mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-18 14:26:43 +00:00
Code Issues 1 Projects Releases Wiki Activity

tests/webrtc: always use a unique SSRC for each stream

Browse source 
Will become more relevant with mid/rid->ssrc mappings

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
This commit is contained in:
Matthew Waters 2021-07-21 14:29:35 +10:00 committed by GStreamer Marge Bot
parent 533d4937fe
commit 246374c4e7
1 changed files with 53 additions and 38 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

91
subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
View file

@ -35,8 +35,8 @@
#include "../../../ext/webrtc/utils.c"
#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 VP8_RTP_CAPS(pt) "application/x-rtp,payload=" G_STRINGIFY(pt) ",encoding-name=VP8,media=video,clock-rate=90000,ssrc=(uint)3484078951"
#define H264_RTP_CAPS(pt) "application/x-rtp,payload=" G_STRINGIFY(pt) ",encoding-name=H264,media=video,clock-rate=90000,ssrc=(uint)3484078952"
#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)
@ -982,20 +982,24 @@ on_sdp_media_setup (struct test_webrtc *t, GstElement * element,
}
static void
add_fake_audio_src_harness (GstHarness * h, gint pt)
add_fake_audio_src_harness (GstHarness * h, gint pt, guint ssrc)
{
GstCaps *caps = gst_caps_from_string (OPUS_RTP_CAPS (pt));
GstStructure *s = gst_caps_get_structure (caps, 0);
if (ssrc != 0)
gst_structure_set (s, "ssrc", G_TYPE_UINT, ssrc, NULL);
gst_structure_set (s, "payload", G_TYPE_INT, pt, NULL);
gst_harness_set_src_caps (h, caps);
gst_harness_add_src_parse (h, "fakesrc is-live=true", TRUE);
}
static void
add_fake_video_src_harness (GstHarness * h, gint pt)
add_fake_video_src_harness (GstHarness * h, gint pt, guint ssrc)
{
GstCaps *caps = gst_caps_from_string (VP8_RTP_CAPS (pt));
GstStructure *s = gst_caps_get_structure (caps, 0);
if (ssrc != 0)
gst_structure_set (s, "ssrc", G_TYPE_UINT, ssrc, NULL);
gst_structure_set (s, "payload", G_TYPE_INT, pt, NULL);
gst_harness_set_src_caps (h, caps);
gst_harness_add_src_parse (h, "fakesrc is-live=true", TRUE);
@ -1012,7 +1016,7 @@ create_audio_test (void)
t->on_pad_added = _pad_added_fakesink;
h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
return t;
@ -1128,7 +1132,7 @@ create_audio_video_test (void)
GstHarness *h;
h = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
add_fake_video_src_harness (h, 97);
add_fake_video_src_harness (h, 97, 0xBEEFDEAD);
t->harnesses = g_list_prepend (t->harnesses, h);
return t;
@ -1192,7 +1196,7 @@ GST_START_TEST (test_media_direction)
/* check the default media directions for transceivers */
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
@ -1696,7 +1700,7 @@ GST_START_TEST (test_add_recvonly_transceiver)
/* setup sendonly peer */
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
@ -1740,6 +1744,7 @@ GST_START_TEST (test_recvonly_sendonly)
/* setup recvonly transceiver */
caps = gst_caps_from_string (OPUS_RTP_CAPS (96));
gst_caps_set_simple (caps, "ssrc", G_TYPE_UINT, 0xDEADBEEF, NULL);
direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY;
g_signal_emit_by_name (t->webrtc1, "add-transceiver", direction, caps,
&trans);
@ -1749,7 +1754,7 @@ GST_START_TEST (test_recvonly_sendonly)
/* setup sendonly stream */
h = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
add_fake_audio_src_harness (h, 96);
add_fake_audio_src_harness (h, 96, 0xBEEFDEAD);
t->harnesses = g_list_prepend (t->harnesses, h);
g_signal_emit_by_name (t->webrtc1, "get-transceivers", &transceivers);
fail_unless (transceivers != NULL);
@ -1762,7 +1767,7 @@ GST_START_TEST (test_recvonly_sendonly)
/* setup sendonly peer */
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
@ -2789,7 +2794,7 @@ GST_START_TEST (test_duplicate_nego)
t->negotiation_data = &negotiation_flag;
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
@ -2832,11 +2837,11 @@ GST_START_TEST (test_dual_audio)
/* test that each mline gets a unique transceiver even with the same caps */
h = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
add_fake_audio_src_harness (h, 96);
add_fake_audio_src_harness (h, 96, 0xBEEFDEAD);
t->harnesses = g_list_prepend (t->harnesses, h);
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
t->on_negotiation_needed = NULL;
@ -3018,13 +3023,13 @@ GST_START_TEST (test_renego_add_stream)
/* negotiate an AV stream and then renegotiate an extra stream */
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
h = gst_harness_new_with_element (t->webrtc1, "sink_2", NULL);
add_fake_audio_src_harness (h, 98);
add_fake_audio_src_harness (h, 98, 0xBEEFFFFF);
t->harnesses = g_list_prepend (t->harnesses, h);
media_formats.next = &renego_fingerprint;
@ -3074,7 +3079,7 @@ GST_START_TEST (test_renego_stream_add_data_channel)
/* negotiate an AV stream and then renegotiate a data channel */
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
@ -3144,7 +3149,7 @@ GST_START_TEST (test_renego_data_channel_add_stream)
test_validate_sdp_full (t, &offer, &answer, 0, FALSE);
h = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
