#define DEFAULT_MAX_STREAMS G_MAXUINT
#define DEFAULT_MAX_TS_OFFSET_ADJUSTMENT G_GUINT64_CONSTANT(0)
#define DEFAULT_MAX_TS_OFFSET G_GINT64_CONSTANT(3000000000)
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+#define DEFAULT_RTSP_USE_BUFFERING FALSE
+#endif
enum
{
PROP_MAX_TS_OFFSET,
PROP_FEC_DECODERS,
PROP_FEC_ENCODERS,
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ PROP_USE_RTSP_BUFFERING /* use for player RTSP buffering */
+#endif
};
#define GST_RTP_BIN_RTCP_SYNC_TYPE (gst_rtp_bin_rtcp_sync_get_type())
gulong buffer_ptreq_sig;
gulong buffer_ntpstop_sig;
gint percent;
-
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gint prev_percent;
+#endif
/* the PT demuxer of the SSRC */
GstElement *demux;
gulong demux_newpad_sig;
create_stream (GstRtpBinSession * session, guint32 ssrc)
{
GstElement *buffer, *demux = NULL;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ GstElement *queue2 = NULL;
+#endif
GstRtpBinStream *stream;
GstRtpBin *rtpbin;
GstState target;
if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
goto no_demux;
}
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ if (rtpbin->use_rtsp_buffering &&
+ rtpbin->buffer_mode == RTP_JITTER_BUFFER_MODE_SLAVE) {
+ if (!(queue2 = gst_element_factory_make ("queue2", NULL)))
+ goto no_queue2;
+ }
+#endif
stream = g_new0 (GstRtpBinStream, 1);
stream->ssrc = ssrc;
stream->rt_delta = 0;
stream->rtp_delta = 0;
stream->percent = 100;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ stream->prev_percent = 0;
+#endif
stream->clock_base = -100 * GST_SECOND;
session->streams = g_slist_prepend (session->streams, stream);
g_object_set (buffer, "max-ts-offset-adjustment",
rtpbin->max_ts_offset_adjustment, NULL);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ /* configure queue2 to use live buffering */
+ if (queue2) {
+ g_object_set_data (G_OBJECT (queue2), "GstRTPBin.stream", stream);
+ g_object_set (queue2, "use-buffering", TRUE, NULL);
+ g_object_set (queue2, "buffer-mode", GST_BUFFERING_LIVE, NULL);
+ }
+#endif
g_signal_emit (rtpbin, gst_rtp_bin_signals[SIGNAL_NEW_JITTERBUFFER], 0,
buffer, session->id, ssrc);
if (!rtpbin->ignore_pt)
gst_bin_add (GST_BIN_CAST (rtpbin), demux);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ if (queue2)
+ gst_bin_add (GST_BIN_CAST (rtpbin), queue2);
+#endif
+
/* link stuff */
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ if (queue2) {
+ gst_element_link_pads_full (buffer, "src", queue2, "sink",
+ GST_PAD_LINK_CHECK_NOTHING);
+ if (demux) {
+ gst_element_link_pads_full (queue2, "src", demux, "sink",
+ GST_PAD_LINK_CHECK_NOTHING);
+ }
+ } else if (demux) {
+ gst_element_link_pads_full (buffer, "src", demux, "sink",
+ GST_PAD_LINK_CHECK_NOTHING);
+ }
+#else
if (demux)
gst_element_link_pads_full (buffer, "src", demux, "sink",
GST_PAD_LINK_CHECK_NOTHING);
-
+#endif
if (rtpbin->buffering) {
guint64 last_out;
gst_element_set_state (buffer, target);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ if (queue2)
+ gst_element_set_state (queue2, target);
+#endif
+
return stream;
/* ERRORS */
g_warning ("rtpbin: could not create rtpptdemux element");
return NULL;
}
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+no_queue2:
+ {
+ gst_object_unref (buffer);
+ gst_object_unref (demux);
+ g_warning ("rtpbin: could not create queue2 element");
+ return NULL;
+ }
+#endif
}
/* called with RTP_BIN_LOCK */
"fec-encoders='fec,0=\"rtpst2022-1-fecenc\\ rows\\=5\\ columns\\=5\";'",
GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ g_object_class_install_property (gobject_class, PROP_USE_RTSP_BUFFERING,
+ g_param_spec_boolean ("use-rtsp-buffering", "Use RTSP buffering",
+ "Use RTSP buffering in RTP_JITTER_BUFFER_MODE_SLAVE buffer mode",
+ DEFAULT_RTSP_USE_BUFFERING,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+#endif
+
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_bin_change_state);
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_rtp_bin_request_new_pad);
rtpbin->max_ts_offset_adjustment = DEFAULT_MAX_TS_OFFSET_ADJUSTMENT;
rtpbin->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
rtpbin->max_ts_offset_is_set = FALSE;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ rtpbin->use_rtsp_buffering = FALSE;
+#endif
/* some default SDES entries */
