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shmsrc: Only connect to sink in PLAYING in live mode

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This commit is contained in:
Olivier Crête 2011-03-09 19:34:39 -05:00 committed by Sjoerd Simons
parent a2b6dfa37e
commit a646afcf88
1 changed files with 58 additions and 9 deletions
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67
sys/shm/gstshmsrc.c
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@ -71,6 +71,8 @@ static GstFlowReturn gst_shm_src_create (GstPushSrc * psrc,
GstBuffer ** outbuf);
static gboolean gst_shm_src_unlock (GstBaseSrc * bsrc);
static gboolean gst_shm_src_unlock_stop (GstBaseSrc * bsrc);
static GstStateChangeReturn gst_shm_src_change_state (GstElement * element,
GstStateChange transition);
static void gst_shm_pipe_inc (GstShmPipe * pipe);
static void gst_shm_pipe_dec (GstShmPipe * pipe);
@ -97,10 +99,12 @@ static void
gst_shm_src_class_init (GstShmSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstPushSrcClass *gstpush_src_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gstpush_src_class = (GstPushSrcClass *) klass;
@ -108,6 +112,8 @@ gst_shm_src_class_init (GstShmSrcClass * klass)
gobject_class->get_property = gst_shm_src_get_property;
gobject_class->finalize = gst_shm_src_finalize;
gstelement_class->change_state = gst_shm_src_change_state;
gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_shm_src_start);
gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_shm_src_stop);
gstbasesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_shm_src_unlock);
@ -197,28 +203,27 @@ gst_shm_src_get_property (GObject * object, guint prop_id,
}
static gboolean
gst_shm_src_start (GstBaseSrc * bsrc)
gst_shm_src_start_reading (GstShmSrc * self)
{
GstShmSrc *self = GST_SHM_SRC (bsrc);
GstShmPipe *gstpipe = g_slice_new0 (GstShmPipe);
gstpipe->use_count = 1;
gstpipe->src = gst_object_ref (self);
if (!self->socket_path) {
GST_ELEMENT_ERROR (bsrc, RESOURCE, NOT_FOUND,
GST_ELEMENT_ERROR (self, RESOURCE, NOT_FOUND,
("No path specified for socket."), (NULL));
return FALSE;
}
GST_DEBUG ("Opening socket %s", self->socket_path);
GST_DEBUG_OBJECT (self, "Opening socket %s", self->socket_path);
GST_OBJECT_LOCK (self);
gstpipe->pipe = sp_client_open (self->socket_path);
GST_OBJECT_UNLOCK (self);
if (!gstpipe->pipe) {
GST_ELEMENT_ERROR (bsrc, RESOURCE, OPEN_READ_WRITE,
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ_WRITE,
("Could not open socket %s: %d %s", self->socket_path, errno,
strerror (errno)), (NULL));
gst_shm_pipe_dec (gstpipe);
@ -237,11 +242,9 @@ gst_shm_src_start (GstBaseSrc * bsrc)
return TRUE;
}
static gboolean
gst_shm_src_stop (GstBaseSrc * bsrc)
static void
gst_shm_src_stop_reading (GstShmSrc * self)
{
GstShmSrc *self = GST_SHM_SRC (bsrc);
GST_DEBUG_OBJECT (self, "Stopping %p", self);
if (self->pipe) {
@ -253,6 +256,22 @@ gst_shm_src_stop (GstBaseSrc * bsrc)
gst_poll_fd_init (&self->pollfd);
gst_poll_set_flushing (self->poll, TRUE);
}
static gboolean
gst_shm_src_start (GstBaseSrc * bsrc)
{
if (gst_base_src_is_live (bsrc))
return TRUE;
else
return gst_shm_src_start_reading (GST_SHM_SRC (bsrc));
}
static gboolean
gst_shm_src_stop (GstBaseSrc * bsrc)
{
if (!gst_base_src_is_live (bsrc))
gst_shm_src_stop_reading (GST_SHM_SRC (bsrc));
return TRUE;
}
@ -339,6 +358,36 @@ gst_shm_src_create (GstPushSrc * psrc, GstBuffer ** outbuf)
return GST_FLOW_OK;
}
static GstStateChangeReturn
gst_shm_src_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstShmSrc *self = GST_SHM_SRC (element);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
if (gst_base_src_is_live (GST_BASE_SRC (element)))
if (!gst_shm_src_start_reading (self))
return GST_STATE_CHANGE_FAILURE;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
if (gst_base_src_is_live (GST_BASE_SRC (element)))
gst_shm_src_stop_reading (self);
default:
break;
}
return ret;
}
static gboolean
gst_shm_src_unlock (GstBaseSrc * bsrc)
{