*/
/**
* SECTION:element-rtspsrc
+ * @title: rtspsrc
*
* Makes a connection to an RTSP server and read the data.
* rtspsrc strictly follows RFC 2326 and therefore does not (yet) support
* rtspsrc acts like a live source and will therefore only generate data in the
* PLAYING state.
*
- * <refsect2>
- * <title>Example launch line</title>
+ * If a RTP session times out then the rtspsrc will generate an element message
+ * named "GstRTSPSrcTimeout". Currently this is only supported for timeouts
+ * triggered by RTCP.
+ *
+ * The message's structure contains three fields:
+ *
+ * #GstRTSPSrcTimeoutCause `cause`: the cause of the timeout.
+ *
+ * #gint `stream-number`: an internal identifier of the stream that timed out.
+ *
+ * #guint `ssrc`: the SSRC of the stream that timed out.
+ *
+ * ## Example launch line
* |[
* gst-launch-1.0 rtspsrc location=rtsp://some.server/url ! fakesink
* ]| Establish a connection to an RTSP server and send the raw RTP packets to a
* fakesink.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
#define DEFAULT_BACKCHANNEL GST_RTSP_BACKCHANNEL_NONE
#define DEFAULT_TEARDOWN_TIMEOUT (100 * GST_MSECOND)
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+#define DEFAULT_START_POSITION 0
+#endif
+
enum
{
PROP_0,
PROP_DEBUG,
PROP_RETRY,
PROP_TIMEOUT,
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ PROP_START_POSITION,
+ PROP_RESUME_POSITION,
+#endif
PROP_TCP_TIMEOUT,
PROP_LATENCY,
PROP_DROP_ON_LATENCY,
}
#endif
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+static void
+gst_rtspsrc_post_error_message (GstRTSPSrc * src, GstRTSPSrcError error_id,
+ const gchar * error_string)
+{
+ GstMessage *message;
+ GstStructure *structure;
+ gboolean ret = TRUE;
+
+ GST_ERROR_OBJECT (src, "[%d] %s", error_id, error_string);
+
+ structure = gst_structure_new ("streaming_error",
+ "error_id", G_TYPE_UINT, error_id,
+ "error_string", G_TYPE_STRING, error_string, NULL);
+
+ message =
+ gst_message_new_custom (GST_MESSAGE_ERROR, GST_OBJECT (src), structure);
+
+ ret = gst_element_post_message (GST_ELEMENT (src), message);
+ if (!ret)
+ GST_ERROR_OBJECT (src, "fail to post error message.");
+
+ return;
+}
+#endif
+
static gboolean
default_select_stream (GstRTSPSrc * src, guint id, GstCaps * caps)
{
"Retry TCP transport after UDP timeout microseconds (0 = disabled)",
0, G_MAXUINT64, DEFAULT_TIMEOUT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ g_object_class_install_property (gobject_class, PROP_START_POSITION,
+ g_param_spec_uint64 ("pending-start-position", "set start position",
+ "Set start position before PLAYING request.",
+ 0, G_MAXUINT64, DEFAULT_START_POSITION,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_RESUME_POSITION,
+ g_param_spec_uint64 ("resume-position", "set resume position",
+ "Set resume position before PLAYING request after pause.",
+ 0, G_MAXUINT64, DEFAULT_START_POSITION,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+#endif
g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
"Fail after timeout microseconds on TCP connections (0 = disabled)",
/**
* GstRTSPSrc:port-range:
*
- * Configure the client port numbers that can be used to recieve RTP and
+ * Configure the client port numbers that can be used to receive RTP and
* RTCP.
