*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
#include <string.h>
+#include <stdlib.h>
#include <gst/app/gstappsrc.h>
#include <gst/app/gstappsink.h>
#include "rtsp-media.h"
-#define DEFAULT_SHARED FALSE
-#define DEFAULT_REUSABLE FALSE
+#define DEFAULT_SHARED FALSE
+#define DEFAULT_REUSABLE FALSE
+#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_TCP
+//#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP_MCAST
+#define DEFAULT_EOS_SHUTDOWN FALSE
+#define DEFAULT_BUFFER_SIZE 0x80000
/* define to dump received RTCP packets */
#undef DUMP_STATS
PROP_0,
PROP_SHARED,
PROP_REUSABLE,
+ PROP_PROTOCOLS,
+ PROP_EOS_SHUTDOWN,
+ PROP_BUFFER_SIZE,
PROP_LAST
};
enum
{
+ SIGNAL_NEW_STREAM,
+ SIGNAL_PREPARED,
SIGNAL_UNPREPARED,
+ SIGNAL_NEW_STATE,
SIGNAL_LAST
};
-static GQuark ssrc_stream_map_key;
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
+#define GST_CAT_DEFAULT rtsp_media_debug
-static void gst_rtsp_media_get_property (GObject *object, guint propid,
- GValue *value, GParamSpec *pspec);
-static void gst_rtsp_media_set_property (GObject *object, guint propid,
- const GValue *value, GParamSpec *pspec);
+static void gst_rtsp_media_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
static void gst_rtsp_media_finalize (GObject * obj);
-static gpointer do_loop (GstRTSPMediaClass *klass);
-static gboolean default_handle_message (GstRTSPMedia *media, GstMessage *message);
-static gboolean default_unprepare (GstRTSPMedia *media);
-static void unlock_streams (GstRTSPMedia *media);
+static gpointer do_loop (GstRTSPMediaClass * klass);
+static gboolean default_handle_message (GstRTSPMedia * media,
+ GstMessage * message);
+static void finish_unprepare (GstRTSPMedia * media);
+static gboolean default_unprepare (GstRTSPMedia * media);
static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
{
GObjectClass *gobject_class;
- GError *error = NULL;
gobject_class = G_OBJECT_CLASS (klass);
gobject_class->finalize = gst_rtsp_media_finalize;
g_object_class_install_property (gobject_class, PROP_SHARED,
- g_param_spec_boolean ("shared", "Shared", "If this media pipeline can be shared",
- DEFAULT_SHARED, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_param_spec_boolean ("shared", "Shared",
+ "If this media pipeline can be shared", DEFAULT_SHARED,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_REUSABLE,
g_param_spec_boolean ("reusable", "Reusable",
"If this media pipeline can be reused after an unprepare",
DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
+ g_param_spec_flags ("protocols", "Protocols",
+ "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
+ DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
+ g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
+ "Send an EOS event to the pipeline before unpreparing",
+ DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
+ g_param_spec_uint ("buffer-size", "Buffer Size",
+ "The kernel UDP buffer size to use", 0, G_MAXUINT,
+ DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
+ g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+ G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
+ g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
+
+ gst_rtsp_media_signals[SIGNAL_PREPARED] =
+ g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+ G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
+ g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
+
gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
+ gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
+ g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+ G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
+ g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 0, G_TYPE_INT);
+
klass->context = g_main_context_new ();
klass->loop = g_main_loop_new (klass->context, TRUE);
- klass->thread = g_thread_create ((GThreadFunc) do_loop, klass, TRUE, &error);
- if (error != NULL) {
- g_critical ("could not start bus thread: %s", error->message);
- }
+ GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
+
+ klass->thread = g_thread_new ("Bus Thread", (GThreadFunc) do_loop, klass);
+
klass->handle_message = default_handle_message;
klass->unprepare = default_unprepare;
-
- ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
}
static void
gst_rtsp_media_init (GstRTSPMedia * media)
{
- media->streams = g_array_new (FALSE, TRUE, sizeof (GstRTSPMediaStream *));
- media->is_live = FALSE;
- media->buffering = FALSE;
-}
-
-static void
-gst_rtsp_media_stream_free (GstRTSPMediaStream *stream)
-{
- if (stream->session)
- g_object_unref (stream->session);
-
- if (stream->caps)
- gst_caps_unref (stream->caps);
-
- if (stream->send_rtp_sink)
- gst_object_unref (stream->send_rtp_sink);
- if (stream->send_rtp_src)
- gst_object_unref (stream->send_rtp_src);
- if (stream->send_rtcp_src)
- gst_object_unref (stream->send_rtcp_src);
- if (stream->recv_rtcp_sink)
- gst_object_unref (stream->recv_rtcp_sink);
- if (stream->recv_rtp_sink)
- gst_object_unref (stream->recv_rtp_sink);
-
- g_list_free (stream->transports);
-
- g_free (stream);
+ media->streams = g_ptr_array_new_with_free_func (g_object_unref);
+ g_mutex_init (&media->lock);
+ g_cond_init (&media->cond);
+ g_rec_mutex_init (&media->state_lock);
+
+ media->shared = DEFAULT_SHARED;
+ media->reusable = DEFAULT_REUSABLE;
+ media->protocols = DEFAULT_PROTOCOLS;
+ media->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
+ media->buffer_size = DEFAULT_BUFFER_SIZE;
}
static void
gst_rtsp_media_finalize (GObject * obj)
{
GstRTSPMedia *media;
- guint i;
media = GST_RTSP_MEDIA (obj);
- g_message ("finalize media %p", media);
-/*
- if (media->pipeline) {
- unlock_streams (media);
- gst_element_set_state (media->pipeline, GST_STATE_NULL);
- gst_object_unref (media->pipeline);
- }
-*/
- for (i = 0; i < media->streams->len; i++) {
- GstRTSPMediaStream *stream;
+ GST_INFO ("finalize media %p", media);
- stream = g_array_index (media->streams, GstRTSPMediaStream *, i);
+ gst_rtsp_media_unprepare (media);
- gst_rtsp_media_stream_free (stream);
- }
- g_array_free (media->streams, TRUE);
+ g_ptr_array_unref (media->streams);
- g_list_foreach (media->dynamic, (GFunc) gst_object_unref, NULL);
- g_list_free (media->dynamic);
+ g_list_free_full (media->dynamic, gst_object_unref);
if (media->source) {
g_source_destroy (media->source);
g_source_unref (media->source);
}
+ if (media->auth)
+ g_object_unref (media->auth);
+ if (media->pool)
+ g_object_unref (media->pool);
+ g_mutex_clear (&media->lock);
+ g_cond_clear (&media->cond);
+ g_rec_mutex_clear (&media->state_lock);
G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
}
static void
-gst_rtsp_media_get_property (GObject *object, guint propid,
- GValue *value, GParamSpec *pspec)
+gst_rtsp_media_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
{
GstRTSPMedia *media = GST_RTSP_MEDIA (object);
case PROP_REUSABLE:
g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
break;
+ case PROP_PROTOCOLS:
+ g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
+ break;
+ case PROP_EOS_SHUTDOWN:
+ g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
+ break;
+ case PROP_BUFFER_SIZE:
+ g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
static void
-gst_rtsp_media_set_property (GObject *object, guint propid,
- const GValue *value, GParamSpec *pspec)
+gst_rtsp_media_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
{
GstRTSPMedia *media = GST_RTSP_MEDIA (object);
case PROP_REUSABLE:
gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
break;
+ case PROP_PROTOCOLS:
+ gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
+ break;
+ case PROP_EOS_SHUTDOWN:
+ gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
+ break;
+ case PROP_BUFFER_SIZE:
+ gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
static gpointer
-do_loop (GstRTSPMediaClass *klass)
+do_loop (GstRTSPMediaClass * klass)
{
- g_message ("enter mainloop");
+ GST_INFO ("enter mainloop");
g_main_loop_run (klass->loop);
- g_message ("exit mainloop");
+ GST_INFO ("exit mainloop");
return NULL;
}
+/* must be called with state lock */
static void
-collect_media_stats (GstRTSPMedia *media)
+collect_media_stats (GstRTSPMedia * media)
{
- GstFormat format;
gint64 position, duration;
media->range.unit = GST_RTSP_RANGE_NPT;
+ GST_INFO ("collect media stats");
+
if (media->is_live) {
media->range.min.type = GST_RTSP_TIME_NOW;
media->range.min.seconds = -1;
media->range.max.type = GST_RTSP_TIME_END;
media->range.max.seconds = -1;
- }
- else {
+ } else {
/* get the position */
- format = GST_FORMAT_TIME;
- if (!gst_element_query_position (media->pipeline, &format, &position)) {
- g_message ("position query failed");
+ if (!gst_element_query_position (media->pipeline, GST_FORMAT_TIME,
+ &position)) {
+ GST_INFO ("position query failed");
position = 0;
}
/* get the duration */
- format = GST_FORMAT_TIME;
- if (!gst_element_query_duration (media->pipeline, &format, &duration)) {
- g_message ("duration query failed");
+ if (!gst_element_query_duration (media->pipeline, GST_FORMAT_TIME,
+ &duration)) {
+ GST_INFO ("duration query failed");
duration = -1;
}
- g_message ("stats: position %"GST_TIME_FORMAT", duration %"GST_TIME_FORMAT,
- GST_TIME_ARGS (position), GST_TIME_ARGS (duration));
+ GST_INFO ("stats: position %" GST_TIME_FORMAT ", duration %"
+ GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (duration));
if (position == -1) {
media->range.min.type = GST_RTSP_TIME_NOW;
media->range.min.seconds = -1;
- }
- else {
+ } else {
media->range.min.type = GST_RTSP_TIME_SECONDS;
- media->range.min.seconds = ((gdouble)position) / GST_SECOND;
+ media->range.min.seconds = ((gdouble) position) / GST_SECOND;
}
if (duration == -1) {
media->range.max.type = GST_RTSP_TIME_END;
media->range.max.seconds = -1;
- }
- else {
+ } else {
media->range.max.type = GST_RTSP_TIME_SECONDS;
- media->range.max.seconds = ((gdouble)duration) / GST_SECOND;
+ media->range.max.seconds = ((gdouble) duration) / GST_SECOND;
}
}
}
* gst_rtsp_media_new:
*
* Create a new #GstRTSPMedia instance. The #GstRTSPMedia object contains the
- * element to produde RTP data for one or more related (audio/video/..)
