mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-03 15:06:34 +00:00
8b96d8ee8d
The key is to make sure the jitterbuffer is set to NULL *before* the ptdemux. The race that existed would basically happen when ptdemux had reached READY, and the jitterbuffer would then push a buffer, triggering a new pad with a new payloadtype being added and ghosted to the rtpbin itself. However, the srcpad of the ptdemux would now be inactive, and all the sticky-event pushed on it would be swallowed, not allowing any to reach the ghost-pad. Then the buffer in-flight would come to the ghostpad, and we would assert that a buffer arrived before the necessary events. By simply re-ordering the state-changes, we ensure that there will be no buffer racing into the ptdemux while its state is being changed, and the problem disappears completely. Notice also that there is not point in disconnecting the signals on the ptdemux before this point, since we need the push-thread to settle down before we can do this in a non-racy way.
991 lines
28 KiB
C
991 lines
28 KiB
C
/* GStreamer
|
|
*
|
|
* unit test for gstrtpbin
|
|
*
|
|
* Copyright (C) <2009> Wim Taymans <wim.taymans@gmail.com>
|
|
*
|
|
* This library is free software; you can redistribute it and/or
|
|
* modify it under the terms of the GNU Library General Public
|
|
* License as published by the Free Software Foundation; either
|
|
* version 2 of the License, or (at your option) any later version.
|
|
*
|
|
* This library is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
* Library General Public License for more details.
|
|
*
|
|
* 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., 51 Franklin St, Fifth Floor,
|
|
* Boston, MA 02110-1301, USA.
|
|
*/
|
|
|
|
#include <gst/check/gstcheck.h>
|
|
#include <gst/check/gsttestclock.h>
|
|
#include <gst/check/gstharness.h>
|
|
|
|
#include <gst/rtp/gstrtpbuffer.h>
|
|
#include <gst/rtp/gstrtcpbuffer.h>
|
|
|
|
GST_START_TEST (test_pads)
|
|
{
|
|
GstElement *element;
|
|
GstPad *pad;
|
|
|
|
element = gst_element_factory_make ("rtpsession", NULL);
|
|
|
|
pad = gst_element_get_request_pad (element, "recv_rtcp_sink");
|
|
gst_object_unref (pad);
|
|
gst_object_unref (element);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_cleanup_send)
|
|
{
|
|
GstElement *rtpbin;
|
|
GstPad *rtp_sink, *rtp_src, *rtcp_src;
|
|
GObject *session;
|
|
gint count = 2;
|
|
|
|
rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
|
|
|
|
while (count--) {
|
|
/* request session 0 */
|
|
rtp_sink = gst_element_get_request_pad (rtpbin, "send_rtp_sink_0");
|
|
fail_unless (rtp_sink != NULL);
|
|
ASSERT_OBJECT_REFCOUNT (rtp_sink, "rtp_sink", 2);
|
|
|
|
/* this static pad should be created automatically now */
|
|
rtp_src = gst_element_get_static_pad (rtpbin, "send_rtp_src_0");
|
|
fail_unless (rtp_src != NULL);
|
|
ASSERT_OBJECT_REFCOUNT (rtp_src, "rtp_src", 2);
|
|
|
|
/* we should be able to get an internal session 0 now */
|
|
g_signal_emit_by_name (rtpbin, "get-internal-session", 0, &session);
|
|
fail_unless (session != NULL);
|
|
g_object_unref (session);
|
|
|
|
/* get the send RTCP pad too */
|
|
rtcp_src = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0");
|
|
fail_unless (rtcp_src != NULL);
|
|
ASSERT_OBJECT_REFCOUNT (rtcp_src, "rtcp_src", 2);
|
|
|
|
gst_element_release_request_pad (rtpbin, rtp_sink);
|
|
/* we should only have our refs to the pads now */
|
|
ASSERT_OBJECT_REFCOUNT (rtp_sink, "rtp_sink", 1);
|
|
ASSERT_OBJECT_REFCOUNT (rtp_src, "rtp_src", 1);
|
|
ASSERT_OBJECT_REFCOUNT (rtcp_src, "rtp_src", 2);
|
|
|
|
/* the other pad should be gone now */
|
|
fail_unless (gst_element_get_static_pad (rtpbin, "send_rtp_src_0") == NULL);
|
|
|
|
/* internal session should still be there */
|
|
g_signal_emit_by_name (rtpbin, "get-internal-session", 0, &session);
|
|
fail_unless (session != NULL);
|
|
g_object_unref (session);
|
|
|
|
/* release the RTCP pad */
|
|
gst_element_release_request_pad (rtpbin, rtcp_src);
|
|