*/
g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
src->debug = DEFAULT_DEBUG;
src->retry = DEFAULT_RETRY;
src->udp_timeout = DEFAULT_TIMEOUT;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ src->start_position = DEFAULT_START_POSITION;
+ src->is_audio_codec_supported = FALSE;
+ src->is_video_codec_supported = FALSE;
+ src->audio_codec = NULL;
+ src->video_codec = NULL;
+ src->video_frame_size = NULL;
+#endif
gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
src->latency = DEFAULT_LATENCY_MS;
src->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
src->version = GST_RTSP_VERSION_INVALID;
src->teardown_timeout = DEFAULT_TEARDOWN_TIMEOUT;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ g_mutex_init (&(src)->pause_lock);
+ g_cond_init (&(src)->open_end);
+#endif
/* get a list of all extensions */
src->extensions = gst_rtsp_ext_list_get ();
rtspsrc = GST_RTSPSRC (object);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ rtspsrc->is_audio_codec_supported = FALSE;
+ rtspsrc->is_video_codec_supported = FALSE;
+ if (rtspsrc->audio_codec) {
+ g_free (rtspsrc->audio_codec);
+ rtspsrc->audio_codec = NULL;
+ }
+ if (rtspsrc->video_codec) {
+ g_free (rtspsrc->video_codec);
+ rtspsrc->video_codec = NULL;
+ }
+ if (rtspsrc->video_frame_size) {
+ g_free (rtspsrc->video_frame_size);
+ rtspsrc->video_frame_size = NULL;
+ }
+#endif
gst_rtsp_ext_list_free (rtspsrc->extensions);
g_free (rtspsrc->conninfo.location);
gst_rtsp_url_free (rtspsrc->conninfo.url);
g_free (rtspsrc->multi_iface);
g_free (rtspsrc->user_agent);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ g_mutex_clear (&(rtspsrc)->pause_lock);
+ g_cond_clear (&(rtspsrc)->open_end);
+#endif
+
if (rtspsrc->sdp) {
gst_sdp_message_free (rtspsrc->sdp);
rtspsrc->sdp = NULL;
case PROP_TIMEOUT:
rtspsrc->udp_timeout = g_value_get_uint64 (value);
break;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ case PROP_START_POSITION:
+ rtspsrc->start_position = g_value_get_uint64 (value);
+ break;
+ case PROP_RESUME_POSITION:
+ rtspsrc->last_pos = g_value_get_uint64 (value);
+ GST_DEBUG_OBJECT (rtspsrc, "src->last_pos value set to %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (rtspsrc->last_pos));
+ break;
+#endif
case PROP_TCP_TIMEOUT:
gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
break;
case PROP_TIMEOUT:
g_value_set_uint64 (value, rtspsrc->udp_timeout);
break;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ case PROP_START_POSITION:
+ g_value_set_uint64 (value, rtspsrc->start_position);
+ break;
+ case PROP_RESUME_POSITION:
+ g_value_set_uint64 (value, rtspsrc->last_pos);
+ break;
+#endif
case PROP_TCP_TIMEOUT:
{
guint64 timeout;
GstStructure *s;
const gchar *enc;
PtMapItem item;
-
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ const gchar *encoder, *mediatype;
+#endif
pt = atoi (gst_sdp_media_get_format (media, i));
GST_DEBUG_OBJECT (src, " looking at %d pt: %d", i, pt);
if (strcmp (enc, "X-ASF-PF") == 0)
stream->container = TRUE;
}
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ if ((mediatype = gst_structure_get_string (s, "media"))) {
+ GST_DEBUG_OBJECT (src, " mediatype : %s", mediatype);
+ if (!strcmp (mediatype, "video")) {
+ if ((encoder = gst_structure_get_string (s, "encoding-name"))) {
+ GST_DEBUG_OBJECT (src, " encoder : %s", encoder);
+ if ((!strcmp (encoder, "H261")) ||
+ (!strcmp (encoder, "H263")) ||
+ (!strcmp (encoder, "H263-1998"))
+ || (!strcmp (encoder, "H263-2000")) || (!strcmp (encoder, "H264"))
+ || (!strcmp (encoder, "MP4V-ES"))) {
+ src->is_video_codec_supported = TRUE;
+ GST_DEBUG_OBJECT (src, "Supported Video Codec %s", encoder);
+ } else {
+ GST_DEBUG_OBJECT (src, "Unsupported Video Codec %s", encoder);
+ }
+ }
+
+ src->video_codec = g_strdup (encoder);
+ src->video_frame_size =
+ g_strdup (gst_structure_get_string (s, "a-framesize"));
+ GST_DEBUG_OBJECT (src, "video_codec %s , video_frame_size %s ",
+ src->video_codec, src->video_frame_size);
+ } else if (!strcmp (mediatype, "audio")) {
+ if ((encoder = gst_structure_get_string (s, "encoding-name"))) {
+ GST_DEBUG_OBJECT (src, " encoder : %s", encoder);
+ if ((!strcmp (encoder, "MP4A-LATM")) ||
+ (!strcmp (encoder, "AMR")) || (!strcmp (encoder, "AMR-WB"))
+ || (!strcmp (encoder, "AMR-NB"))
+ || (!strcmp (encoder, "mpeg4-generic"))
+ || (!strcmp (encoder, "MPEG4-GENERIC"))
+ || (!strcmp (encoder, "QCELP")) || ((strstr (encoder, "G726"))
+ || (strstr (encoder, "PCMU")))) {