+ * element to produce RTP data for one or more related (audio/video/..)
* streams.
*
* Returns: a new #GstRTSPMedia object.
* pipeline.
*/
void
-gst_rtsp_media_set_shared (GstRTSPMedia *media, gboolean shared)
+gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
{
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+ g_mutex_lock (&media->lock);
media->shared = shared;
+ g_mutex_unlock (&media->lock);
}
/**
* Returns: %TRUE if the media can be shared between clients.
*/
gboolean
-gst_rtsp_media_is_shared (GstRTSPMedia *media)
+gst_rtsp_media_is_shared (GstRTSPMedia * media)
{
+ gboolean res;
+
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
- return media->shared;
+ g_mutex_lock (&media->lock);
+ res = media->shared;
+ g_mutex_unlock (&media->lock);
+
+ return res;
}
/**
* been unprepared.
*/
void
-gst_rtsp_media_set_reusable (GstRTSPMedia *media, gboolean reusable)
+gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
{
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+ g_mutex_lock (&media->lock);
media->reusable = reusable;
+ g_mutex_unlock (&media->lock);
}
/**
* Returns: %TRUE if the media can be reused
*/
gboolean
-gst_rtsp_media_is_reusable (GstRTSPMedia *media)
+gst_rtsp_media_is_reusable (GstRTSPMedia * media)
{
+ gboolean res;
+
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
- return media->reusable;
+ g_mutex_lock (&media->lock);
+ res = media->reusable;
+ g_mutex_unlock (&media->lock);
+
+ return res;
}
/**
- * gst_rtsp_media_n_streams:
+ * gst_rtsp_media_set_protocols:
* @media: a #GstRTSPMedia
+ * @protocols: the new flags
*
- * Get the number of streams in this media.
- *
- * Returns: The number of streams.
+ * Configure the allowed lower transport for @media.
*/
-guint
-gst_rtsp_media_n_streams (GstRTSPMedia *media)
+void
+gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
{
- g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
- return media->streams->len;
+ g_mutex_lock (&media->lock);
+ media->protocols = protocols;
+ g_mutex_unlock (&media->lock);
}
/**
- * gst_rtsp_media_get_stream:
+ * gst_rtsp_media_get_protocols:
* @media: a #GstRTSPMedia
- * @idx: the stream index
*
- * Retrieve the stream with index @idx from @media.
+ * Get the allowed protocols of @media.
*
- * Returns: the #GstRTSPMediaStream at index @idx or %NULL when a stream with
- * that index did not exist.
+ * Returns: a #GstRTSPLowerTrans
*/
-GstRTSPMediaStream *
-gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx)
+GstRTSPLowerTrans
+gst_rtsp_media_get_protocols (GstRTSPMedia * media)
{
- GstRTSPMediaStream *res;
-
- g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+ GstRTSPLowerTrans res;
- if (idx < media->streams->len)
- res = g_array_index (media->streams, GstRTSPMediaStream *, idx);
- else
- res = NULL;
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
+ GST_RTSP_LOWER_TRANS_UNKNOWN);
+
+ g_mutex_lock (&media->lock);
+ res = media->protocols;
+ g_mutex_unlock (&media->lock);
return res;
}
/**
- * gst_rtsp_media_seek:
- * @stream: a #GstRTSPMediaStream
- * @range: a #GstRTSPTimeRange
+ * gst_rtsp_media_set_eos_shutdown:
+ * @media: a #GstRTSPMedia
+ * @eos_shutdown: the new value
*
- * Seek the pipeline to @range.
+ * Set or unset if an EOS event will be sent to the pipeline for @media before
+ * it is unprepared.
+ */
+void
+gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
+{
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ g_mutex_lock (&media->lock);
+ media->eos_shutdown = eos_shutdown;
+ g_mutex_unlock (&media->lock);
+}
+
+/**
+ * gst_rtsp_media_is_eos_shutdown:
+ * @media: a #GstRTSPMedia
*
- * Returns: %TRUE on success.
+ * Check if the pipeline for @media will send an EOS down the pipeline before
+ * unpreparing.
+ *
+ * Returns: %TRUE if the media will send EOS before unpreparing.
*/
gboolean
-gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range)
+gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
{
- GstSeekFlags flags;
gboolean res;
- gint64 start, stop;
- GstSeekType start_type, stop_type;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
- g_return_val_if_fail (range != NULL, FALSE);
-
- if (range->unit != GST_RTSP_RANGE_NPT)
- goto not_supported;
-
- /* depends on the current playing state of the pipeline. We might need to
- * queue this until we get EOS. */
- flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_KEY_UNIT;
- start_type = stop_type = GST_SEEK_TYPE_NONE;
-
- switch (range->min.type) {
- case GST_RTSP_TIME_NOW:
- start = -1;
- break;
- case GST_RTSP_TIME_SECONDS:
- /* only seek when something changed */
- if (media->range.min.seconds == range->min.seconds) {
- start = -1;
- } else {
- start = range->min.seconds * GST_SECOND;
- start_type = GST_SEEK_TYPE_SET;
- }
- break;
- case GST_RTSP_TIME_END:
- default:
- goto weird_type;
- }
- switch (range->max.type) {
- case GST_RTSP_TIME_SECONDS:
- /* only seek when something changed */
- if (media->range.max.seconds == range->max.seconds) {
- stop = -1;
- } else {
- stop = range->max.seconds * GST_SECOND;
- stop_type = GST_SEEK_TYPE_SET;
- }
- break;
- case GST_RTSP_TIME_END:
- stop = -1;
- stop_type = GST_SEEK_TYPE_SET;
- break;
- case GST_RTSP_TIME_NOW:
- default:
- goto weird_type;
- }
-
- if (start != -1 || stop != -1) {
- g_message ("seeking to %"GST_TIME_FORMAT" - %"GST_TIME_FORMAT,
- GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
-
- res = gst_element_seek (media->pipeline, 1.0, GST_FORMAT_TIME,
- flags, start_type, start, stop_type, stop);
-
- /* and block for the seek to complete */
- g_message ("done seeking %d", res);
- gst_element_get_state (media->pipeline, NULL, NULL, -1);
- g_message ("prerolled again");
-
- collect_media_stats (media);
- }
- else {
- g_message ("no seek needed");
- res = TRUE;
- }
+ g_mutex_lock (&media->lock);
+ res = media->eos_shutdown;
+ g_mutex_unlock (&media->lock);
return res;
-
- /* ERRORS */
-not_supported:
- {
- g_warning ("seek unit %d not supported", range->unit);
- return FALSE;
- }
-weird_type:
- {
- g_warning ("weird range type %d not supported", range->min.type);
- return FALSE;
- }
}
/**
- * gst_rtsp_media_stream_rtp:
- * @stream: a #GstRTSPMediaStream
- * @buffer: a #GstBuffer
- *
- * Handle an RTP buffer for the stream. This method is usually called when a
- * message has been received from a client using the TCP transport.
- *
- * This function takes ownership of @buffer.
+ * gst_rtsp_media_set_buffer_size:
+ * @media: a #GstRTSPMedia
+ * @size: the new value
*
- * Returns: a GstFlowReturn.
+ * Set the kernel UDP buffer size.
*/
-GstFlowReturn
-gst_rtsp_media_stream_rtp (GstRTSPMediaStream *stream, GstBuffer *buffer)
+void
+gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
{
- GstFlowReturn ret;
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
- ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[0]), buffer);
+ GST_LOG_OBJECT (media, "set buffer size %u", size);
- return ret;
+ g_mutex_lock (&media->lock);
+ media->buffer_size = size;
+ g_mutex_unlock (&media->lock);
}
/**
- * gst_rtsp_media_stream_rtcp:
- * @stream: a #GstRTSPMediaStream
- * @buffer: a #GstBuffer
- *
- * Handle an RTCP buffer for the stream. This method is usually called when a
- * message has been received from a client using the TCP transport.