/* we should only have our refs to the pads now */
|
|
ASSERT_OBJECT_REFCOUNT (rtp_sink, "rtp_sink", 1);
|
|
ASSERT_OBJECT_REFCOUNT (rtp_src, "rtp_src", 1);
|
|
ASSERT_OBJECT_REFCOUNT (rtcp_src, "rtp_src", 1);
|
|
|
|
/* the session should be gone now */
|
|
g_signal_emit_by_name (rtpbin, "get-internal-session", 0, &session);
|
|
fail_unless (session == NULL);
|
|
|
|
/* unref the request pad and the static pad */
|
|
gst_object_unref (rtp_sink);
|
|
gst_object_unref (rtp_src);
|
|
gst_object_unref (rtcp_src);
|
|
}
|
|
|
|
gst_object_unref (rtpbin);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
typedef struct
|
|
{
|
|
guint16 seqnum;
|
|
gboolean pad_added;
|
|
GstPad *pad;
|
|
GMutex lock;
|
|
GCond cond;
|
|
GstPad *sinkpad;
|
|
GList *pads;
|
|
GstCaps *caps;
|
|
} CleanupData;
|
|
|
|
static void
|
|
init_data (CleanupData * data)
|
|
{
|
|
data->seqnum = 10;
|
|
data->pad_added = FALSE;
|
|
g_mutex_init (&data->lock);
|
|
g_cond_init (&data->cond);
|
|
data->pads = NULL;
|
|
data->caps = NULL;
|
|
}
|
|
|
|
static void
|
|
clean_data (CleanupData * data)
|
|
{
|
|
g_list_foreach (data->pads, (GFunc) gst_object_unref, NULL);
|
|
g_list_free (data->pads);
|
|
g_mutex_clear (&data->lock);
|
|
g_cond_clear (&data->cond);
|
|
if (data->caps)
|
|
gst_caps_unref (data->caps);
|
|
}
|
|
|
|
static guint8 rtp_packet[] = { 0x80, 0x60, 0x94, 0xbc, 0x8f, 0x37, 0x4e, 0xb8,
|
|
0x44, 0xa8, 0xf3, 0x7c, 0x06, 0x6a, 0x0c, 0xce,
|
|
0x13, 0x25, 0x19, 0x69, 0x1f, 0x93, 0x25, 0x9d,
|
|
0x2b, 0x82, 0x31, 0x3b, 0x36, 0xc1, 0x3c, 0x13
|
|
};
|
|
|
|
static GstFlowReturn
|
|
chain_rtp_packet (GstPad * pad, CleanupData * data)
|
|
{
|
|
GstFlowReturn res;
|
|
GstSegment segment;
|
|
GstBuffer *buffer;
|
|
GstMapInfo map;
|
|
|
|
if (data->caps == NULL) {
|
|
data->caps = gst_caps_from_string ("application/x-rtp,"
|
|
"media=(string)audio, clock-rate=(int)44100, "
|
|
"encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1");
|
|
data->seqnum = 0;
|
|
}
|
|
|
|
gst_pad_send_event (pad, gst_event_new_stream_start (GST_OBJECT_NAME (pad)));
|
|
gst_pad_send_event (pad, gst_event_new_caps (data->caps));
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
gst_pad_send_event (pad, gst_event_new_segment (&segment));
|
|
|
|
buffer = gst_buffer_new_and_alloc (sizeof (rtp_packet));
|
|
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
|
|
memcpy (map.data, rtp_packet, sizeof (rtp_packet));
|
|
|
|
map.data[2] = (data->seqnum >> 8) & 0xff;
|
|
map.data[3] = data->seqnum & 0xff;
|
|
|
|
data->seqnum++;
|
|
gst_buffer_unmap (buffer, &map);
|
|
|
|
GST_BUFFER_DTS (buffer) = 0;
|
|
|
|
res = gst_pad_chain (pad, buffer);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
dummy_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
|
|
{
|
|
gst_buffer_unref (buffer);
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("application/x-rtp"));
|
|
|
|
|
|
static GstPad *
|
|
make_sinkpad (CleanupData * data)
|
|
{
|
|
GstPad *pad;
|
|
|
|
pad = gst_pad_new_from_static_template (&sink_factory, "sink");
|
|
|
|
gst_pad_set_chain_function (pad, dummy_chain);
|
|
gst_pad_set_active (pad, TRUE);
|
|
|
|
data->pads = g_list_prepend (data->pads, pad);
|
|
|
|
return pad;
|
|
}
|
|
|
|
static void
|
|
pad_added_cb (GstElement * rtpbin, GstPad * pad, CleanupData * data)
|
|
{
|
|
GstPad *sinkpad;
|
|
|
|
GST_DEBUG ("pad added %s:%s\n", GST_DEBUG_PAD_NAME (pad));
|
|
|
|
if (GST_PAD_IS_SINK (pad))
|
|
return;
|
|
|
|
fail_unless (data->pad_added == FALSE);
|
|
|
|
sinkpad = make_sinkpad (data);
|
|
fail_unless (gst_pad_link (pad, sinkpad) == GST_PAD_LINK_OK);
|
|
|
|
g_mutex_lock (&data->lock);
|
|
data->pad_added = TRUE;
|
|
data->pad = pad;
|
|
g_cond_signal (&data->cond);
|
|
g_mutex_unlock (&data->lock);
|
|
}
|
|
|
|
static void
|
|
pad_removed_cb (GstElement * rtpbin, GstPad * pad, CleanupData * data)
|
|
{
|
|
GST_DEBUG ("pad removed %s:%s\n", GST_DEBUG_PAD_NAME (pad));
|
|
|
|
if (data->pad != pad)
|
|
return;
|
|
|
|