+ src->is_audio_codec_supported = TRUE;
+ GST_DEBUG_OBJECT (src, "Supported Audio Codec %s", encoder);
+ } else {
+ GST_DEBUG_OBJECT (src, "Unsupported Audio Codec %s", encoder);
+ }
+ }
+
+ src->audio_codec = g_strdup (encoder);
+ GST_DEBUG_OBJECT (src, "audio_codec %s ", src->audio_codec);
+ }
+ }
+#endif
/* Merge in global caps */
/* Intersect will merge in missing fields to the current caps */
for (i = 0; i < 2; i++) {
if (stream->udpsrc[i]) {
gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
- gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
+ if (gst_object_has_as_parent (GST_OBJECT (stream->udpsrc[i]),
+ GST_OBJECT (src)))
+ gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
gst_object_unref (stream->udpsrc[i]);
}
if (stream->channelpad[i])
if (stream->udpsink[i]) {
gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
- gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
+ if (gst_object_has_as_parent (GST_OBJECT (stream->udpsink[i]),
+ GST_OBJECT (src)))
+ gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
gst_object_unref (stream->udpsink[i]);
}
}
{
GList *walk;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ GST_WARNING_OBJECT (src, "Setting [%s] element state to: %s \n",
+ GST_ELEMENT_NAME (GST_ELEMENT_CAST (src)),
+ gst_element_state_get_name (state));
+#endif
if (src->manager)
gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
}
static void
-gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
+gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing,
+ guint32 seqnum)
{
GstEvent *event;
gint cmd;
if (flush) {
event = gst_event_new_flush_start ();
+ gst_event_set_seqnum (event, seqnum);
GST_DEBUG_OBJECT (src, "start flush");
cmd = CMD_WAIT;
state = GST_STATE_PAUSED;
} else {
event = gst_event_new_flush_stop (FALSE);
+ gst_event_set_seqnum (event, seqnum);
GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
cmd = CMD_LOOP;
if (playing)
* blocking in preroll). */
if (flush) {
GST_DEBUG_OBJECT (src, "starting flush");
- gst_rtspsrc_flush (src, TRUE, FALSE);
+ gst_rtspsrc_flush (src, TRUE, FALSE, gst_event_get_seqnum (event));
} else {
if (src->task) {
gst_task_pause (src->task);
if (flush) {
/* if we started flush, we stop now */
GST_DEBUG_OBJECT (src, "stopping flush");
- gst_rtspsrc_flush (src, FALSE, playing);
+ gst_rtspsrc_flush (src, FALSE, playing, gst_event_get_seqnum (event));
}
/* now we did the seek and can activate the new segment values */
}
}
+static GstPadProbeReturn
+udpsrc_probe_cb (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
+{
+ guint32 *segment_seqnum = user_data;
+
+ switch (GST_EVENT_TYPE (info->data)) {
+ case GST_EVENT_SEGMENT:
+ if (!gst_event_is_writable (info->data))
+ info->data = gst_event_make_writable (info->data);
+
+ *segment_seqnum = gst_event_get_seqnum (info->data);
+ default:
+ break;
+ }
+
+ return GST_PAD_PROBE_OK;
+}
+
static gboolean
copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
{
}
static void
-on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
+on_timeout_common (GObject * session, GObject * source, GstRTSPStream * stream)
{
GstRTSPSrc *src = stream->parent;
guint ssrc;
}
static void
+on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GstRTSPSrc *src = stream->parent;
+
+ /* timeout, post element message */
+ gst_element_post_message (GST_ELEMENT_CAST (src),
+ gst_message_new_element (GST_OBJECT_CAST (src),
+ gst_structure_new ("GstRTSPSrcTimeout",
+ "cause", G_TYPE_ENUM, GST_RTSP_SRC_TIMEOUT_CAUSE_RTCP,
+ "stream-number", G_TYPE_INT, stream->id, "ssrc", G_TYPE_UINT,
+ stream->ssrc, NULL)));
+
+ on_timeout_common (session, source, stream);
+}
+
+static void
on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
{
GstRTSPStream *stream;
if (!(src->manager = gst_element_factory_make (manager, "manager")))
goto manager_failed;
}
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ if (g_strcmp0 (manager, "rtpbin") == 0) {
+ /* set for player rtsp buffering */
+ g_object_set (src->manager, "use-rtsp-buffering", TRUE, NULL);
+ }
+#endif
/* we manage this element */
gst_element_set_locked_state (src->manager, TRUE);
g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
stream);
- g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
- stream);
+ g_signal_connect (rtpsession, "on-bye-timeout",
+ (GCallback) on_timeout_common, stream);
g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
stream);
g_signal_connect (rtpsession, "on-ssrc-active",
GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocked, src, NULL);
+ gst_pad_add_probe (stream->blockedpad,
+ GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM, udpsrc_probe_cb,
+ &(stream->segment_seqnum[0]), NULL);
+
if (stream->channelpad[0]) {
GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
/* configure for UDP delivery, we need to connect the UDP pads to
GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
+ gst_pad_add_probe (pad,
+ GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM, udpsrc_probe_cb,
+ &(stream->segment_seqnum[1]), NULL);
gst_pad_link_full (pad, stream->channelpad[1],
GST_PAD_LINK_CHECK_NOTHING);
gst_object_unref (pad);
goto done;
if (stream->udpsrc[0]) {
- gst_event_ref (event);
- res = gst_element_send_event (stream->udpsrc[0], event);
+ GstEvent *sent_event;
+
+ if (GST_EVENT_TYPE (event) == GST_EVENT_EOS) {
+ sent_event = gst_event_new_eos ();
+ gst_event_set_seqnum (sent_event, stream->segment_seqnum[0]);
+ } else {
+ sent_event = gst_event_ref (event);
+ }
+
+ res = gst_element_send_event (stream->udpsrc[0], sent_event);
} else if (stream->channelpad[0]) {
gst_event_ref (event);
if (GST_PAD_IS_SRC (stream->channelpad[0]))
}
if (stream->udpsrc[1]) {
- gst_event_ref (event);
- res &= gst_element_send_event (stream->udpsrc[1], event);
+ GstEvent *sent_event;
+
+ if (GST_EVENT_TYPE (event) == GST_EVENT_EOS) {
+ sent_event = gst_event_new_eos ();
+ if (stream->segment_seqnum[1] != GST_SEQNUM_INVALID) {
+ gst_event_set_seqnum (sent_event, stream->segment_seqnum[1]);
+ }
+ } else {
+ sent_event = gst_event_ref (event);
+ }
+
+ res &= gst_element_send_event (stream->udpsrc[1], sent_event);
} else if (stream->channelpad[1]) {
gst_event_ref (event);
if (GST_PAD_IS_SRC (stream->channelpad[1]))
{
gchar *str = gst_rtsp_strresult (res);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_SERVER,
+ "Could not receive message.");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
+#endif
g_free (str);
gst_rtsp_message_unset (&message);
{
gchar *str = gst_rtsp_strresult (res);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_SERVICE_UNAVAILABLE,
+ "Could not handle server message.");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not handle server message. (%s)", str));
+#endif
g_free (str);
gst_rtsp_message_unset (&message);
return GST_FLOW_ERROR;
src->conninfo.connected = FALSE;
if (res != GST_RTSP_EINTR) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+ "Could not connect to server.");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
("Could not connect to server. (%s)", str));
+#endif
g_free (str);
ret = GST_FLOW_ERROR;
} else {
{
gchar *str = gst_rtsp_strresult (res);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_SERVER_DISCONNECTED,
+ "Could not receive message.");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
+#endif
g_free (str);
return GST_FLOW_ERROR;
}
gst_rtsp_message_unset (&message);
if (res != GST_RTSP_EINTR) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src,
+ GST_RTSPSRC_ERROR_SERVICE_UNAVAILABLE,
+ "Could not handle server message.");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not handle server message. (%s)", str));
+#endif
g_free (str);
ret = GST_FLOW_ERROR;
} else {
{
src->cur_protocols = 0;
/* no transport possible, post an error and stop */
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_TRANSPORT,
+ "Could not receive any UDP packets for seconds, maybe your firewall is blocking it. No other protocols to try.");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not receive any UDP packets for %.4f seconds, maybe your "
"firewall is blocking it. No other protocols to try.",
gst_guint64_to_gdouble (src->udp_timeout) / 1000000.0));
+#endif
return GST_RTSP_ERROR;
}
open_failed:
static void
gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
{
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ GstMessage *s;
+ GST_WARNING_OBJECT (src, "Got cmd %s", cmd_to_string (cmd));
+#endif
+
switch (cmd) {
case CMD_OPEN:
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ GST_DEBUG_OBJECT (src,
+ "rtsp_duration %" GST_TIME_FORMAT
+ ", rtsp_audio_codec %s , rtsp_video_codec %s , rtsp_video_frame_size %s",
+ GST_TIME_ARGS (src->segment.duration), src->audio_codec,
+ src->video_codec, src->video_frame_size);
+
+ /* post message */
+ s = gst_message_new_element (GST_OBJECT_CAST (src),
+ gst_structure_new ("rtspsrc_properties",
+ "rtsp_duration", G_TYPE_UINT64, src->segment.duration,