+ * gst_rtsp_media_get_buffer_size:
+ * @media: a #GstRTSPMedia
*
- * This function takes ownership of @buffer.
+ * Get the kernel UDP buffer size.
*
- * Returns: a GstFlowReturn.
+ * Returns: the kernel UDP buffer size.
*/
-GstFlowReturn
-gst_rtsp_media_stream_rtcp (GstRTSPMediaStream *stream, GstBuffer *buffer)
+guint
+gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
{
- GstFlowReturn ret;
+ guint res;
- ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[1]), buffer);
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
- return ret;
+ g_mutex_unlock (&media->lock);
+ res = media->buffer_size;
+ g_mutex_unlock (&media->lock);
+
+ return res;
}
-/* Allocate the udp ports and sockets */
-static gboolean
-alloc_udp_ports (GstRTSPMediaStream * stream)
+/**
+ * gst_rtsp_media_set_auth:
+ * @media: a #GstRTSPMedia
+ * @auth: a #GstRTSPAuth
+ *
+ * configure @auth to be used as the authentication manager of @media.
+ */
+void
+gst_rtsp_media_set_auth (GstRTSPMedia * media, GstRTSPAuth * auth)
{
- GstStateChangeReturn ret;
- GstElement *udpsrc0, *udpsrc1;
- GstElement *udpsink0, *udpsink1;
- gint tmp_rtp, tmp_rtcp;
- guint count;
- gint rtpport, rtcpport, sockfd;
-
- udpsrc0 = NULL;
- udpsrc1 = NULL;
- udpsink0 = NULL;
- udpsink1 = NULL;
- count = 0;
-
- /* Start with random port */
- tmp_rtp = 0;
-
- /* try to allocate 2 UDP ports, the RTP port should be an even
- * number and the RTCP port should be the next (uneven) port */
-again:
- udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", NULL);
- if (udpsrc0 == NULL)
- goto no_udp_protocol;
- g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, NULL);
-
- ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
- if (ret == GST_STATE_CHANGE_FAILURE) {
- if (tmp_rtp != 0) {
- tmp_rtp += 2;
- if (++count > 20)
- goto no_ports;
-
- gst_element_set_state (udpsrc0, GST_STATE_NULL);
- gst_object_unref (udpsrc0);
-
- goto again;
- }
- goto no_udp_protocol;
- }
-
- g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
+ GstRTSPAuth *old;
- /* check if port is even */
- if ((tmp_rtp & 1) != 0) {
- /* port not even, close and allocate another */
- if (++count > 20)
- goto no_ports;
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
- gst_element_set_state (udpsrc0, GST_STATE_NULL);
- gst_object_unref (udpsrc0);
+ GST_LOG_OBJECT (media, "set auth %p", auth);
- tmp_rtp++;
- goto again;
- }
+ g_mutex_lock (&media->lock);
+ if ((old = media->auth) != auth)
+ media->auth = auth ? g_object_ref (auth) : NULL;
+ else
+ old = NULL;
+ g_mutex_unlock (&media->lock);
- /* allocate port+1 for RTCP now */
- udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", NULL);
- if (udpsrc1 == NULL)
- goto no_udp_rtcp_protocol;
+ if (old)
+ g_object_unref (old);
+}
- /* set port */
- tmp_rtcp = tmp_rtp + 1;
- g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, NULL);
+/**
+ * gst_rtsp_media_get_auth:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the #GstRTSPAuth used as the authentication manager of @media.
+ *
+ * Returns: (transfer full): the #GstRTSPAuth of @media. g_object_unref() after
+ * usage.
+ */
+GstRTSPAuth *
+gst_rtsp_media_get_auth (GstRTSPMedia * media)
+{
+ GstRTSPAuth *result;
- ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
- /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
- if (ret == GST_STATE_CHANGE_FAILURE) {
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
- if (++count > 20)
- goto no_ports;
+ g_mutex_lock (&media->lock);
+ if ((result = media->auth))
+ g_object_ref (result);
+ g_mutex_unlock (&media->lock);
- gst_element_set_state (udpsrc0, GST_STATE_NULL);
- gst_object_unref (udpsrc0);
+ return result;
+}
- gst_element_set_state (udpsrc1, GST_STATE_NULL);
- gst_object_unref (udpsrc1);
+/**
+ * gst_rtsp_media_set_address_pool:
+ * @media: a #GstRTSPMedia
+ * @pool: a #GstRTSPAddressPool
+ *
+ * configure @pool to be used as the address pool of @media.
+ */
+void
+gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
+ GstRTSPAddressPool * pool)
+{
+ GstRTSPAddressPool *old;
- tmp_rtp += 2;
- goto again;
- }
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
- /* all fine, do port check */
- g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
- g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
-
- /* this should not happen... */
- if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
- goto port_error;
-
- udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
- if (!udpsink0)
- goto no_udp_protocol;
-
- g_object_get (G_OBJECT (udpsrc0), "sock", &sockfd, NULL);
- g_object_set (G_OBJECT (udpsink0), "sockfd", sockfd, NULL);
- g_object_set (G_OBJECT (udpsink0), "closefd", FALSE, NULL);
-
- udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
- if (!udpsink1)
- goto no_udp_protocol;
-
- g_object_get (G_OBJECT (udpsrc1), "sock", &sockfd, NULL);
- g_object_set (G_OBJECT (udpsink1), "sockfd", sockfd, NULL);
- g_object_set (G_OBJECT (udpsink1), "closefd", FALSE, NULL);
- g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
- g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
-
- /* we keep these elements, we configure all in configure_transport when the
- * server told us to really use the UDP ports. */
- stream->udpsrc[0] = udpsrc0;
- stream->udpsrc[1] = udpsrc1;
- stream->udpsink[0] = udpsink0;
- stream->udpsink[1] = udpsink1;
- stream->server_port.min = rtpport;
- stream->server_port.max = rtcpport;
+ GST_LOG_OBJECT (media, "set address pool %p", pool);
- return TRUE;
+ g_mutex_lock (&media->lock);
+ if ((old = media->pool) != pool)
+ media->pool = pool ? g_object_ref (pool) : NULL;
+ else
+ old = NULL;
+ g_ptr_array_foreach (media->streams, (GFunc) gst_rtsp_stream_set_address_pool,
+ pool);
+ g_mutex_unlock (&media->lock);
- /* ERRORS */
-no_udp_protocol:
- {
- goto cleanup;
- }
-no_ports:
- {
- goto cleanup;
- }
-no_udp_rtcp_protocol:
- {
- goto cleanup;
- }
-port_error:
- {
- goto cleanup;
- }
-cleanup:
- {
- if (udpsrc0) {
- gst_element_set_state (udpsrc0, GST_STATE_NULL);
- gst_object_unref (udpsrc0);
- }
- if (udpsrc1) {
- gst_element_set_state (udpsrc1, GST_STATE_NULL);
- gst_object_unref (udpsrc1);
- }
- if (udpsink0) {
- gst_element_set_state (udpsink0, GST_STATE_NULL);
- gst_object_unref (udpsink0);
- }
- if (udpsink1) {
- gst_element_set_state (udpsink1, GST_STATE_NULL);
- gst_object_unref (udpsink1);
- }
- return FALSE;
- }
+ if (old)
+ g_object_unref (old);
}
-static void
-caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPMediaStream * stream)
+/**
+ * gst_rtsp_media_get_address_pool:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the #GstRTSPAddressPool used as the address pool of @media.
+ *
+ * Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after
+ * usage.
+ */
+GstRTSPAddressPool *
+gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
{
- gchar *capsstr;
- GstCaps *newcaps, *oldcaps;
+ GstRTSPAddressPool *result;
- if ((newcaps = GST_PAD_CAPS (pad)))
- gst_caps_ref (newcaps);
-
- oldcaps = stream->caps;
- stream->caps = newcaps;
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
- if (oldcaps)
- gst_caps_unref (oldcaps);
+ g_mutex_lock (&media->lock);
+ if ((result = media->pool))
+ g_object_ref (result);
+ g_mutex_unlock (&media->lock);
- capsstr = gst_caps_to_string (newcaps);
- g_message ("stream %p received caps %p, %s", stream, newcaps, capsstr);
- g_free (capsstr);
+ return result;
}
-static void
-dump_structure (const GstStructure *s)
+/**
+ * gst_rtsp_media_collect_streams:
+ * @media: a #GstRTSPMedia
+ *
+ * Find all payloader elements, they should be named pay%d in the
+ * element of @media, and create #GstRTSPStreams for them.
+ *
+ * Collect all dynamic elements, named dynpay%d, and add them to
+ * the list of dynamic elements.