fail_unless (data->pad_added == TRUE);
|
|
|
|
g_mutex_lock (&data->lock);
|
|
data->pad_added = FALSE;
|
|
g_cond_signal (&data->cond);
|
|
g_mutex_unlock (&data->lock);
|
|
}
|
|
|
|
GST_START_TEST (test_cleanup_recv)
|
|
{
|
|
GstElement *rtpbin;
|
|
GstPad *rtp_sink;
|
|
CleanupData data;
|
|
GstStateChangeReturn ret;
|
|
GstFlowReturn res;
|
|
gint count = 2;
|
|
|
|
init_data (&data);
|
|
|
|
rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
|
|
|
|
g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added_cb, &data);
|
|
g_signal_connect (rtpbin, "pad-removed", (GCallback) pad_removed_cb, &data);
|
|
|
|
ret = gst_element_set_state (rtpbin, GST_STATE_PLAYING);
|
|
fail_unless (ret == GST_STATE_CHANGE_SUCCESS);
|
|
|
|
while (count--) {
|
|
/* request session 0 */
|
|
rtp_sink = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_0");
|
|
fail_unless (rtp_sink != NULL);
|
|
ASSERT_OBJECT_REFCOUNT (rtp_sink, "rtp_sink", 2);
|
|
|
|
/* no sourcepads are created yet */
|
|
fail_unless (rtpbin->numsinkpads == 1);
|
|
fail_unless (rtpbin->numsrcpads == 0);
|
|
|
|
res = chain_rtp_packet (rtp_sink, &data);
|
|
GST_DEBUG ("res %d, %s\n", res, gst_flow_get_name (res));
|
|
fail_unless (res == GST_FLOW_OK);
|
|
|
|
res = chain_rtp_packet (rtp_sink, &data);
|
|
GST_DEBUG ("res %d, %s\n", res, gst_flow_get_name (res));
|
|
fail_unless (res == GST_FLOW_OK);
|
|
|
|
/* we wait for the new pad to appear now */
|
|
g_mutex_lock (&data.lock);
|
|
while (!data.pad_added)
|
|
g_cond_wait (&data.cond, &data.lock);
|
|
g_mutex_unlock (&data.lock);
|
|
|
|
/* sourcepad created now */
|
|
fail_unless (rtpbin->numsinkpads == 1);
|
|
fail_unless (rtpbin->numsrcpads == 1);
|
|
|
|
/* remove the session */
|
|
gst_element_release_request_pad (rtpbin, rtp_sink);
|
|
gst_object_unref (rtp_sink);
|
|
|
|
/* pad should be gone now */
|
|
g_mutex_lock (&data.lock);
|
|
while (data.pad_added)
|
|
g_cond_wait (&data.cond, &data.lock);
|
|
g_mutex_unlock (&data.lock);
|
|
|
|
/* nothing left anymore now */
|
|
fail_unless (rtpbin->numsinkpads == 0);
|
|
fail_unless (rtpbin->numsrcpads == 0);
|
|
}
|
|
|
|
ret = gst_element_set_state (rtpbin, GST_STATE_NULL);
|
|
fail_unless (ret == GST_STATE_CHANGE_SUCCESS);
|
|
|
|
gst_object_unref (rtpbin);
|
|
|
|
clean_data (&data);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_cleanup_recv2)
|
|
{
|
|
GstElement *rtpbin;
|
|
GstPad *rtp_sink;
|
|
CleanupData data;
|
|
GstStateChangeReturn ret;
|
|
GstFlowReturn res;
|
|
gint count = 2;
|
|
|
|
init_data (&data);
|
|
|
|
rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
|
|
|
|
g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added_cb, &data);
|
|
g_signal_connect (rtpbin, "pad-removed", (GCallback) pad_removed_cb, &data);
|
|
|
|
ret = gst_element_set_state (rtpbin, GST_STATE_PLAYING);
|
|
fail_unless (ret == GST_STATE_CHANGE_SUCCESS);
|
|
|
|
/* request session 0 */
|
|
rtp_sink = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_0");
|
|
fail_unless (rtp_sink != NULL);
|
|
ASSERT_OBJECT_REFCOUNT (rtp_sink, "rtp_sink", 2);
|
|
|
|
while (count--) {
|
|
/* no sourcepads are created yet */
|
|
fail_unless (rtpbin->numsinkpads == 1);
|
|
fail_unless (rtpbin->numsrcpads == 0);
|
|
|
|
res = chain_rtp_packet (rtp_sink, &data);
|
|
GST_DEBUG ("res %d, %s\n", res, gst_flow_get_name (res));
|
|
fail_unless (res == GST_FLOW_OK);
|
|
|
|
res = chain_rtp_packet (rtp_sink, &data);
|
|
GST_DEBUG ("res %d, %s\n", res, gst_flow_get_name (res));
|
|
fail_unless (res == GST_FLOW_OK);
|
|
|
|
/* we wait for the new pad to appear now */
|
|
g_mutex_lock (&data.lock);
|
|
while (!data.pad_added)
|
|
g_cond_wait (&data.cond, &data.lock);
|
|
g_mutex_unlock (&data.lock);
|
|
|
|
/* sourcepad created now */
|
|
fail_unless (rtpbin->numsinkpads == 1);
|
|
fail_unless (rtpbin->numsrcpads == 1);
|
|
|
|
/* change state */
|
|
ret = gst_element_set_state (rtpbin, GST_STATE_NULL);
|
|
fail_unless (ret == GST_STATE_CHANGE_SUCCESS);
|
|
|
|
/* pad should be gone now */