+ "rtsp_audio_codec", G_TYPE_STRING, src->audio_codec,
+ "rtsp_video_codec", G_TYPE_STRING, src->video_codec,
+ "rtsp_video_frame_size", G_TYPE_STRING, src->video_frame_size,
+ NULL));
+
+ gst_element_post_message (GST_ELEMENT_CAST (src), s);
+#endif
GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ /* rtspsrc PAUSE state should be here for parsing sdp before PAUSE state changed. */
+ g_mutex_lock (&(src)->pause_lock);
+ g_cond_signal (&(src)->open_end);
+ g_mutex_unlock (&(src)->pause_lock);
+#endif
break;
case CMD_PLAY:
GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
switch (cmd) {
case CMD_OPEN:
GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ /* Ending conditional wait for pause when open fails.*/
+ g_mutex_lock (&(src)->pause_lock);
+ g_cond_signal (&(src)->open_end);
+ g_mutex_unlock (&(src)->pause_lock);
+ GST_WARNING_OBJECT (src,
+ "ending conditional wait for pause as open is failed.");
+#endif
break;
case CMD_PLAY:
GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
{
/* Output an error indicating that we couldn't connect because there were
* no supported authentication protocols */
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_NOT_AUTHORIZED,
+ "No supported authentication protocol was found");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
("No supported authentication protocol was found"));
+#endif
return FALSE;
}
no_user_pass:
gchar *str = gst_rtsp_strresult (res);
if (res != GST_RTSP_EINTR) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src,
+ GST_RTSPSRC_ERROR_SERVER_DISCONNECTED,
+ "Could not receive message.");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
+#endif
} else {
GST_WARNING_OBJECT (src, "receive interrupted");
}
gchar *str = gst_rtsp_strresult (res);
if (res != GST_RTSP_EINTR) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+ "Could not send message.");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
+#endif
} else {
GST_WARNING_OBJECT (src, "send interrupted");
}
switch (response->type_data.response.code) {
case GST_RTSP_STS_NOT_FOUND:
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+ "STS NOT FOUND");
+#else
RTSP_SRC_RESPONSE_ERROR (src, response, RESOURCE, NOT_FOUND,
"Not found");
+#endif
break;
case GST_RTSP_STS_UNAUTHORIZED:
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_NOT_AUTHORIZED,
+ "STS NOT AUTHORIZED");
+#else
RTSP_SRC_RESPONSE_ERROR (src, response, RESOURCE, NOT_AUTHORIZED,
"Unauthorized");
+#endif
break;
case GST_RTSP_STS_MOVED_PERMANENTLY:
case GST_RTSP_STS_MOVE_TEMPORARILY:
res = GST_RTSP_OK;
break;
default:
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_UNEXPECTED_MSG,
+ "Got error response from Server");
+#else
RTSP_SRC_RESPONSE_ERROR (src, response, RESOURCE, READ,
"Unhandled error");
+#endif
break;
}
/* if we return ERROR we should unset the response ourselves */
/* ERRORS */
no_describe:
{
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_METHOD_NOT_ALLOWED,
+ "Server does not support DESCRIBE.");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
("Server does not support DESCRIBE."));
+#endif
return FALSE;
}
no_setup:
{
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_METHOD_NOT_ALLOWED,
+ "Server does not support SETUP.");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
("Server does not support SETUP."));
+#endif
return FALSE;
}
}
/* ERRORS */
no_protocols:
{
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_INVALID_PROTOCOL,
+ "Could not connect to server, no protocols left");
+#else
/* no transport possible, post an error and stop */
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not connect to server, no protocols left"));
+#endif
return GST_RTSP_ERROR;
}
no_streams:
{
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONTENT_NOT_FOUND,
+ "SDP contains no streams");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("SDP contains no streams"));
+#endif
return GST_RTSP_ERROR;
}
create_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+ "Could not create request.");
+#else
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
+#endif
g_free (str);
goto cleanup_error;
}
setup_transport_failed:
{
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+ "Could not setup transport.");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Could not setup transport."));
+#endif
res = GST_RTSP_ERROR;
goto cleanup_error;
}
response_error:
{
+#ifndef TIZEN_FEATURE_RTSP_MODIFICATION