+ */
+void
+gst_rtsp_media_collect_streams (GstRTSPMedia * media)
{
- gchar *sstr;
+ GstElement *element, *elem;
+ GstPad *pad;
+ gint i;
+ gboolean have_elem;
- sstr = gst_structure_to_string (s);
- g_message ("structure: %s", sstr);
- g_free (sstr);
-}
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
-static GstRTSPMediaTrans *
-find_transport (GstRTSPMediaStream *stream, const gchar *rtcp_from)
-{
- GList *walk;
- GstRTSPMediaTrans *result = NULL;
- const gchar *tmp;
- gchar *dest;
- guint port;
+ element = media->element;
- if (rtcp_from == NULL)
- return NULL;
+ have_elem = TRUE;
+ for (i = 0; have_elem; i++) {
+ gchar *name;
- tmp = g_strrstr (rtcp_from, ":");
- if (tmp == NULL)
- return NULL;
+ have_elem = FALSE;
- port = atoi (tmp + 1);
- dest = g_strndup (rtcp_from, tmp - rtcp_from);
+ name = g_strdup_printf ("pay%d", i);
+ if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
+ GST_INFO ("found stream %d with payloader %p", i, elem);
- g_message ("finding %s:%d", dest, port);
+ /* take the pad of the payloader */
+ pad = gst_element_get_static_pad (elem, "src");
+ /* create the stream */
+ gst_rtsp_media_create_stream (media, elem, pad);
+ g_object_unref (pad);
- for (walk = stream->transports; walk; walk = g_list_next (walk)) {
- GstRTSPMediaTrans *trans = walk->data;
- gint min, max;
+ gst_object_unref (elem);
- min = trans->transport->client_port.min;
- max = trans->transport->client_port.max;
+ have_elem = TRUE;
+ }
+ g_free (name);
- if ((strcmp (trans->transport->destination, dest) == 0) && (min == port || max == port)) {
- result = trans;
- break;
+ name = g_strdup_printf ("dynpay%d", i);
+ if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
+ /* a stream that will dynamically create pads to provide RTP packets */
+
+ GST_INFO ("found dynamic element %d, %p", i, elem);
+
+ g_mutex_lock (&media->lock);
+ media->dynamic = g_list_prepend (media->dynamic, elem);
+ g_mutex_unlock (&media->lock);
+
+ have_elem = TRUE;
}
+ g_free (name);
}
- g_free (dest);
-
- return result;
}
-static void
-on_new_ssrc (GObject *session, GObject *source, GstRTSPMediaStream *stream)
+/**
+ * gst_rtsp_media_create_stream:
+ * @media: a #GstRTSPMedia
+ * @payloader: a #GstElement
+ * @srcpad: a source #GstPad
+ *
+ * Create a new stream in @media that provides RTP data on @srcpad.
+ * @srcpad should be a pad of an element inside @media->element.
+ *
+ * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
+ * as @media exists.
+ */
+GstRTSPStream *
+gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
+ GstPad * pad)
{
- GstStructure *stats;
- GstRTSPMediaTrans *trans;
+ GstRTSPStream *stream;
+ GstPad *srcpad;
+ gchar *name;
+ gint idx;
- g_message ("%p: new source %p", stream, source);
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+ g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
+ g_return_val_if_fail (GST_IS_PAD (pad), NULL);
+ g_return_val_if_fail (GST_PAD_IS_SRC (pad), NULL);
- /* see if we have a stream to match with the origin of the RTCP packet */
- trans = g_object_get_qdata (source, ssrc_stream_map_key);
- if (trans == NULL) {
- g_object_get (source, "stats", &stats, NULL);
- if (stats) {
- const gchar *rtcp_from;
+ g_mutex_lock (&media->lock);
+ idx = media->streams->len;
- dump_structure (stats);
+ name = g_strdup_printf ("src_%u", idx);
+ srcpad = gst_ghost_pad_new (name, pad);
+ gst_pad_set_active (srcpad, TRUE);
+ gst_element_add_pad (media->element, srcpad);
+ g_free (name);
- rtcp_from = gst_structure_get_string (stats, "rtcp-from");
- if ((trans = find_transport (stream, rtcp_from))) {
- g_message ("%p: found transport %p for source %p", stream, trans, source);
+ stream = gst_rtsp_stream_new (idx, payloader, srcpad);
+ if (media->pool)
+ gst_rtsp_stream_set_address_pool (stream, media->pool);
- /* keep ref to the source */
- trans->rtpsource = source;
+ g_ptr_array_add (media->streams, stream);
+ g_mutex_unlock (&media->lock);
- g_object_set_qdata (source, ssrc_stream_map_key, trans);
- }
- gst_structure_free (stats);
- }
- } else {
- g_message ("%p: source %p for transport %p", stream, source, trans);
- }
-}
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
+ NULL);
-static void
-on_ssrc_sdes (GObject *session, GObject *source, GstRTSPMediaStream *stream)
-{
- g_message ("%p: new SDES %p", stream, source);
+ return stream;
}
-static void
-on_ssrc_active (GObject *session, GObject *source, GstRTSPMediaStream *stream)
+/**
+ * gst_rtsp_media_n_streams:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the number of streams in this media.
+ *
+ * Returns: The number of streams.
+ */
+guint
+gst_rtsp_media_n_streams (GstRTSPMedia * media)
{
- GstRTSPMediaTrans *trans;
+ guint res;
- trans = g_object_get_qdata (source, ssrc_stream_map_key);
-
- g_message ("%p: source %p in transport %p is active", stream, source, trans);
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
- if (trans && trans->keep_alive)
- trans->keep_alive (trans->ka_user_data);
+ g_mutex_lock (&media->lock);
+ res = media->streams->len;
+ g_mutex_unlock (&media->lock);
-#ifdef DUMP_STATS
- {
- GstStructure *stats;
- g_object_get (source, "stats", &stats, NULL);
- if (stats) {
- dump_structure (stats);
- gst_structure_free (stats);
- }
- }
-#endif
+ return res;
}
-static void
-on_bye_ssrc (GObject *session, GObject *source, GstRTSPMediaStream *stream)
+/**
+ * gst_rtsp_media_get_stream:
+ * @media: a #GstRTSPMedia
+ * @idx: the stream index
+ *
+ * Retrieve the stream with index @idx from @media.
+ *
+ * Returns: (transfer none): the #GstRTSPStream at index @idx or %NULL when a stream with
+ * that index did not exist.
+ */
+GstRTSPStream *
+gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
{
- g_message ("%p: source %p bye", stream, source);
-}
+ GstRTSPStream *res;
-static void
-on_bye_timeout (GObject *session, GObject *source, GstRTSPMediaStream *stream)
-{
- GstRTSPMediaTrans *trans;
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
- g_message ("%p: source %p bye timeout", stream, source);
+ g_mutex_lock (&media->lock);
+ if (idx < media->streams->len)
+ res = g_ptr_array_index (media->streams, idx);
+ else
+ res = NULL;
+ g_mutex_unlock (&media->lock);
- if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
- trans->rtpsource = NULL;
- trans->timeout = TRUE;
- }
+ return res;
}
-static void
-on_timeout (GObject *session, GObject *source, GstRTSPMediaStream *stream)
+/**
+ * gst_rtsp_media_get_range_string:
+ * @media: a #GstRTSPMedia
+ * @play: for the PLAY request
+ *
+ * Get the current range as a string.
+ *
+ * Returns: The range as a string, g_free() after usage.