|
|
g_mutex_lock (&data.lock);
|
|
while (data.pad_added)
|
|
g_cond_wait (&data.cond, &data.lock);
|
|
g_mutex_unlock (&data.lock);
|
|
|
|
/* back to playing for the next round */
|
|
ret = gst_element_set_state (rtpbin, GST_STATE_PLAYING);
|
|
fail_unless (ret == GST_STATE_CHANGE_SUCCESS);
|
|
}
|
|
|
|
/* remove the session */
|
|
gst_element_release_request_pad (rtpbin, rtp_sink);
|
|
gst_object_unref (rtp_sink);
|
|
|
|
/* nothing left anymore now */
|
|
fail_unless (rtpbin->numsinkpads == 0);
|
|
fail_unless (rtpbin->numsrcpads == 0);
|
|
|
|
ret = gst_element_set_state (rtpbin, GST_STATE_NULL);
|
|
fail_unless (ret == GST_STATE_CHANGE_SUCCESS);
|
|
|
|
gst_object_unref (rtpbin);
|
|
|
|
clean_data (&data);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_request_pad_by_template_name)
|
|
{
|
|
GstElement *rtpbin;
|
|
GstPad *rtp_sink1, *rtp_sink2, *rtp_sink3;
|
|
|
|
rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
|
|
rtp_sink1 = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_%u");
|
|
fail_unless (rtp_sink1 != NULL);
|
|
fail_unless_equals_string (GST_PAD_NAME (rtp_sink1), "recv_rtp_sink_0");
|
|
ASSERT_OBJECT_REFCOUNT (rtp_sink1, "rtp_sink1", 2);
|
|
|
|
rtp_sink2 = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_%u");
|
|
fail_unless (rtp_sink2 != NULL);
|
|
fail_unless_equals_string (GST_PAD_NAME (rtp_sink2), "recv_rtp_sink_1");
|
|
ASSERT_OBJECT_REFCOUNT (rtp_sink2, "rtp_sink2", 2);
|
|
|
|
rtp_sink3 = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_%u");
|
|
fail_unless (rtp_sink3 != NULL);
|
|
fail_unless_equals_string (GST_PAD_NAME (rtp_sink3), "recv_rtp_sink_2");
|
|
ASSERT_OBJECT_REFCOUNT (rtp_sink3, "rtp_sink3", 2);
|
|
|
|
|
|
gst_element_release_request_pad (rtpbin, rtp_sink2);
|
|
gst_element_release_request_pad (rtpbin, rtp_sink1);
|
|
gst_element_release_request_pad (rtpbin, rtp_sink3);
|
|
ASSERT_OBJECT_REFCOUNT (rtp_sink3, "rtp_sink3", 1);
|
|
ASSERT_OBJECT_REFCOUNT (rtp_sink2, "rtp_sink2", 1);
|
|
ASSERT_OBJECT_REFCOUNT (rtp_sink1, "rtp_sink", 1);
|
|
gst_object_unref (rtp_sink1);
|
|
gst_object_unref (rtp_sink2);
|
|
gst_object_unref (rtp_sink3);
|
|
|
|
gst_object_unref (rtpbin);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static GstElement *
|
|
encoder_cb (GstElement * rtpbin, guint sessid, GstElement * bin)
|
|
{
|
|
GstPad *srcpad, *sinkpad;
|
|
|
|
fail_unless (sessid == 2);
|
|
|
|
GST_DEBUG ("making encoder");
|
|
sinkpad = gst_ghost_pad_new_no_target ("rtp_sink_2", GST_PAD_SINK);
|
|
srcpad = gst_ghost_pad_new_no_target ("rtp_src_2", GST_PAD_SRC);
|
|
|
|
gst_element_add_pad (bin, sinkpad);
|
|
gst_element_add_pad (bin, srcpad);
|
|
|
|
return gst_object_ref (bin);
|
|
}
|
|
|
|
static GstElement *
|
|
encoder_cb2 (GstElement * rtpbin, guint sessid, GstElement * bin)
|
|
{
|
|
GstPad *srcpad, *sinkpad;
|
|
|
|
fail_unless (sessid == 3);
|
|
|
|
GST_DEBUG ("making encoder");
|
|
sinkpad = gst_ghost_pad_new_no_target ("rtp_sink_3", GST_PAD_SINK);
|
|
srcpad = gst_ghost_pad_new_no_target ("rtp_src_3", GST_PAD_SRC);
|
|
|
|
gst_element_add_pad (bin, sinkpad);
|
|
gst_element_add_pad (bin, srcpad);
|
|
|
|
return gst_object_ref (bin);
|
|
}
|
|
|
|
GST_START_TEST (test_encoder)
|
|
{
|
|
GstElement *rtpbin, *bin;
|
|
GstPad *rtp_sink1, *rtp_sink2;
|
|
gulong id;
|
|
|
|
bin = gst_bin_new ("rtpenc");
|
|
|
|
rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
|
|
|
|
id = g_signal_connect (rtpbin, "request-rtp-encoder", (GCallback) encoder_cb,
|
|
bin);
|
|
|
|
rtp_sink1 = gst_element_get_request_pad (rtpbin, "send_rtp_sink_2");
|
|
fail_unless (rtp_sink1 != NULL);
|
|
fail_unless_equals_string (GST_PAD_NAME (rtp_sink1), "send_rtp_sink_2");
|
|
ASSERT_OBJECT_REFCOUNT (rtp_sink1, "rtp_sink1", 2);
|
|
|
|
g_signal_handler_disconnect (rtpbin, id);
|
|
|
|
id = g_signal_connect (rtpbin, "request-rtp-encoder", (GCallback) encoder_cb2,
|
|
bin);
|
|
|
|
rtp_sink2 = gst_element_get_request_pad (rtpbin, "send_rtp_sink_3");
|
|
fail_unless (rtp_sink2 != NULL);
