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GstRTSPResult res;
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gint try = 0;
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again:
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if (!src->short_header)
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gst_rtsp_ext_list_before_send (src->extensions, request);
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GST_DEBUG_OBJECT (src, "sending message");
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DEBUG_RTSP (src, request);
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res = gst_rtspsrc_connection_send (src, conninfo, request, src->ptcp_timeout);
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if (res < 0)
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goto send_error;
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gst_rtsp_connection_reset_timeout (conninfo->connection);
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if (!response)
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return res;
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res = gst_rtsp_src_receive_response (src, conninfo, response, code);
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if (res == GST_RTSP_EEOF) {
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GST_WARNING_OBJECT (src, "server closed connection");
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/* only try once after reconnect, then fallthrough and error out */
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if ((try == 0) && !src->interleaved && src->udp_reconnect) {
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try++;
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/* if reconnect succeeds, try again */
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if ((res = gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) == 0)
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goto again;
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}
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}
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gst_rtsp_ext_list_after_send (src->extensions, request, response);
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return res;
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send_error:
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{
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gchar *str = gst_rtsp_strresult (res);
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if (res != GST_RTSP_EINTR) {
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GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
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("Could not send message. (%s)", str));
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} else {
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GST_WARNING_OBJECT (src, "send interrupted");
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}
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g_free (str);
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return res;
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}
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}
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/**
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* gst_rtspsrc_send:
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* @src: the rtsp source
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* @conn: the connection to send on
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* @conninfo: the connection information to send on
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* @request: must point to a valid request
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* @response: must point to an empty #GstRTSPMessage
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* @code: an optional code result
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* @versions: List of versions to try, setting it back onto the @request message
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* if not set, `src->version` will be used as RTSP version.
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*
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* send @request and retrieve the response in @response. optionally @code can be
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* non-NULL in which case it will contain the status code of the response.
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@ -5645,13 +5707,14 @@ server_eof:
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static GstRTSPResult
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gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
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GstRTSPMessage * request, GstRTSPMessage * response,
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GstRTSPStatusCode * code)
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GstRTSPStatusCode * code, GstRTSPVersion * versions)
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{
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GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
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GstRTSPResult res = GST_RTSP_ERROR;
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gint count;
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gboolean retry;
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GstRTSPMethod method = GST_RTSP_INVALID;
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gint version_retry = 0;
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count = 0;
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do {
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|
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@ -5664,6 +5727,9 @@ gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
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/* save method so we can disable it when the server complains */
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method = request->type_data.request.method;
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|
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if (!versions)
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request->type_data.request.version = src->version;
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|
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if ((res =
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gst_rtspsrc_try_send (src, conninfo, request, response,
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&int_code)) < 0)
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|
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@ -5678,6 +5744,19 @@ gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
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retry = TRUE;
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}
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break;
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case GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED:
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GST_INFO_OBJECT (src, "Version %s not supported by the server",
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gst_rtsp_version_as_text (versions[version_retry]));
|
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if (versions && versions[version_retry] != GST_RTSP_VERSION_INVALID) {
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GST_INFO_OBJECT (src, "Unsupported version %s => trying %s",
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gst_rtsp_version_as_text (request->type_data.request.version),
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gst_rtsp_version_as_text (versions[version_retry]));
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request->type_data.request.version = versions[version_retry];
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retry = TRUE;
|
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version_retry++;
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break;
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}
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/* falltrough */
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default:
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break;
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}
|
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|
|
@ -5769,7 +5848,7 @@ static GstRTSPResult
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gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
|
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GstRTSPMessage * response, GstRTSPSrc * src)
|
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{
|
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return gst_rtspsrc_send (src, &src->conninfo, request, response, NULL);
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return gst_rtspsrc_send (src, &src->conninfo, request, response, NULL, NULL);
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}
|
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|
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|
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|
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@ -5792,11 +5871,19 @@ gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
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while (TRUE) {
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respoptions = NULL;
|
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gst_rtsp_message_get_header (response, field, &respoptions, indx);
|
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if (indx == 0 && !respoptions) {
|
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/* if no Allow header was found then try the Public header... */
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field = GST_RTSP_HDR_PUBLIC;
|
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gst_rtsp_message_get_header (response, field, &respoptions, indx);
|
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if (!respoptions)
|
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break;
|
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src->methods |= gst_rtsp_options_from_text (respoptions);
|
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|
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indx++;
|
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}
|
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|
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indx = 0;
|
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field = GST_RTSP_HDR_PUBLIC;
|
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while (TRUE) {
|
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respoptions = NULL;
|
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gst_rtsp_message_get_header (response, field, &respoptions, indx);
|
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if (!respoptions)
|
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break;
|
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|
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|
|
@ -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;
|
||||
|
||||
|
|
@ -6322,79 +6590,25 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
|
|||
goto response_error;
|
||||
}
|
||||
|
||||
|
||||
if (!pipelined_request_id) {
|
||||
/* 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;
|
||||
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;
|
||||
|
||||
|
|
@ -6415,13 +6629,11 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
|
|||
}
|
||||
}
|
||||
}
|
||||
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);
|
||||
}
|
||||
|
||||
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);
|
||||
|
|
|
|||
|
|
@ -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 {
|
||||
|
|
|
|||
Loading…
Reference in a new issue