+ */
+gchar *
+gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play)
{
- GstRTSPMediaTrans *trans;
+ gchar *result;
+ GstRTSPTimeRange range;
- g_message ("%p: source %p timeout", stream, source);
+ g_mutex_lock (&media->lock);
+ /* make copy */
+ range = media->range;
- if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
- trans->rtpsource = NULL;
- trans->timeout = TRUE;
+ if (!play && media->n_active > 0) {
+ range.min.type = GST_RTSP_TIME_NOW;
+ range.min.seconds = -1;
}
+ g_mutex_unlock (&media->lock);
+
+ result = gst_rtsp_range_to_string (&range);
+
+ return result;
}
-static GstFlowReturn
-handle_new_buffer (GstAppSink *sink, gpointer user_data)
+/**
+ * gst_rtsp_media_seek:
+ * @media: a #GstRTSPMedia
+ * @range: a #GstRTSPTimeRange
+ *
+ * Seek the pipeline to @range.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
{
- GList *walk;
- GstBuffer *buffer;
- GstRTSPMediaStream *stream;
+ GstSeekFlags flags;
+ gboolean res;
+ gint64 start, stop;
+ GstSeekType start_type, stop_type;
- buffer = gst_app_sink_pull_buffer (sink);
- if (!buffer)
- return GST_FLOW_OK;
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+ g_return_val_if_fail (range != NULL, FALSE);
- stream = (GstRTSPMediaStream *) user_data;
+ g_rec_mutex_lock (&media->state_lock);
+ if (!media->seekable)
+ goto not_seekable;
- for (walk = stream->transports; walk; walk = g_list_next (walk)) {
- GstRTSPMediaTrans *tr = (GstRTSPMediaTrans *) walk->data;
+ if (range->unit != GST_RTSP_RANGE_NPT)
+ goto not_supported;
- if (GST_ELEMENT_CAST (sink) == stream->appsink[0]) {
- if (tr->send_rtp)
- tr->send_rtp (buffer, tr->transport->interleaved.min, tr->user_data);
- }
- else {
- if (tr->send_rtcp)
- tr->send_rtcp (buffer, tr->transport->interleaved.max, tr->user_data);
- }
- }
- gst_buffer_unref (buffer);
+ /* depends on the current playing state of the pipeline. We might need to
+ * queue this until we get EOS. */
+ flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_KEY_UNIT;
- return GST_FLOW_OK;
-}
+ start_type = stop_type = GST_SEEK_TYPE_NONE;
-static GstAppSinkCallbacks sink_cb = {
- NULL, /* not interested in EOS */
- NULL, /* not interested in preroll buffers */
- handle_new_buffer
-};
+ switch (range->min.type) {
+ case GST_RTSP_TIME_NOW:
+ start = -1;
+ break;
+ case GST_RTSP_TIME_SECONDS:
+ /* only seek when something changed */
+ if (media->range.min.seconds == range->min.seconds) {
+ start = -1;
+ } else {
+ start = range->min.seconds * GST_SECOND;
+ start_type = GST_SEEK_TYPE_SET;
+ }
+ break;
+ case GST_RTSP_TIME_END:
+ default:
+ goto weird_type;
+ }
+ switch (range->max.type) {
+ case GST_RTSP_TIME_SECONDS:
+ /* only seek when something changed */
+ if (media->range.max.seconds == range->max.seconds) {
+ stop = -1;
+ } else {
+ stop = range->max.seconds * GST_SECOND;
+ stop_type = GST_SEEK_TYPE_SET;
+ }
+ break;
+ case GST_RTSP_TIME_END:
+ stop = -1;
+ stop_type = GST_SEEK_TYPE_SET;
+ break;
+ case GST_RTSP_TIME_NOW:
+ default:
+ goto weird_type;
+ }
-/* prepare the pipeline objects to handle @stream in @media */
-static gboolean
-setup_stream (GstRTSPMediaStream *stream, guint idx, GstRTSPMedia *media)
-{
- gchar *name;
- GstPad *pad, *teepad, *selpad;
- GstPadLinkReturn ret;
- gint i;
+ if (start != -1 || stop != -1) {
+ GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
- /* allocate udp ports, we will have 4 of them, 2 for receiving RTP/RTCP and 2
- * for sending RTP/RTCP. The sender and receiver ports are shared between the
- * elements */
- if (!alloc_udp_ports (stream))
- return FALSE;
+ res = gst_element_seek (media->pipeline, 1.0, GST_FORMAT_TIME,
+ flags, start_type, start, stop_type, stop);
- /* add the ports to the pipeline */
- for (i = 0; i < 2; i++) {
- gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsink[i]);
- gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsrc[i]);
- }
+ /* and block for the seek to complete */
+ GST_INFO ("done seeking %d", res);
+ gst_element_get_state (media->pipeline, NULL, NULL, -1);
+ GST_INFO ("prerolled again");
- /* create elements for the TCP transfer */
- for (i = 0; i < 2; i++) {
- stream->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
- stream->appsink[i] = gst_element_factory_make ("appsink", NULL);
- g_object_set (stream->appsink[i], "async", FALSE, "sync", FALSE, NULL);
- g_object_set (stream->appsink[i], "emit-signals", FALSE, NULL);
- g_object_set (stream->appsink[i], "preroll-queue-len", 1, NULL);
- gst_bin_add (GST_BIN_CAST (media->pipeline), stream->appsink[i]);
- gst_bin_add (GST_BIN_CAST (media->pipeline), stream->appsrc[i]);
- gst_app_sink_set_callbacks (GST_APP_SINK_CAST (stream->appsink[i]),
- &sink_cb, stream, NULL);
+ collect_media_stats (media);
+ } else {
+ GST_INFO ("no seek needed");
+ res = TRUE;
}
+ g_rec_mutex_unlock (&media->state_lock);
- /* hook up the stream to the RTP session elements. */
- name = g_strdup_printf ("send_rtp_sink_%d", idx);
- stream->send_rtp_sink = gst_element_get_request_pad (media->rtpbin, name);
- g_free (name);
- name = g_strdup_printf ("send_rtp_src_%d", idx);
- stream->send_rtp_src = gst_element_get_static_pad (media->rtpbin, name);
- g_free (name);
- name = g_strdup_printf ("send_rtcp_src_%d", idx);
- stream->send_rtcp_src = gst_element_get_request_pad (media->rtpbin, name);
- g_free (name);
- name = g_strdup_printf ("recv_rtcp_sink_%d", idx);
- stream->recv_rtcp_sink = gst_element_get_request_pad (media->rtpbin, name);
- g_free (name);
- name = g_strdup_printf ("recv_rtp_sink_%d", idx);
- stream->recv_rtp_sink = gst_element_get_request_pad (media->rtpbin, name);
- g_free (name);
-
- /* get the session */
- g_signal_emit_by_name (media->rtpbin, "get-internal-session", idx,
- &stream->session);
-
- g_signal_connect (stream->session, "on-new-ssrc", (GCallback) on_new_ssrc,
- stream);
- g_signal_connect (stream->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
- stream);
- g_signal_connect (stream->session, "on-ssrc-active", (GCallback) on_ssrc_active,
- stream);
- g_signal_connect (stream->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
- stream);
- g_signal_connect (stream->session, "on-bye-timeout", (GCallback) on_bye_timeout,
- stream);
- g_signal_connect (stream->session, "on-timeout", (GCallback) on_timeout,
- stream);
-
- /* link the RTP pad to the session manager */
- ret = gst_pad_link (stream->srcpad, stream->send_rtp_sink);
- if (ret != GST_PAD_LINK_OK)
- goto link_failed;
-
- /* make tee for RTP and link to stream */
- stream->tee[0] = gst_element_factory_make ("tee", NULL);
- gst_bin_add (GST_BIN_CAST (media->pipeline), stream->tee[0]);
-
- pad = gst_element_get_static_pad (stream->tee[0], "sink");
- gst_pad_link (stream->send_rtp_src, pad);
- gst_object_unref (pad);
-
- /* link RTP sink, we're pretty sure this will work. */
- teepad = gst_element_get_request_pad (stream->tee[0], "src%d");
- pad = gst_element_get_static_pad (stream->udpsink[0], "sink");
- gst_pad_link (teepad, pad);
- gst_object_unref (pad);
- gst_object_unref (teepad);
-
- teepad = gst_element_get_request_pad (stream->tee[0], "src%d");
- pad = gst_element_get_static_pad (stream->appsink[0], "sink");
- gst_pad_link (teepad, pad);
- gst_object_unref (pad);
- gst_object_unref (teepad);
-
- /* make tee for RTCP */
- stream->tee[1] = gst_element_factory_make ("tee", NULL);
- gst_bin_add (GST_BIN_CAST (media->pipeline), stream->tee[1]);
-
- pad = gst_element_get_static_pad (stream->tee[1], "sink");
- gst_pad_link (stream->send_rtcp_src, pad);
- gst_object_unref (pad);
-
- /* link RTCP elements */
- teepad = gst_element_get_request_pad (stream->tee[1], "src%d");
- pad = gst_element_get_static_pad (stream->udpsink[1], "sink");
- gst_pad_link (teepad, pad);
- gst_object_unref (pad);
- gst_object_unref (teepad);
-
- teepad = gst_element_get_request_pad (stream->tee[1], "src%d");
- pad = gst_element_get_static_pad (stream->appsink[1], "sink");
- gst_pad_link (teepad, pad);
- gst_object_unref (pad);
- gst_object_unref (teepad);
-
- /* make selector for the RTP receivers */
- stream->selector[0] = gst_element_factory_make ("input-selector", NULL);
- g_object_set (stream->selector[0], "select-all", TRUE, NULL);
- gst_bin_add (GST_BIN_CAST (media->pipeline), stream->selector[0]);
-
- pad = gst_element_get_static_pad (stream->selector[0], "src");
- gst_pad_link (pad, stream->recv_rtp_sink);
- gst_object_unref (pad);
-
- selpad = gst_element_get_request_pad (stream->selector[0], "sink%d");
- pad = gst_element_get_static_pad (stream->udpsrc[0], "src");
- gst_pad_link (pad, selpad);
- gst_object_unref (pad);
- gst_object_unref (selpad);
-
- selpad = gst_element_get_request_pad (stream->selector[0], "sink%d");
- pad = gst_element_get_static_pad (stream->appsrc[0], "src");
- gst_pad_link (pad, selpad);
- gst_object_unref (pad);
- gst_object_unref (selpad);
-
- /* make selector for the RTCP receivers */
- stream->selector[1] = gst_element_factory_make ("input-selector", NULL);
- g_object_set (stream->selector[1], "select-all", TRUE, NULL);
- gst_bin_add (GST_BIN_CAST (media->pipeline), stream->selector[1]);
-