|
|
|
|
/* remove the session */
|
|
gst_element_release_request_pad (rtpbin, rtp_sink1);
|
|
gst_object_unref (rtp_sink1);
|
|
|
|
gst_element_release_request_pad (rtpbin, rtp_sink2);
|
|
gst_object_unref (rtp_sink2);
|
|
|
|
/* nothing left anymore now */
|
|
fail_unless (rtpbin->numsinkpads == 0);
|
|
fail_unless (rtpbin->numsrcpads == 0);
|
|
|
|
gst_object_unref (rtpbin);
|
|
gst_object_unref (bin);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static GstElement *
|
|
decoder_cb (GstElement * rtpbin, guint sessid, gpointer user_data)
|
|
{
|
|
GstElement *bin;
|
|
GstPad *srcpad, *sinkpad;
|
|
|
|
bin = gst_bin_new (NULL);
|
|
|
|
GST_DEBUG ("making decoder");
|
|
sinkpad = gst_ghost_pad_new_no_target ("rtp_sink", GST_PAD_SINK);
|
|
srcpad = gst_ghost_pad_new_no_target ("rtp_src", GST_PAD_SRC);
|
|
|
|
gst_element_add_pad (bin, sinkpad);
|
|
gst_element_add_pad (bin, srcpad);
|
|
|
|
return bin;
|
|
}
|
|
|
|
GST_START_TEST (test_decoder)
|
|
{
|
|
GstElement *rtpbin;
|
|
GstPad *rtp_sink1, *rtp_sink2;
|
|
gulong id;
|
|
|
|
|
|
rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
|
|
|
|
id = g_signal_connect (rtpbin, "request-rtp-decoder", (GCallback) decoder_cb,
|
|
NULL);
|
|
|
|
rtp_sink1 = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_2");
|
|
fail_unless (rtp_sink1 != NULL);
|
|
fail_unless_equals_string (GST_PAD_NAME (rtp_sink1), "recv_rtp_sink_2");
|
|
ASSERT_OBJECT_REFCOUNT (rtp_sink1, "rtp_sink1", 2);
|
|
|
|
rtp_sink2 = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_3");
|
|
fail_unless (rtp_sink2 != NULL);
|
|
|
|
g_signal_handler_disconnect (rtpbin, id);
|
|
|
|
/* remove the session */
|
|
gst_element_release_request_pad (rtpbin, rtp_sink1);
|
|
gst_object_unref (rtp_sink1);
|
|
|
|
gst_element_release_request_pad (rtpbin, rtp_sink2);
|
|
gst_object_unref (rtp_sink2);
|
|
|
|
/* nothing left anymore now */
|
|
fail_unless (rtpbin->numsinkpads == 0);
|
|
fail_unless (rtpbin->numsrcpads == 0);
|
|
|
|
gst_object_unref (rtpbin);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static GstElement *
|
|
aux_sender_cb (GstElement * rtpbin, guint sessid, gpointer user_data)
|
|
{
|
|
GstElement *bin;
|
|
GstPad *srcpad, *sinkpad;
|
|
|
|
bin = (GstElement *) user_data;
|
|
|
|
GST_DEBUG ("making AUX sender");
|
|
sinkpad = gst_ghost_pad_new_no_target ("sink_2", GST_PAD_SINK);
|
|
gst_element_add_pad (bin, sinkpad);
|
|
|
|
srcpad = gst_ghost_pad_new_no_target ("src_2", GST_PAD_SRC);
|
|
gst_element_add_pad (bin, srcpad);
|
|
srcpad = gst_ghost_pad_new_no_target ("src_1", GST_PAD_SRC);
|
|
gst_element_add_pad (bin, srcpad);
|
|
srcpad = gst_ghost_pad_new_no_target ("src_3", GST_PAD_SRC);
|
|
gst_element_add_pad (bin, srcpad);
|
|
|
|
return bin;
|
|
}
|
|
|
|
GST_START_TEST (test_aux_sender)
|
|
{
|
|
GstElement *rtpbin;
|
|
GstPad *rtp_sink1, *rtp_src, *rtcp_src;
|
|
gulong id;
|
|
GstElement *aux_sender = gst_object_ref_sink (gst_bin_new ("aux-sender"));
|
|
|
|
gst_object_ref (aux_sender);
|
|
|
|
rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
|
|
|
|
id = g_signal_connect (rtpbin, "request-aux-sender",
|
|
(GCallback) aux_sender_cb, aux_sender);
|
|
|
|
rtp_sink1 = gst_element_get_request_pad (rtpbin, "send_rtp_sink_2");
|
|
fail_unless (rtp_sink1 != NULL);
|
|
fail_unless_equals_string (GST_PAD_NAME (rtp_sink1), "send_rtp_sink_2");
|
|
ASSERT_OBJECT_REFCOUNT (rtp_sink1, "rtp_sink1", 2);
|
|
|
|
g_signal_handler_disconnect (rtpbin, id);
|
|
|
|
rtp_src = gst_element_get_static_pad (rtpbin, "send_rtp_src_2");
|
|
fail_unless (rtp_src != NULL);
|
|
gst_object_unref (rtp_src);
|
|
|
|
rtp_src = gst_element_get_static_pad (rtpbin, "send_rtp_src_1");
|
|
fail_unless (rtp_src != NULL);
|
|
gst_object_unref (rtp_src);
|
|
|
|
rtcp_src = gst_element_get_request_pad (rtpbin, "send_rtcp_src_1");
|
|
fail_unless (rtcp_src != NULL);
|
|
gst_element_release_request_pad (rtpbin, rtcp_src);
|
|
gst_object_unref (rtcp_src);
|
|
|
|
rtp_src = gst_element_get_static_pad (rtpbin, "send_rtp_src_3");