const gchar *str = gst_rtsp_status_as_text (code);
+#endif
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_UNEXPECTED_MSG,
+ "Error from Server .");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Error (%d): %s", code, GST_STR_NULL (str)));
+#endif
res = GST_RTSP_ERROR;
goto cleanup_error;
}
gchar *str = gst_rtsp_strresult (res);
if (res != GST_RTSP_EINTR) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+ "Could not send message.");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
+#endif
} else {
GST_WARNING_OBJECT (src, "send interrupted");
}
{
/* none of the available error codes is really right .. */
if (unsupported_real) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src,
+ GST_RTSPSRC_ERROR_UNSUPPORTED_MEDIA_TYPE,
+ "No supported stream was found. You might need to install a GStreamer RTSP extension plugin for Real media streams.");
+#else
GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
(_("No supported stream was found. You might need to install a "
"GStreamer RTSP extension plugin for Real media streams.")),
(NULL));
+#endif
} else {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src,
+ GST_RTSPSRC_ERROR_UNSUPPORTED_MEDIA_TYPE,
+ "No supported stream was found. You might need to allow more transport protocols or may otherwise be missing the right GStreamer RTSP extension plugin.");
+#else
GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
(_("No supported stream was found. You might need to allow "
"more transport protocols or may otherwise be missing "
"the right GStreamer RTSP extension plugin.")), (NULL));
+#endif
}
return GST_RTSP_ERROR;
}
/* we need to start playback without clipping from the position reported by
* the server */
segment->start = seconds;
+#ifndef TIZEN_FEATURE_RTSP_MODIFICATION
+/*
+The range-min points to the start of the segment , not the current position.
+After getting the current position from MSL during normal pause/resume or during seek , we should not
+update the segment->position again with the rtp header npt timestamp.
+*/
segment->position = seconds;
+#endif
if (therange->max.type == GST_RTSP_TIME_NOW)
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ seconds = 0;
+#else
seconds = -1;
+#endif
else if (therange->max.type == GST_RTSP_TIME_END)
seconds = -1;
else
src->control = g_strdup (control);
}
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ src->is_audio_codec_supported = FALSE;
+ src->is_video_codec_supported = FALSE;
+#endif
+
/* create streams */
n_streams = gst_sdp_message_medias_len (sdp);
for (i = 0; i < n_streams; i++) {
}
src->state = GST_RTSP_STATE_INIT;
-
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ /* Check for the support for the Media codecs */
+ if ((!src->is_audio_codec_supported) && (!src->is_video_codec_supported)) {
+ GST_ERROR_OBJECT (src, "UnSupported Media Type !!!! \n");
+ goto unsupported_file_type;
+ } else {
+ GST_DEBUG_OBJECT (src, "Supported Media Type. \n");
+ }
+#endif
/* setup streams */
if ((res = gst_rtspsrc_setup_streams_start (src, async)) < 0)
goto setup_failed;
gst_rtspsrc_cleanup (src);
return res;
}
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+unsupported_file_type:
+ {
+ gst_rtspsrc_post_error_message (src,
+ GST_RTSPSRC_ERROR_UNSUPPORTED_MEDIA_TYPE,
+ "No supported stream was found");
+ res = GST_RTSP_ERROR;
+ gst_rtspsrc_cleanup (src);
+ return res;
+ }
+#endif
}
static GstRTSPResult
/* ERRORS */
no_url:
{
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_INVALID_URL,
+ "No valid RTSP URL was provided");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
("No valid RTSP URL was provided"));
+#endif
goto cleanup_error;
}
connect_failed:
gchar *str = gst_rtsp_strresult (res);
if (res != GST_RTSP_EINTR) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+ "Failed to connect.");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
("Failed to connect. (%s)", str));
+#endif
} else {
GST_WARNING_OBJECT (src, "connect interrupted");
}
{
gchar *str = gst_rtsp_strresult (res);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+ "Could not create request.");
+#else
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
+#endif
g_free (str);
goto cleanup_error;
}
}
wrong_content_type:
{
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_OPTION_NOT_SUPPORTED,
+ "Server does not support SDP. ");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Server does not support SDP, got %s.", respcont));
+#endif
res = GST_RTSP_ERROR;