- pad = gst_element_get_static_pad (stream->selector[1], "src");
- gst_pad_link (pad, stream->recv_rtcp_sink);
- gst_object_unref (pad);
-
- selpad = gst_element_get_request_pad (stream->selector[1], "sink%d");
- pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
- gst_pad_link (pad, selpad);
- gst_object_unref (pad);
- gst_object_unref (selpad);
-
- selpad = gst_element_get_request_pad (stream->selector[1], "sink%d");
- pad = gst_element_get_static_pad (stream->appsrc[1], "src");
- gst_pad_link (pad, selpad);
- gst_object_unref (pad);
- gst_object_unref (selpad);
-
- /* we set and keep these to playing so that they don't cause NO_PREROLL return
- * values */
- gst_element_set_state (stream->udpsrc[0], GST_STATE_PLAYING);
- gst_element_set_state (stream->udpsrc[1], GST_STATE_PLAYING);
- gst_element_set_locked_state (stream->udpsrc[0], TRUE);
- gst_element_set_locked_state (stream->udpsrc[1], TRUE);
-
- /* be notified of caps changes */
- stream->caps_sig = g_signal_connect (stream->send_rtp_sink, "notify::caps",
- (GCallback) caps_notify, stream);
-
- stream->prepared = TRUE;
-
- return TRUE;
+ return res;
/* ERRORS */
-link_failed:
+not_seekable:
+ {
+ g_rec_mutex_unlock (&media->state_lock);
+ GST_INFO ("pipeline is not seekable");
+ return TRUE;
+ }
+not_supported:
+ {
+ g_rec_mutex_unlock (&media->state_lock);
+ GST_WARNING ("seek unit %d not supported", range->unit);
+ return FALSE;
+ }
+weird_type:
{
- g_warning ("failed to link stream %d", idx);
+ g_rec_mutex_unlock (&media->state_lock);
+ GST_WARNING ("weird range type %d not supported", range->min.type);
return FALSE;
}
}
static void
-unlock_streams (GstRTSPMedia *media)
+gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
{
- guint i, n_streams;
-
- /* unlock the udp src elements */
- n_streams = gst_rtsp_media_n_streams (media);
- for (i = 0; i < n_streams; i++) {
- GstRTSPMediaStream *stream;
-
- stream = gst_rtsp_media_get_stream (media, i);
+ g_mutex_lock (&media->lock);
+ /* never overwrite the error status */
+ if (media->status != GST_RTSP_MEDIA_STATUS_ERROR)
+ media->status = status;
+ GST_DEBUG ("setting new status to %d", status);
+ g_cond_broadcast (&media->cond);
+ g_mutex_unlock (&media->lock);
+}
- gst_element_set_locked_state (stream->udpsrc[0], FALSE);
- gst_element_set_locked_state (stream->udpsrc[1], FALSE);
+static GstRTSPMediaStatus
+gst_rtsp_media_get_status (GstRTSPMedia * media)
+{
+ GstRTSPMediaStatus result;
+ gint64 end_time;
+
+ g_mutex_lock (&media->lock);
+ end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
+ /* while we are preparing, wait */
+ while (media->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
+ GST_DEBUG ("waiting for status change");
+ if (!g_cond_wait_until (&media->cond, &media->lock, end_time)) {
+ GST_DEBUG ("timeout, assuming error status");
+ media->status = GST_RTSP_MEDIA_STATUS_ERROR;
+ }
}
+ /* could be success or error */
+ result = media->status;
+ GST_DEBUG ("got status %d", result);
+ g_mutex_unlock (&media->lock);
+
+ return result;
}
+/* called with state-lock */
static gboolean
-default_handle_message (GstRTSPMedia *media, GstMessage *message)
+default_handle_message (GstRTSPMedia * media, GstMessage * message)
{
GstMessageType type;
media->buffering = FALSE;
/* if the desired state is playing, go back */
if (media->target_state == GST_STATE_PLAYING) {
- g_message ("Buffering done, setting pipeline to PLAYING");
+ GST_INFO ("Buffering done, setting pipeline to PLAYING");
gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
+ } else {
+ GST_INFO ("Buffering done");
}
- else {
- g_message ("Buffering done");
- }
} else {
/* buffering busy */
if (media->buffering == FALSE) {
- if (media->target_state == GST_STATE_PLAYING) {
+ if (media->target_state == GST_STATE_PLAYING) {
/* we were not buffering but PLAYING, PAUSE the pipeline. */
- g_message ("Buffering, setting pipeline to PAUSED ...");
+ GST_INFO ("Buffering, setting pipeline to PAUSED ...");
gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
- }
- else {
- g_message ("Buffering ...");
- }
+ } else {
+ GST_INFO ("Buffering ...");
+ }
}
media->buffering = TRUE;
}
gchar *debug;
gst_message_parse_error (message, &gerror, &debug);
- g_warning ("%p: got error %s (%s)", media, gerror->message, debug);
+ GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
g_error_free (gerror);
g_free (debug);
+
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
break;
}
case GST_MESSAGE_WARNING:
gchar *debug;
gst_message_parse_warning (message, &gerror, &debug);
- g_warning ("%p: got warning %s (%s)", media, gerror->message, debug);
+ GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
g_error_free (gerror);
g_free (debug);
break;
break;
case GST_MESSAGE_STREAM_STATUS:
break;
+ case GST_MESSAGE_ASYNC_DONE:
+ if (!media->adding) {
+ /* when we are dynamically adding pads, the addition of the udpsrc will
+ * temporarily produce ASYNC_DONE messages. We have to ignore them and
+ * wait for the final ASYNC_DONE after everything prerolled */
+ GST_INFO ("%p: got ASYNC_DONE", media);
+ collect_media_stats (media);
+
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+ } else {
+ GST_INFO ("%p: ignoring ASYNC_DONE", media);
+ }
+ break;
+ case GST_MESSAGE_EOS:
+ GST_INFO ("%p: got EOS", media);
+
+ if (media->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
+ GST_DEBUG ("shutting down after EOS");
+ finish_unprepare (media);
+ g_object_unref (media);
+ }
+ break;
default:
- g_message ("%p: got message type %s", media, gst_message_type_get_name (type));
+ GST_INFO ("%p: got message type %s", media,
+ gst_message_type_get_name (type));
break;
}
return TRUE;
}
static gboolean
-bus_message (GstBus *bus, GstMessage *message, GstRTSPMedia *media)
+bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
{
GstRTSPMediaClass *klass;
gboolean ret;
-
+
klass = GST_RTSP_MEDIA_GET_CLASS (media);
+ g_rec_mutex_lock (&media->state_lock);
if (klass->handle_message)
ret = klass->handle_message (media, message);
else
ret = FALSE;
+ g_rec_mutex_unlock (&media->state_lock);
return ret;
}
static void
-pad_added_cb (GstElement *element, GstPad *pad, GstRTSPMedia *media)
+watch_destroyed (GstRTSPMedia * media)
{
- GstRTSPMediaStream *stream;
- gchar *name;
- gint i;
-
- i = media->streams->len + 1;
-
- g_message ("pad added %s:%s, stream %d", GST_DEBUG_PAD_NAME (pad), i);
+ GST_DEBUG_OBJECT (media, "source destroyed");
+ gst_object_unref (media);
+}
- stream = g_new0 (GstRTSPMediaStream, 1);
- stream->payloader = element;
+/* called from streaming threads */
+static void
+pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
+{
+ GstRTSPStream *stream;
- name = g_strdup_printf ("dynpay%d", i);
+ /* FIXME, element is likely not a payloader, find the payloader here */
+ stream = gst_rtsp_media_create_stream (media, element, pad);
- /* ghost the pad of the payloader to the element */
- stream->srcpad = gst_ghost_pad_new (name, pad);
- gst_pad_set_active (stream->srcpad, TRUE);
- gst_element_add_pad (media->element, stream->srcpad);
- g_free (name);
+ GST_INFO ("pad added %s:%s, stream %d", GST_DEBUG_PAD_NAME (pad),
+ stream->idx);
- /* add stream now */
- g_array_append_val (media->streams, stream);
+ g_rec_mutex_lock (&media->state_lock);
+ /* we will be adding elements below that will cause ASYNC_DONE to be
+ * posted in the bus. We want to ignore those messages until the
+ * pipeline really prerolled. */
+ media->adding = TRUE;
- setup_stream (stream, i, media);
+ /* join the element in the PAUSED state because this callback is
+ * called from the streaming thread and it is PAUSED */
+ gst_rtsp_stream_join_bin (stream, GST_BIN (media->pipeline),
+ media->rtpbin, GST_STATE_PAUSED);
- for (i = 0; i < 2; i++) {
- gst_element_set_state (stream->udpsink[i], GST_STATE_PAUSED);
- gst_element_set_state (stream->appsink[i], GST_STATE_PAUSED);
- gst_element_set_state (stream->tee[i], GST_STATE_PAUSED);
- gst_element_set_state (stream->selector[i], GST_STATE_PAUSED);
- gst_element_set_state (stream->appsrc[i], GST_STATE_PAUSED);
- }
+ media->adding = FALSE;
+ g_rec_mutex_unlock (&media->state_lock);
}
static void
-no_more_pads_cb (GstElement *element, GstRTSPMedia *media)
+no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
{
- g_message ("no more pads");
- if (media->fakesink) {
- gst_object_ref (media->fakesink);
- gst_bin_remove (GST_BIN (media->pipeline), media->fakesink);
- gst_element_set_state (media->fakesink, GST_STATE_NULL);
- gst_object_unref (media->fakesink);
+ GstElement *fakesink;
+
+ g_mutex_lock (&media->lock);
+ GST_INFO ("no more pads");
+ if ((fakesink = media->fakesink)) {
+ gst_object_ref (fakesink);
media->fakesink = NULL;
- g_message ("removed fakesink");
+ g_mutex_unlock (&media->lock);
+
+ gst_bin_remove (GST_BIN (media->pipeline), fakesink);
+ gst_element_set_state (fakesink, GST_STATE_NULL);
+ gst_object_unref (fakesink);
+ GST_INFO ("removed fakesink");
}
}
/**
* gst_rtsp_media_prepare:
- * @obj: a #GstRTSPMedia
+ * @media: a #GstRTSPMedia
*
* Prepare @media for streaming. This function will create the pipeline and
* other objects to manage the streaming.
