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webrtcbin: Add a test for setting codec preferences as part of "on-new-transceiver" when setting the remote offer

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2310>
This commit is contained in:
Sebastian Dröge 2021-06-25 12:14:03 +03:00 committed by GStreamer Marge Bot
parent 348d4229e7
commit dcc49f846b
1 changed files with 102 additions and 0 deletions
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102
tests/check/elements/webrtcbin.c
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@ -36,6 +36,7 @@
#define OPUS_RTP_CAPS(pt) "application/x-rtp,payload=" G_STRINGIFY(pt) ",encoding-name=OPUS,media=audio,clock-rate=48000,ssrc=(uint)3384078950"
#define VP8_RTP_CAPS(pt) "application/x-rtp,payload=" G_STRINGIFY(pt) ",encoding-name=VP8,media=video,clock-rate=90000,ssrc=(uint)3484078950"
#define H264_RTP_CAPS(pt) "application/x-rtp,payload=" G_STRINGIFY(pt) ",encoding-name=H264,media=video,clock-rate=90000,ssrc=(uint)3484078951"
#define TEST_IS_OFFER_ELEMENT(t, e) ((((t)->offerror == 1 && (e) == (t)->webrtc1) || ((t)->offerror == 2 && (e) == (t)->webrtc2)) ? TRUE : FALSE)
#define TEST_GET_OFFEROR(t) (TEST_IS_OFFER_ELEMENT(t, t->webrtc1) ? (t)->webrtc1 : t->webrtc2)
@ -1224,6 +1225,8 @@ on_sdp_media_payload_types (struct test_webrtc *t, GstElement * element,
fail_unless_equals_string (attr->value, "99 rtx/90000");
} else if (g_str_has_prefix (attr->value, "100")) {
fail_unless_equals_string (attr->value, "100 rtx/90000");
} else if (g_str_has_prefix (attr->value, "101")) {
fail_unless_equals_string (attr->value, "101 H264/90000");
}
}
}
@ -4126,6 +4129,104 @@ GST_START_TEST (test_codec_preferences_negotiation_srcpad)
GST_END_TEST;
static void
_on_new_transceiver_codec_preferences_h264 (GstElement * webrtcbin,
GstWebRTCRTPTransceiver * trans, gpointer * user_data)
{
GstCaps *caps;
caps = gst_caps_from_string ("application/x-rtp,encoding-name=(string)H264");
g_object_set (trans, "codec-preferences", caps, NULL);
gst_caps_unref (caps);
}
static void
on_sdp_media_payload_types_only_h264 (struct test_webrtc *t,
GstElement * element, GstWebRTCSessionDescription * desc,
gpointer user_data)
{
const GstSDPMedia *vmedia;
guint video_mline = GPOINTER_TO_UINT (user_data);
guint j;
vmedia = gst_sdp_message_get_media (desc->sdp, video_mline);
for (j = 0; j < gst_sdp_media_attributes_len (vmedia); j++) {
const GstSDPAttribute *attr = gst_sdp_media_get_attribute (vmedia, j);
if (!g_strcmp0 (attr->key, "rtpmap")) {
fail_unless_equals_string (attr->value, "101 H264/90000");
}
}
}
GST_START_TEST (test_codec_preferences_in_on_new_transceiver)
{
struct test_webrtc *t = test_webrtc_new ();
GstWebRTCRTPTransceiverDirection direction;
GstWebRTCRTPTransceiver *trans;
VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
NULL, NULL);
guint offer_media_format_count[] = { 2 };
guint answer_media_format_count[] = { 1 };
VAL_SDP_INIT (offer_media_formats, on_sdp_media_count_formats,
offer_media_format_count, &no_duplicate_payloads);
VAL_SDP_INIT (answer_media_formats, on_sdp_media_count_formats,
answer_media_format_count, &no_duplicate_payloads);
VAL_SDP_INIT (offer_count, _count_num_sdp_media, GUINT_TO_POINTER (1),
&offer_media_formats);
VAL_SDP_INIT (answer_count, _count_num_sdp_media, GUINT_TO_POINTER (1),
&answer_media_formats);
VAL_SDP_INIT (offer_payloads, on_sdp_media_payload_types,
GUINT_TO_POINTER (0), &offer_count);
VAL_SDP_INIT (answer_payloads, on_sdp_media_payload_types_only_h264,
GUINT_TO_POINTER (0), &answer_count);
const gchar *expected_offer_setup[] = { "actpass", };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&offer_payloads);
const gchar *expected_answer_setup[] = { "active", };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&answer_payloads);
const gchar *expected_offer_direction[] = { "sendonly", };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { "recvonly", };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
GstCaps *caps;
GstHarness *h;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
/* setup sendonly transceiver with VP8 and H264 */
caps = gst_caps_from_string (VP8_RTP_CAPS (97) ";" H264_RTP_CAPS (101));
direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY;
g_signal_emit_by_name (t->webrtc1, "add-transceiver", direction, caps,
&trans);
gst_caps_unref (caps);
fail_unless (trans != NULL);
gst_object_unref (trans);
/* setup recvonly peer */
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_video_src_harness (h, 101);
t->harnesses = g_list_prepend (t->harnesses, h);
/* connect to "on-new-transceiver" to set codec-preferences to H264 */
g_signal_connect (t->webrtc2, "on-new-transceiver",
G_CALLBACK (_on_new_transceiver_codec_preferences_h264), NULL);
/* Answer SDP should now have H264 only. Without the codec-preferences it
* would only have VP8 because that comes first in the SDP */
test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_END_TEST;
static Suite *
webrtcbin_suite (void)
@ -4177,6 +4278,7 @@ webrtcbin_suite (void)
tcase_add_test (tc, test_codec_preferences_caps);
tcase_add_test (tc, test_codec_preferences_negotiation_sinkpad);
tcase_add_test (tc, test_codec_preferences_negotiation_srcpad);
tcase_add_test (tc, test_codec_preferences_in_on_new_transceiver);
if (sctpenc && sctpdec) {
tcase_add_test (tc, test_data_channel_create);
tcase_add_test (tc, test_data_channel_remote_notify);