|
|
fail_unless (rtp_src != NULL);
|
|
gst_object_unref (rtp_src);
|
|
|
|
/* remove the session */
|
|
gst_element_release_request_pad (rtpbin, rtp_sink1);
|
|
gst_object_unref (rtp_sink1);
|
|
|
|
/* We have sinked the initial reference before returning it
|
|
* in the request callback, the ref count should now be 1 because
|
|
* the return of the signal is transfer full, and rtpbin should
|
|
* have released that reference by now, but we had taken an
|
|
* extra reference to perform this check
|
|
*/
|
|
ASSERT_OBJECT_REFCOUNT (aux_sender, "aux-sender", 1);
|
|
|
|
gst_object_unref (aux_sender);
|
|
gst_object_unref (rtpbin);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static GstElement *
|
|
aux_receiver_cb (GstElement * rtpbin, guint sessid, gpointer user_data)
|
|
{
|
|
GstElement *bin;
|
|
GstPad *srcpad, *sinkpad;
|
|
|
|
bin = gst_bin_new (NULL);
|
|
|
|
GST_DEBUG ("making AUX receiver");
|
|
srcpad = gst_ghost_pad_new_no_target ("src_2", GST_PAD_SRC);
|
|
gst_element_add_pad (bin, srcpad);
|
|
|
|
sinkpad = gst_ghost_pad_new_no_target ("sink_2", GST_PAD_SINK);
|
|
gst_element_add_pad (bin, sinkpad);
|
|
sinkpad = gst_ghost_pad_new_no_target ("sink_1", GST_PAD_SINK);
|
|
gst_element_add_pad (bin, sinkpad);
|
|
sinkpad = gst_ghost_pad_new_no_target ("sink_3", GST_PAD_SINK);
|
|
gst_element_add_pad (bin, sinkpad);
|
|
|
|
return bin;
|
|
}
|
|
|
|
GST_START_TEST (test_aux_receiver)
|
|
{
|
|
GstElement *rtpbin;
|
|
GstPad *rtp_sink1, *rtp_sink2, *rtcp_sink;
|
|
gulong id;
|
|
|
|
rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
|
|
|
|
id = g_signal_connect (rtpbin, "request-aux-receiver",
|
|
(GCallback) aux_receiver_cb, NULL);
|
|
|
|
rtp_sink1 = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_2");
|
|
fail_unless (rtp_sink1 != NULL);
|
|
|
|
rtp_sink2 = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_1");
|
|
fail_unless (rtp_sink2 != NULL);
|
|
|
|
g_signal_handler_disconnect (rtpbin, id);
|
|
|
|
rtcp_sink = gst_element_get_request_pad (rtpbin, "recv_rtcp_sink_1");
|
|
fail_unless (rtcp_sink != NULL);
|
|
gst_element_release_request_pad (rtpbin, rtcp_sink);
|
|
gst_object_unref (rtcp_sink);
|
|
|
|
/* remove the session */
|
|
gst_element_release_request_pad (rtpbin, rtp_sink1);
|
|
gst_object_unref (rtp_sink1);
|
|
gst_element_release_request_pad (rtpbin, rtp_sink2);
|
|
gst_object_unref (rtp_sink2);
|
|
|
|
gst_object_unref (rtpbin);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_sender_eos)
|
|
{
|
|
GstElement *rtpsession;
|
|
GstBuffer *rtp_buffer;
|
|
GstBuffer *rtcp_buffer;
|
|
GstRTPBuffer rtpbuf = GST_RTP_BUFFER_INIT;
|
|
GstRTCPBuffer rtcpbuf = GST_RTCP_BUFFER_INIT;
|
|
GstRTCPPacket rtcppacket;
|
|
static GstStaticPadTemplate recv_tmpl =
|
|
GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("ANY"));
|
|
GstPad *send_rtp_sink;
|
|
GstPad *recv_rtcp_sink;
|
|
GstCaps *caps;
|
|
GstSegment segment;
|
|
GstPad *rtp_sink, *rtcp_sink;
|
|
GstClock *clock;
|
|
GstTestClock *tclock;
|
|
GstStructure *s;
|
|
guint ssrc = 1;
|
|
guint32 ssrc_in, packet_count, octet_count;
|
|
gboolean got_bye = FALSE;
|
|
|
|
clock = gst_test_clock_new ();
|
|
gst_system_clock_set_default (clock);
|
|
tclock = GST_TEST_CLOCK (clock);
|
|
gst_test_clock_set_time (tclock, 0);
|
|
|
|
rtpsession = gst_element_factory_make ("rtpsession", NULL);
|
|
send_rtp_sink = gst_element_get_request_pad (rtpsession, "send_rtp_sink");
|
|
recv_rtcp_sink = gst_element_get_request_pad (rtpsession, "recv_rtcp_sink");
|
|
|
|
|
|
rtp_sink = gst_check_setup_sink_pad_by_name (rtpsession, &recv_tmpl,
|
|
"send_rtp_src");
|
|
rtcp_sink = gst_check_setup_sink_pad_by_name (rtpsession, &recv_tmpl,
|
|
"send_rtcp_src");
|
|
|
|
gst_pad_set_active (rtp_sink, TRUE);
|
|
gst_pad_set_active (rtcp_sink, TRUE);
|
|
|
|
gst_element_set_state (rtpsession, GST_STATE_PLAYING);
|
|
|
|
/* Send initial events */
|
|
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