* Returns: %TRUE on success.
*/
gboolean
-gst_rtsp_media_prepare (GstRTSPMedia *media)
+gst_rtsp_media_prepare (GstRTSPMedia * media)
{
GstStateChangeReturn ret;
- guint i, n_streams;
+ GstRTSPMediaStatus status;
+ guint i;
GstRTSPMediaClass *klass;
GstBus *bus;
GList *walk;
- if (media->prepared)
+ g_rec_mutex_lock (&media->state_lock);
+ if (media->status == GST_RTSP_MEDIA_STATUS_PREPARED)
goto was_prepared;
+ if (media->status == GST_RTSP_MEDIA_STATUS_PREPARING)
+ goto wait_status;
+
+ if (media->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
+ goto not_unprepared;
+
if (!media->reusable && media->reused)
goto is_reused;
- g_message ("preparing media %p", media);
+ media->rtpbin = gst_element_factory_make ("rtpbin", NULL);
+ if (media->rtpbin == NULL)
+ goto no_rtpbin;
+
+ GST_INFO ("preparing media %p", media);
+
+ /* reset some variables */
+ media->is_live = FALSE;
+ media->seekable = FALSE;
+ media->buffering = FALSE;
+ /* we're preparing now */
+ media->status = GST_RTSP_MEDIA_STATUS_PREPARING;
bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (media->pipeline));
media->source = gst_bus_create_watch (bus);
gst_object_unref (bus);
- g_source_set_callback (media->source, (GSourceFunc) bus_message, media, NULL);
+ g_source_set_callback (media->source, (GSourceFunc) bus_message,
+ gst_object_ref (media), (GDestroyNotify) watch_destroyed);
klass = GST_RTSP_MEDIA_GET_CLASS (media);
media->id = g_source_attach (media->source, klass->context);
- media->rtpbin = gst_element_factory_make ("gstrtpbin", NULL);
-
/* add stuff to the bin */
gst_bin_add (GST_BIN (media->pipeline), media->rtpbin);
/* link streams we already have, other streams might appear when we have
* dynamic elements */
- n_streams = gst_rtsp_media_n_streams (media);
- for (i = 0; i < n_streams; i++) {
- GstRTSPMediaStream *stream;
+ for (i = 0; i < media->streams->len; i++) {
+ GstRTSPStream *stream;
- stream = gst_rtsp_media_get_stream (media, i);
+ stream = g_ptr_array_index (media->streams, i);
- setup_stream (stream, i, media);
+ gst_rtsp_stream_join_bin (stream, GST_BIN (media->pipeline),
+ media->rtpbin, GST_STATE_NULL);
}
for (walk = media->dynamic; walk; walk = g_list_next (walk)) {
GstElement *elem = walk->data;
+ GST_INFO ("adding callbacks for dynamic element %p", elem);
+
g_signal_connect (elem, "pad-added", (GCallback) pad_added_cb, media);
g_signal_connect (elem, "no-more-pads", (GCallback) no_more_pads_cb, media);
+ /* we add a fakesink here in order to make the state change async. We remove
+ * the fakesink again in the no-more-pads callback. */
media->fakesink = gst_element_factory_make ("fakesink", "fakesink");
gst_bin_add (GST_BIN (media->pipeline), media->fakesink);
}
+ GST_INFO ("setting pipeline to PAUSED for media %p", media);
/* first go to PAUSED */
ret = gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
media->target_state = GST_STATE_PAUSED;
switch (ret) {
case GST_STATE_CHANGE_SUCCESS:
+ GST_INFO ("SUCCESS state change for media %p", media);
+ media->seekable = TRUE;
break;
case GST_STATE_CHANGE_ASYNC:
+ GST_INFO ("ASYNC state change for media %p", media);
+ media->seekable = TRUE;
break;
case GST_STATE_CHANGE_NO_PREROLL:
/* we need to go to PLAYING */
- g_message ("live media %p", media);
+ GST_INFO ("NO_PREROLL state change: live media %p", media);
+ /* FIXME we disable seeking for live streams for now. We should perform a
+ * seeking query in preroll instead */
+ media->seekable = FALSE;
media->is_live = TRUE;
ret = gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE)
case GST_STATE_CHANGE_FAILURE:
goto state_failed;
}
+wait_status:
+ g_rec_mutex_unlock (&media->state_lock);
- /* now wait for all pads to be prerolled */
- ret = gst_element_get_state (media->pipeline, NULL, NULL, -1);
- if (ret == GST_STATE_CHANGE_FAILURE)
+ /* now wait for all pads to be prerolled, FIXME, we should somehow be
+ * able to do this async so that we don't block the server thread. */
+ status = gst_rtsp_media_get_status (media);
+ if (status == GST_RTSP_MEDIA_STATUS_ERROR)
goto state_failed;
- /* collect stats about the media */
- collect_media_stats (media);
-
- g_message ("object %p is prerolled", media);
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
- media->prepared = TRUE;
+ GST_INFO ("object %p is prerolled", media);
return TRUE;
/* OK */
was_prepared:
{
+ GST_LOG ("media %p was prepared", media);
+ g_rec_mutex_unlock (&media->state_lock);
return TRUE;
}
/* ERRORS */
-state_failed:
+not_unprepared:
{
- g_warning ("failed to preroll pipeline");
- unlock_streams (media);
- gst_element_set_state (media->pipeline, GST_STATE_NULL);
+ GST_WARNING ("media %p was not unprepared", media);
+ g_rec_mutex_unlock (&media->state_lock);
return FALSE;
}
is_reused:
{
- g_warning ("can not reuse media %p", media);
+ g_rec_mutex_unlock (&media->state_lock);
+ GST_WARNING ("can not reuse media %p", media);
+ return FALSE;
+ }
+no_rtpbin:
+ {
+ g_rec_mutex_unlock (&media->state_lock);
+ GST_WARNING ("no rtpbin element");
+ g_warning ("failed to create element 'rtpbin', check your installation");
+ return FALSE;
+ }
+state_failed:
+ {
+ GST_WARNING ("failed to preroll pipeline");
+ gst_rtsp_media_unprepare (media);
+ g_rec_mutex_unlock (&media->state_lock);
return FALSE;
}
}
+/* must be called with state-lock */
+static void
+finish_unprepare (GstRTSPMedia * media)
+{
+ gint i;
+
+ GST_DEBUG ("shutting down");
+
+ gst_element_set_state (media->pipeline, GST_STATE_NULL);
+
+ for (i = 0; i < media->streams->len; i++) {
+ GstRTSPStream *stream;
+
+ GST_INFO ("Removing elements of stream %d from pipeline", i);
+
+ stream = g_ptr_array_index (media->streams, i);
+
+ gst_rtsp_stream_leave_bin (stream, GST_BIN (media->pipeline),
+ media->rtpbin);
+ }
+ g_ptr_array_set_size (media->streams, 0);
+
+ gst_bin_remove (GST_BIN (media->pipeline), media->rtpbin);
+ media->rtpbin = NULL;
+
+ gst_object_unref (media->pipeline);
+ media->pipeline = NULL;
+
+ media->reused = TRUE;
+ media->status = GST_RTSP_MEDIA_STATUS_UNPREPARED;
+
+ /* when the media is not reusable, this will effectively unref the media and
+ * recreate it */
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
+}
+
+/* called with state-lock */
+static gboolean
+default_unprepare (GstRTSPMedia * media)
+{
+ if (media->eos_shutdown) {
+ GST_DEBUG ("sending EOS for shutdown");
+ /* ref so that we don't disappear */
+ g_object_ref (media);
+ gst_element_send_event (media->pipeline, gst_event_new_eos ());
+ /* we need to go to playing again for the EOS to propagate, normally in this
+ * state, nothing is receiving data from us anymore so this is ok. */
+ gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
+ media->status = GST_RTSP_MEDIA_STATUS_UNPREPARING;
+ } else {
+ finish_unprepare (media);
+ }
+ return TRUE;
+}
+
/**
* gst_rtsp_media_unprepare:
- * @obj: a #GstRTSPMedia
+ * @media: a #GstRTSPMedia
*
* Unprepare @media. After this call, the media should be prepared again before
* it can be used again. If the media is set to be non-reusable, a new instance
* Returns: %TRUE on success.