goto cleanup_error;
}
no_describe:
{
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_METHOD_NOT_ALLOWED,
+ "Server can not provide an SDP.");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Server can not provide an SDP."));
+#endif
res = GST_RTSP_ERROR;
goto cleanup_error;
}
{
gchar *str = gst_rtsp_strresult (res);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+ "Could not create request.");
+#else
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
+#endif
g_free (str);
goto close;
}
gst_rtsp_message_unset (&request);
if (res != GST_RTSP_EINTR) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+ "Could not send message.");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
+#endif
} else {
GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
}
gen_range_header (GstRTSPSrc * src, GstSegment * segment)
{
gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
-
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ if (src->start_position != 0 && segment->position == 0) {
+ segment->position = src->start_position;
+ src->start_position = 0;
+ }
+#endif
if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
g_strlcpy (val_str, "now", sizeof (val_str));
} else {
((gdouble) segment->position) / GST_SECOND);
}
}
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ GST_DEBUG_OBJECT (src, "Range Header Added : npt=%s-", val_str);
+#endif
return g_strdup_printf ("npt=%s-", val_str);
}
goto create_request_failed;
if (src->need_range && src->seekable >= 0.0) {
+#ifndef TIZEN_FEATURE_RTSP_MODIFICATION
hval = gen_range_header (src, segment);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
+#endif
/* store the newsegment event so it can be sent from the streaming thread. */
src->need_segment = TRUE;
}
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ else {
+/*
+ Updating position with the MSL current position as gst_rtspsrc_get_position() does not return correct position.
+*/
+ GST_DEBUG_OBJECT (src,
+ " During normal pause-resume , segment->position=%" GST_TIME_FORMAT
+ ",src->start_position=%" GST_TIME_FORMAT,
+ GST_TIME_ARGS (segment->position),
+ GST_TIME_ARGS (src->start_position));
+ segment->position = src->last_pos;
+ }
+
+/*
+ Sending the npt range request for each play request for updating the segment position properly.
+*/
+ hval = gen_range_header (src, segment);
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
+#endif
if (segment->rate != 1.0) {
gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
{
gchar *str = gst_rtsp_strresult (res);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+ "Could not create request. ");
+#else
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
+#endif
g_free (str);
goto done;
}
gst_rtsp_message_unset (&request);
if (res != GST_RTSP_EINTR) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+ "Could not send message.");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
+#endif
} else {
GST_WARNING_OBJECT (src, "PLAY interrupted");
}
{
gchar *str = gst_rtsp_strresult (res);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+ "Could not create request.");
+#else
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
+#endif
g_free (str);
goto done;
}
gst_rtsp_message_unset (&request);
if (res != GST_RTSP_EINTR) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+ "Could not send message.");
+#else
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
+#endif
} else {
GST_WARNING_OBJECT (src, "PAUSE interrupted");
}
{
GstRTSPSrc *rtspsrc;
GstStateChangeReturn ret;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ guint64 end_time;
+#endif
rtspsrc = GST_RTSPSRC (element);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ GST_WARNING_OBJECT (rtspsrc, "State change transition: %d \n", transition);
+#endif
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
ret = GST_STATE_CHANGE_SUCCESS;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ /* don't change to PAUSE state before complete stream opend.
+ see gst_rtspsrc_loop_complete_cmd() */
+ g_mutex_lock (&(rtspsrc)->pause_lock);
+ end_time = g_get_monotonic_time () + 10 * G_TIME_SPAN_SECOND;
+ if (!g_cond_wait_until (&(rtspsrc)->open_end, &(rtspsrc)->pause_lock,
+ end_time)) {
+ GST_WARNING_OBJECT (rtspsrc,
+ "time out: stream opend is not completed yet..");
+ }
+ g_mutex_unlock (&(rtspsrc)->pause_lock);
+#endif
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
projects / platform / upstream / gst-plugins-good.git / blobdiff