fail_unless (gst_pad_send_event (send_rtp_sink,
|
|
gst_event_new_stream_start ("id")));
|
|
fail_unless (gst_pad_send_event (send_rtp_sink,
|
|
gst_event_new_segment (&segment)));
|
|
|
|
fail_unless (gst_pad_send_event (recv_rtcp_sink,
|
|
gst_event_new_stream_start ("id")));
|
|
fail_unless (gst_pad_send_event (recv_rtcp_sink,
|
|
gst_event_new_segment (&segment)));
|
|
|
|
/* Get the suggested SSRC from the rtpsession */
|
|
|
|
caps = gst_pad_query_caps (send_rtp_sink, NULL);
|
|
s = gst_caps_get_structure (caps, 0);
|
|
gst_structure_get (s, "ssrc", G_TYPE_UINT, &ssrc, NULL);
|
|
gst_caps_unref (caps);
|
|
|
|
/* Send a RTP packet */
|
|
|
|
rtp_buffer = gst_rtp_buffer_new_allocate (10, 0, 0);
|
|
gst_rtp_buffer_map (rtp_buffer, GST_MAP_READWRITE, &rtpbuf);
|
|
gst_rtp_buffer_set_ssrc (&rtpbuf, 1);
|
|
gst_rtp_buffer_set_seq (&rtpbuf, 0);
|
|
gst_rtp_buffer_unmap (&rtpbuf);
|
|
|
|
fail_unless (gst_pad_chain (send_rtp_sink, rtp_buffer) == GST_FLOW_OK);
|
|
|
|
/* Make sure it went through */
|
|
fail_unless_equals_int (g_list_length (buffers), 1);
|
|
fail_unless_equals_pointer (buffers->data, rtp_buffer);
|
|
gst_check_drop_buffers ();
|
|
|
|
/* Advance time and send a packet to prevent source sender timeout */
|
|
gst_test_clock_set_time (tclock, 1 * GST_SECOND);
|
|
|
|
/* Just send a send packet to prevent timeout */
|
|
rtp_buffer = gst_rtp_buffer_new_allocate (10, 0, 0);
|
|
gst_rtp_buffer_map (rtp_buffer, GST_MAP_READWRITE, &rtpbuf);
|
|
gst_rtp_buffer_set_ssrc (&rtpbuf, 1);
|
|
gst_rtp_buffer_set_seq (&rtpbuf, 1);
|
|
gst_rtp_buffer_set_timestamp (&rtpbuf, 10);
|
|
gst_rtp_buffer_unmap (&rtpbuf);
|
|
|
|
fail_unless (gst_pad_chain (send_rtp_sink, rtp_buffer) == GST_FLOW_OK);
|
|
|
|
/* Make sure it went through */
|
|
fail_unless_equals_int (g_list_length (buffers), 1);
|
|
fail_unless_equals_pointer (buffers->data, rtp_buffer);
|
|
gst_check_drop_buffers ();
|
|
|
|
/* Advance clock twice and we should have one RTCP packet at least */
|
|
gst_test_clock_crank (tclock);
|
|
gst_test_clock_crank (tclock);
|
|
|
|
g_mutex_lock (&check_mutex);
|
|
while (buffers == NULL)
|
|
g_cond_wait (&check_cond, &check_mutex);
|
|
|
|
fail_unless (gst_rtcp_buffer_map (buffers->data, GST_MAP_READ, &rtcpbuf));
|
|
|
|
fail_unless (gst_rtcp_buffer_get_first_packet (&rtcpbuf, &rtcppacket));
|
|
|
|
fail_unless_equals_int (gst_rtcp_packet_get_type (&rtcppacket),
|
|
GST_RTCP_TYPE_SR);
|
|
gst_rtcp_packet_sr_get_sender_info (&rtcppacket, &ssrc_in, NULL, NULL,
|
|
&packet_count, &octet_count);
|
|
fail_unless_equals_int (packet_count, 2);
|
|
fail_unless_equals_int (octet_count, 20);
|
|
|
|
fail_unless (gst_rtcp_packet_move_to_next (&rtcppacket));
|
|
fail_unless_equals_int (gst_rtcp_packet_get_type (&rtcppacket),
|
|
GST_RTCP_TYPE_SDES);
|
|
|
|
gst_rtcp_buffer_unmap (&rtcpbuf);
|
|
gst_check_drop_buffers ();
|
|
|
|
g_mutex_unlock (&check_mutex);
|
|
|
|
|
|
/* Create and send a valid RTCP reply packet */
|
|
rtcp_buffer = gst_rtcp_buffer_new (1500);
|
|
gst_rtcp_buffer_map (rtcp_buffer, GST_MAP_READWRITE, &rtcpbuf);
|
|
gst_rtcp_buffer_add_packet (&rtcpbuf, GST_RTCP_TYPE_RR, &rtcppacket);
|
|
gst_rtcp_packet_rr_set_ssrc (&rtcppacket, ssrc + 1);
|
|
gst_rtcp_packet_add_rb (&rtcppacket, ssrc, 0, 0, 0, 0, 0, 0);
|
|
gst_rtcp_buffer_add_packet (&rtcpbuf, GST_RTCP_TYPE_SDES, &rtcppacket);
|
|
gst_rtcp_packet_sdes_add_item (&rtcppacket, ssrc + 1);
|
|
gst_rtcp_packet_sdes_add_entry (&rtcppacket, GST_RTCP_SDES_CNAME, 3,
|
|
(guint8 *) "a@a");
|
|
gst_rtcp_packet_sdes_add_entry (&rtcppacket, GST_RTCP_SDES_NAME, 2,
|
|
(guint8 *) "aa");
|
|
gst_rtcp_packet_sdes_add_entry (&rtcppacket, GST_RTCP_SDES_END, 0,
|
|
(guint8 *) "");
|
|
gst_rtcp_buffer_unmap (&rtcpbuf);
|
|
fail_unless (gst_pad_chain (recv_rtcp_sink, rtcp_buffer) == GST_FLOW_OK);
|
|
|
|
|
|
/* Send a EOS to trigger sending a BYE message */
|
|
fail_unless (gst_pad_send_event (send_rtp_sink, gst_event_new_eos ()));
|
|
|
|
/* Crank to process EOS and wait for BYE */
|
|
for (;;) {
|