cname = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
rtpbin = GST_RTP_BIN (object);
switch (prop_id) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ case PROP_USE_RTSP_BUFFERING:
+ GST_RTP_BIN_LOCK (rtpbin);
+ rtpbin->use_rtsp_buffering = g_value_get_boolean (value);
+ GST_RTP_BIN_UNLOCK (rtpbin);
+ break;
+#endif
case PROP_LATENCY:
GST_RTP_BIN_LOCK (rtpbin);
rtpbin->latency_ms = g_value_get_uint (value);
rtpbin = GST_RTP_BIN (object);
switch (prop_id) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ case PROP_USE_RTSP_BUFFERING:
+ GST_RTP_BIN_LOCK (rtpbin);
+ g_value_set_boolean (value, rtpbin->use_rtsp_buffering);
+ GST_RTP_BIN_UNLOCK (rtpbin);
+ break;
+#endif
case PROP_LATENCY:
GST_RTP_BIN_LOCK (rtpbin);
g_value_set_uint (value, rtpbin->latency_ms);
gint min_percent = 100;
GSList *sessions, *streams;
GstRtpBinStream *stream;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gboolean buffering_flag = FALSE, update_buffering_status = TRUE;
+#endif
gboolean change = FALSE, active = FALSE;
GstClockTime min_out_time;
GstBufferingMode mode;
streams = g_slist_next (streams)) {
GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ if (rtpbin->use_rtsp_buffering &&
+ rtpbin->buffer_mode == RTP_JITTER_BUFFER_MODE_SLAVE) {
+ GstPad *temp_pad_src = NULL;
+ GstCaps *temp_caps_src = NULL;
+ GstStructure *caps_structure;
+ const gchar *caps_str_media = NULL;
+ temp_pad_src = gst_element_get_static_pad (stream->buffer, "src");
+ temp_caps_src = gst_pad_get_current_caps (temp_pad_src);
+ GST_DEBUG_OBJECT (bin,
+ "stream %p percent %d : temp_caps_src=%" GST_PTR_FORMAT,
+ stream, stream->percent, temp_caps_src);
+ if (temp_caps_src) {
+ caps_structure = gst_caps_get_structure (temp_caps_src, 0);
+ caps_str_media =
+ gst_structure_get_string (caps_structure, "media");
+ if (caps_str_media != NULL) {
+ if ((strcmp (caps_str_media, "video") != 0)
+ && (strcmp (caps_str_media, "audio") != 0)) {
+ GST_DEBUG_OBJECT (bin,
+ "Non Audio/Video Stream.. ignoring the same !!");
+ gst_caps_unref (temp_caps_src);
+ gst_object_unref (temp_pad_src);
+ continue;
+ } else if (stream->percent >= 100) {
+ /* Most of the time buffering icon displays in rtsp playback.
+ Optimizing the buffering updation code. Whenever any stream percentage
+ reaches 100 do not post buffering messages. */
+ if (stream->prev_percent < 100)
+ buffering_flag = TRUE;
+ else
+ update_buffering_status = FALSE;
+ }
+ }
+ gst_caps_unref (temp_caps_src);
+ }
+ gst_object_unref (temp_pad_src);
+ /* Updating prev stream percentage */
+ stream->prev_percent = stream->percent;
+ } else {
+ GST_DEBUG_OBJECT (bin, "stream %p percent %d", stream,
+ stream->percent);
+ }
+#else
GST_DEBUG_OBJECT (bin, "stream %p percent %d", stream,
stream->percent);
+#endif
/* find min percent */
if (min_percent > stream->percent)
min_percent = stream->percent;
}
GST_DEBUG_OBJECT (bin, "min percent %d", min_percent);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ if (!(rtpbin->use_rtsp_buffering &&
+ rtpbin->buffer_mode == RTP_JITTER_BUFFER_MODE_SLAVE)) {
+#endif
if (rtpbin->buffering) {
if (min_percent == 100) {
rtpbin->buffering = FALSE;
change = TRUE;
}
}
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ }
+#endif
GST_RTP_BIN_UNLOCK (rtpbin);
gst_message_unref (message);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ if (rtpbin->use_rtsp_buffering &&
+ rtpbin->buffer_mode == RTP_JITTER_BUFFER_MODE_SLAVE) {
+ if (update_buffering_status == FALSE)
+ break;
+ if (buffering_flag) {
+ min_percent = 100;
+ GST_DEBUG_OBJECT (bin, "forcefully change min_percent to 100!!!");
+ }
+ }
+#endif
/* make a new buffering message with the min value */
message =
gst_message_new_buffering (GST_OBJECT_CAST (bin), min_percent);
gst_message_set_buffering_stats (message, mode, avg_in, avg_out,
buffering_left);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ if (rtpbin->use_rtsp_buffering &&
+ rtpbin->buffer_mode == RTP_JITTER_BUFFER_MODE_SLAVE)
+ goto slave_buffering;
+#endif
if (G_UNLIKELY (change)) {
GstClock *clock;
guint64 running_time = 0;
GST_RTP_BIN_UNLOCK (rtpbin);
}
}
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+slave_buffering:
+#endif
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
projects / platform / upstream / gstreamer.git / blobdiff