*/
gboolean
-gst_rtsp_media_unprepare (GstRTSPMedia *media)
+gst_rtsp_media_unprepare (GstRTSPMedia * media)
{
- GstRTSPMediaClass *klass;
gboolean success;
- if (!media->prepared)
- return TRUE;
+ g_rec_mutex_lock (&media->state_lock);
+ if (media->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
+ goto was_unprepared;
- g_message ("unprepare media %p", media);
+ GST_INFO ("unprepare media %p", media);
media->target_state = GST_STATE_NULL;
+ success = TRUE;
- klass = GST_RTSP_MEDIA_GET_CLASS (media);
- if (klass->unprepare)
- success = klass->unprepare (media);
- else
- success = TRUE;
+ if (media->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
+ GstRTSPMediaClass *klass;
- media->prepared = FALSE;
- media->reused = TRUE;
-
- /* when the media is not reusable, this will effectively unref the media and
- * recreate it */
- g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+ if (klass->unprepare)
+ success = klass->unprepare (media);
+ } else {
+ finish_unprepare (media);
+ }
+ if (media->source) {
+ g_source_destroy (media->source);
+ g_source_unref (media->source);
+ media->source = NULL;
+ }
+ g_rec_mutex_unlock (&media->state_lock);
return success;
-}
-static gboolean
-default_unprepare (GstRTSPMedia *media) {
- gst_element_set_state (media->pipeline, GST_STATE_NULL);
- return TRUE;
+was_unprepared:
+ {
+ g_rec_mutex_unlock (&media->state_lock);
+ GST_INFO ("media %p was already unprepared", media);
+ return TRUE;
+ }
}
/**
* gst_rtsp_media_set_state:
* @media: a #GstRTSPMedia
* @state: the target state of the media
- * @transports: a GArray of #GstRTSPMediaTrans pointers
+ * @transports: a #GPtrArray of #GstRTSPStreamTransport pointers
*
* Set the state of @media to @state and for the transports in @transports.
*
* Returns: %TRUE on success.
*/
gboolean
-gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state, GArray *transports)
+gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
+ GPtrArray * transports)
{
gint i;
- GstStateChangeReturn ret;
gboolean add, remove, do_state;
gint old_active;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
g_return_val_if_fail (transports != NULL, FALSE);
+ g_rec_mutex_lock (&media->state_lock);
+
/* NULL and READY are the same */
if (state == GST_STATE_READY)
state = GST_STATE_NULL;
add = remove = FALSE;
- g_message ("going to state %s media %p", gst_element_state_get_name (state), media);
+ GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
+ media);
switch (state) {
case GST_STATE_NULL:
- /* unlock the streams so that they follow the state changes from now on */
- unlock_streams (media);
- /* fallthrough */
case GST_STATE_PAUSED:
/* we're going from PLAYING to PAUSED, READY or NULL, remove */
if (media->target_state == GST_STATE_PLAYING)
- remove = TRUE;
+ remove = TRUE;
break;
case GST_STATE_PLAYING:
/* we're going to PLAYING, add */
default:
break;
}
- old_active = media->active;
+ old_active = media->n_active;
for (i = 0; i < transports->len; i++) {
- GstRTSPMediaTrans *tr;
- GstRTSPMediaStream *stream;
- GstRTSPTransport *trans;
+ GstRTSPStreamTransport *trans;
/* we need a non-NULL entry in the array */
- tr = g_array_index (transports, GstRTSPMediaTrans *, i);
- if (tr == NULL)
+ trans = g_ptr_array_index (transports, i);
+ if (trans == NULL)
continue;
/* we need a transport */
- if (!(trans = tr->transport))
+ if (!trans->transport)
continue;
- /* get the stream and add the destinations */
- stream = gst_rtsp_media_get_stream (media, tr->idx);
- switch (trans->lower_transport) {
- case GST_RTSP_LOWER_TRANS_UDP:
- case GST_RTSP_LOWER_TRANS_UDP_MCAST:
- {
- gchar *dest;
- gint min, max;
-
- dest = trans->destination;
- min = trans->client_port.min;
- max = trans->client_port.max;
-
- if (add && !tr->active) {
- g_message ("adding %s:%d-%d", dest, min, max);
- g_signal_emit_by_name (stream->udpsink[0], "add", dest, min, NULL);
- g_signal_emit_by_name (stream->udpsink[1], "add", dest, max, NULL);
- stream->transports = g_list_prepend (stream->transports, tr);
- tr->active = TRUE;
- media->active++;
- } else if (remove && tr->active) {
- g_message ("removing %s:%d-%d", dest, min, max);
- g_signal_emit_by_name (stream->udpsink[0], "remove", dest, min, NULL);
- g_signal_emit_by_name (stream->udpsink[1], "remove", dest, max, NULL);
- stream->transports = g_list_remove (stream->transports, tr);
- tr->active = FALSE;
- media->active--;
- }
- break;
- }
- case GST_RTSP_LOWER_TRANS_TCP:
- if (add && !tr->active) {
- g_message ("adding TCP %s", trans->destination);
- stream->transports = g_list_prepend (stream->transports, tr);
- tr->active = TRUE;
- media->active++;
- } else if (remove && tr->active) {
- g_message ("removing TCP %s", trans->destination);
- stream->transports = g_list_remove (stream->transports, tr);
- tr->active = FALSE;
- media->active--;
- }
- break;
- default:
- g_message ("Unknown transport %d", trans->lower_transport);
- break;
+ if (add) {
+ if (gst_rtsp_stream_add_transport (trans->stream, trans))
+ media->n_active++;
+ } else if (remove) {
+ if (gst_rtsp_stream_remove_transport (trans->stream, trans))
+ media->n_active--;
}
}
if (old_active == 0 && add)
do_state = TRUE;
/* if we have no more active media, do the downward state changes */
- else if (media->active == 0)
+ else if (media->n_active == 0)
do_state = TRUE;
else
do_state = FALSE;
- g_message ("active %d media %p", media->active, media);
+ GST_INFO ("state %d active %d media %p do_state %d", state, media->n_active,
+ media, do_state);
- if (do_state && media->target_state != state) {
- if (state == GST_STATE_NULL) {
- gst_rtsp_media_unprepare (media);
- } else {
- g_message ("state %s media %p", gst_element_state_get_name (state), media);
- media->target_state = state;
- ret = gst_element_set_state (media->pipeline, state);
+ if (media->target_state != state) {
+ if (do_state) {
+ if (state == GST_STATE_NULL) {
+ gst_rtsp_media_unprepare (media);
+ } else {
+ GST_INFO ("state %s media %p", gst_element_state_get_name (state),
+ media);
+ media->target_state = state;
+ gst_element_set_state (media->pipeline, state);
+ }
}
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
+ NULL);
}
/* remember where we are */
- if (state == GST_STATE_PAUSED)
+ if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
+ old_active != media->n_active))
collect_media_stats (media);
- return TRUE;
-}
+ g_rec_mutex_unlock (&media->state_lock);
-void
-gst_rtsp_media_remove_elements (GstRTSPMedia *media)
-{
- gint i, j;
-
- unlock_streams (media);
-
- for (i = 0; i < media->streams->len; i++) {
- GstRTSPMediaStream *stream;
-
- g_message ("Removing elements of stream %d from pipeline", i);
-
- stream = g_array_index (media->streams, GstRTSPMediaStream *, i);
-
- gst_pad_unlink (stream->srcpad, stream->send_rtp_sink);
-
- for (j = 0; j < 2; j++) {
- gst_element_set_state (stream->udpsrc[j], GST_STATE_NULL);
- gst_element_set_state (stream->udpsink[j], GST_STATE_NULL);
- gst_element_set_state (stream->appsrc[j], GST_STATE_NULL);
- gst_element_set_state (stream->appsink[j], GST_STATE_NULL);
- gst_element_set_state (stream->tee[j], GST_STATE_NULL);
- gst_element_set_state (stream->selector[j], GST_STATE_NULL);
-
- gst_bin_remove (GST_BIN (media->pipeline), stream->udpsrc[j]);
- gst_bin_remove (GST_BIN (media->pipeline), stream->udpsink[j]);
- gst_bin_remove (GST_BIN (media->pipeline), stream->appsrc[j]);
- gst_bin_remove (GST_BIN (media->pipeline), stream->appsink[j]);
- gst_bin_remove (GST_BIN (media->pipeline), stream->tee[j]);
- gst_bin_remove (GST_BIN (media->pipeline), stream->selector[j]);
- }
- gst_rtsp_media_stream_free (stream);
- }
- g_array_remove_range (media->streams, 0, media->streams->len);
-
- gst_element_set_state (media->rtpbin, GST_STATE_NULL);
- gst_bin_remove (GST_BIN (media->pipeline), media->rtpbin);
+ return TRUE;
}
-
projects / platform / upstream / gstreamer.git / blobdiff