|
gst_test_clock_crank (tclock);
|
|
g_mutex_lock (&check_mutex);
|
|
while (buffers == NULL)
|
|
g_cond_wait (&check_cond, &check_mutex);
|
|
|
|
fail_unless (gst_rtcp_buffer_map (g_list_last (buffers)->data, GST_MAP_READ,
|
|
&rtcpbuf));
|
|
fail_unless (gst_rtcp_buffer_get_first_packet (&rtcpbuf, &rtcppacket));
|
|
|
|
while (gst_rtcp_packet_move_to_next (&rtcppacket)) {
|
|
if (gst_rtcp_packet_get_type (&rtcppacket) == GST_RTCP_TYPE_BYE) {
|
|
got_bye = TRUE;
|
|
break;
|
|
}
|
|
}
|
|
g_mutex_unlock (&check_mutex);
|
|
gst_rtcp_buffer_unmap (&rtcpbuf);
|
|
|
|
if (got_bye)
|
|
break;
|
|
}
|
|
|
|
gst_check_drop_buffers ();
|
|
|
|
|
|
fail_unless (GST_PAD_IS_EOS (rtp_sink));
|
|
fail_unless (GST_PAD_IS_EOS (rtcp_sink));
|
|
|
|
gst_pad_set_active (rtp_sink, FALSE);
|
|
gst_pad_set_active (rtcp_sink, FALSE);
|
|
|
|
gst_check_teardown_pad_by_name (rtpsession, "send_rtp_src");
|
|
gst_check_teardown_pad_by_name (rtpsession, "send_rtcp_src");
|
|
gst_element_release_request_pad (rtpsession, send_rtp_sink);
|
|
gst_object_unref (send_rtp_sink);
|
|
gst_element_release_request_pad (rtpsession, recv_rtcp_sink);
|
|
gst_object_unref (recv_rtcp_sink);
|
|
|
|
gst_check_teardown_element (rtpsession);
|
|
|
|
gst_system_clock_set_default (NULL);
|
|
gst_object_unref (clock);
|
|
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static GstBuffer *
|
|
generate_rtp_buffer (GstClockTime ts,
|
|
guint seqnum, guint32 rtp_ts, guint pt, guint ssrc)
|
|
{
|
|
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
|
|
GstBuffer *buf = gst_rtp_buffer_new_allocate (0, 0, 0);
|
|
GST_BUFFER_PTS (buf) = ts;
|
|
GST_BUFFER_DTS (buf) = ts;
|
|
|
|
gst_rtp_buffer_map (buf, GST_MAP_READWRITE, &rtp);
|
|
gst_rtp_buffer_set_payload_type (&rtp, pt);
|
|
gst_rtp_buffer_set_seq (&rtp, seqnum);
|
|
gst_rtp_buffer_set_timestamp (&rtp, rtp_ts);
|
|
gst_rtp_buffer_set_ssrc (&rtp, ssrc);
|
|
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
return buf;
|
|
}
|
|
|
|
static GstCaps *
|
|
_request_pt_map (G_GNUC_UNUSED GstElement * rtpbin,
|
|
G_GNUC_UNUSED guint session_id, G_GNUC_UNUSED guint pt,
|
|
const GstCaps * caps)
|
|
{
|
|
return gst_caps_copy (caps);
|
|
}
|
|
|
|
static void
|
|
_pad_added (G_GNUC_UNUSED GstElement * rtpbin, GstPad * pad, GstHarness * h)
|
|
{
|
|
gst_harness_add_element_src_pad (h, pad);
|
|
}
|
|
|
|
GST_START_TEST (test_quick_shutdown)
|
|
{
|
|
for (guint r = 0; r < 1000; r++) {
|
|
guint i;
|
|
GstHarness *h = gst_harness_new_with_padnames ("rtpbin",
|
|
"recv_rtp_sink_0", NULL);
|
|
GstCaps *caps = gst_caps_new_simple ("application/x-rtp",
|
|
"clock-rate", G_TYPE_INT, 8000,
|
|
"payload", G_TYPE_INT, 100, NULL);
|
|
|
|
g_signal_connect (h->element, "request-pt-map",
|
|
G_CALLBACK (_request_pt_map), caps);
|
|
g_signal_connect (h->element, "pad-added", G_CALLBACK (_pad_added), h);
|
|
|
|
gst_harness_set_src_caps (h, gst_caps_copy (caps));
|
|
|
|
for (i = 0; i < 50; i++) {
|
|
gst_harness_push (h,
|
|
generate_rtp_buffer (i * GST_MSECOND * 20, i, i * 160, 100, 1234));
|
|
}
|
|
gst_harness_crank_single_clock_wait (h);
|
|
|
|
gst_caps_unref (caps);
|
|
gst_harness_teardown (h);
|
|
}
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static Suite *
|
|
rtpbin_suite (void)
|
|
{
|
|
Suite *s = suite_create ("rtpbin");
|
|
TCase *tc_chain = tcase_create ("general");
|
|
|
|
suite_add_tcase (s, tc_chain);
|
|
tcase_add_test (tc_chain, test_pads);
|
|
tcase_add_test (tc_chain, test_cleanup_send);
|
|
tcase_add_test (tc_chain, test_cleanup_recv);
|
|
tcase_add_test (tc_chain, test_cleanup_recv2);
|
|
tcase_add_test (tc_chain, test_request_pad_by_template_name);
|
|
tcase_add_test (tc_chain, test_encoder);
|
|
tcase_add_test (tc_chain, test_decoder);
|
|
tcase_add_test (tc_chain, test_aux_sender);
|
|
tcase_add_test (tc_chain, test_aux_receiver);
|
|
tcase_add_test (tc_chain, test_sender_eos);
|
|
tcase_add_test (tc_chain, test_quick_shutdown);
|
|
|
|
return s;
|
|
}
|
|
|
|
GST_CHECK_MAIN (rtpbin);
|