mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-26 11:41:09 +00:00
Code Issues 1 Projects Releases Wiki Activity

webrtcbin: connect output stream on recv transceivers

Browse source 
With MR 7156, transceivers and transports are created earlier,
but for sendrecv media we could get `not-linked` errors due to
transportreceivebin not being connected to rtpbin yet when incoming
data arrives.

This condition wasn't being tested in elements_webrtcbin, but could be
reproduced in the webrtcbidirectional example. This commit now also
adds a test for this, so that this doesn't regress anymore.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7294>
This commit is contained in:
Carlos Bentzen 2024-08-02 11:21:13 +02:00
parent cad3e63546
commit efa0a3ec6a
2 changed files with 70 additions and 0 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

6
subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
View file

@ -6500,6 +6500,12 @@ _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
webrtc_transceiver_set_transport (wtrans, stream); webrtc_transceiver_set_transport (wtrans, stream);
} }
} }
wtrans = WEBRTC_TRANSCEIVER (trans);
if (wtrans->stream
&& (direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV
|| direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY))
_connect_output_stream (webrtc, wtrans->stream, transport_idx);
} }
ret = TRUE; ret = TRUE;

64
subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
View file

@ -4601,6 +4601,69 @@ _pad_added_harness (struct test_webrtc *t, GstElement * element,
} }
} }
GST_START_TEST (test_audio_sendrecv)
{
struct test_webrtc *t = test_webrtc_new ();
GstHarness *h1, *h2;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
h1 = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
add_audio_test_src_harness (h1, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h1);
h2 = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_audio_test_src_harness (h2, 0xBEEFDEAD);
t->harnesses = g_list_prepend (t->harnesses, h2);
VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
NULL, NULL);
guint media_format_count[] = { 1 };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, &no_duplicate_payloads);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (1),
&media_formats);
const gchar *expected_offer_setup[] = { "actpass", };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &count);
const gchar *expected_answer_setup[] = { "active", };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&count);
const gchar *expected_offer_direction[] = { "sendrecv", };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { "sendrecv", };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
GstWebRTCKind expected_kind = GST_WEBRTC_KIND_AUDIO;
g_signal_connect (t->webrtc1, "on-new-transceiver",
G_CALLBACK (on_new_transceiver_expected_kind),
GUINT_TO_POINTER (expected_kind));
g_signal_connect (t->webrtc2, "on-new-transceiver",
G_CALLBACK (on_new_transceiver_expected_kind),
GUINT_TO_POINTER (expected_kind));
test_validate_sdp (t, &offer, &answer);
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
/* Exchange a few buffers between webrtcbin1 and webrtcbin2 to check
that they can handle incoming data and we get no errors on the bus. */
for (int i = 0; i < 5; i++) {
gst_harness_push_from_src (h1);
gst_harness_push_from_src (h2);
}
test_webrtc_free (t);
}
GST_END_TEST;
static void static void
new_jitterbuffer_set_fast_start (GstElement * rtpbin, new_jitterbuffer_set_fast_start (GstElement * rtpbin,
GstElement * rtpjitterbuffer, guint session_id, guint ssrc, GstElement * rtpjitterbuffer, guint session_id, guint ssrc,
@ -6001,6 +6064,7 @@ webrtcbin_suite (void)
tcase_add_test (tc, test_session_stats); tcase_add_test (tc, test_session_stats);
tcase_add_test (tc, test_stats_with_stream); tcase_add_test (tc, test_stats_with_stream);
tcase_add_test (tc, test_audio); tcase_add_test (tc, test_audio);
tcase_add_test (tc, test_audio_sendrecv);
tcase_add_test (tc, test_ice_port_restriction); tcase_add_test (tc, test_ice_port_restriction);
tcase_add_test (tc, test_audio_video); tcase_add_test (tc, test_audio_video);
tcase_add_test (tc, test_media_direction); tcase_add_test (tc, test_media_direction);