add_fake_audio_src_harness (h, 97);
add_fake_audio_src_harness (h, 97, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
media_formats.next = &renego_fingerprint;
@ -3200,7 +3205,7 @@ GST_START_TEST (test_renego_stream_data_channel_add_stream)
t->on_pad_added = _pad_added_fakesink;
h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
add_fake_audio_src_harness (h, 97);
add_fake_audio_src_harness (h, 97, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
fail_if (gst_element_set_state (t->webrtc1,
@ -3214,7 +3219,7 @@ GST_START_TEST (test_renego_stream_data_channel_add_stream)
test_validate_sdp_full (t, &offer, &answer, 0, FALSE);
h = gst_harness_new_with_element (t->webrtc1, "sink_2", NULL);
add_fake_audio_src_harness (h, 97);
add_fake_audio_src_harness (h, 97, 0xBEEFDEAD);
t->harnesses = g_list_prepend (t->harnesses, h);
media_formats.next = &renego_fingerprint;
@ -3293,13 +3298,13 @@ GST_START_TEST (test_bundle_renego_add_stream)
/* negotiate an AV stream and then renegotiate an extra stream */
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
h = gst_harness_new_with_element (t->webrtc1, "sink_2", NULL);
add_fake_audio_src_harness (h, 98);
add_fake_audio_src_harness (h, 98, 0xBEEFFFFF);
t->harnesses = g_list_prepend (t->harnesses, h);
offer_setup.next = &offer_bundle_only_sdp;
@ -3373,13 +3378,13 @@ GST_START_TEST (test_bundle_max_compat_max_bundle_renego_add_stream)
/* negotiate an AV stream and then renegotiate an extra stream */
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
h = gst_harness_new_with_element (t->webrtc1, "sink_2", NULL);
add_fake_audio_src_harness (h, 98);
add_fake_audio_src_harness (h, 98, 0xBEEFFFFF);
t->harnesses = g_list_prepend (t->harnesses, h);
media_formats.next = &bundle_sdp;
@ -3421,7 +3426,7 @@ GST_START_TEST (test_renego_transceiver_set_direction)
/* negotiate an AV stream and then change the transceiver direction */
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
@ -3602,7 +3607,7 @@ GST_START_TEST (test_bundle_codec_preferences_rtx_no_duplicate_payloads)
/* setup sendonly peer */
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_video_src_harness (h, 96);
add_fake_video_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
@ -3782,7 +3787,7 @@ GST_START_TEST (test_reject_request_pad)
fail_unless (trans != NULL);
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_video_src_harness (h, 96);
add_fake_video_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
@ -3876,7 +3881,7 @@ GST_START_TEST (test_reject_create_offer)
/* setup sendonly peer */
h = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
add_fake_audio_src_harness (h, 96);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
/* Check that if there is no 0, we can't create an offer with a hole */
@ -3896,7 +3901,7 @@ GST_START_TEST (test_reject_create_offer)
gst_promise_unref (promise);
h = gst_harness_new_with_element (t->webrtc1, "sink_%u", NULL);
add_fake_video_src_harness (h, 97);
add_fake_video_src_harness (h, 97, 0xBEEFDEAD);
t->harnesses = g_list_prepend (t->harnesses, h);
/* Adding a second sink, which will fill m-line 0, should fix it */
@ -3926,7 +3931,7 @@ GST_START_TEST (test_reject_set_description)
/* setup peer 1 */
h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
/* Create a second side with specific video caps */
@ -4040,7 +4045,7 @@ GST_START_TEST (test_force_second_media)
/* setup peer */
h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
/* Create a second side with specific video caps */
@ -4052,7 +4057,7 @@ GST_START_TEST (test_force_second_media)
fail_unless (pad != NULL);
h = gst_harness_new_with_element (t->webrtc2, GST_PAD_NAME (pad), NULL);
gst_object_unref (pad);
add_fake_video_src_harness (h, 97);
add_fake_video_src_harness (h, 97, 0xBEEFDEAD);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer_count, &answer_count);
@ -4151,7 +4156,7 @@ GST_START_TEST (test_codec_preferences_negotiation_sinkpad)
gst_object_unref (transceiver);
gst_object_unref (pad);
add_fake_video_src_harness (h, 96);
add_fake_video_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
promise = gst_promise_new ();
@ -4181,14 +4186,24 @@ GST_END_TEST;
static void
add_audio_test_src_harness (GstHarness * h)
add_audio_test_src_harness (GstHarness * h, guint ssrc)
{
#define L16_CAPS "application/x-rtp, payload=11, media=audio," \
" encoding-name=L16, clock-rate=44100, ssrc=(uint)3484078952"
GstCaps *caps = gst_caps_from_string (L16_CAPS);
gst_harness_set_src_caps (h, caps);
GstElement *capsfilter;
if (ssrc != 0) {
gst_caps_set_simple (caps, "ssrc", G_TYPE_UINT, ssrc, NULL);
}
gst_harness_add_src_parse (h, "audiotestsrc is-live=true ! rtpL16pay ! "
L16_CAPS " ! identity", TRUE);
"capsfilter name=capsfilter ! identity", TRUE);
capsfilter =
gst_bin_get_by_name (GST_BIN (h->src_harness->element), "capsfilter");
g_object_set (G_OBJECT (capsfilter), "caps", caps, NULL);
gst_harness_set_src_caps (h, caps);
caps = NULL;
gst_clear_object (&capsfilter);
#undef L16_CAPS
}
static void
@ -4259,7 +4274,7 @@ GST_START_TEST (test_codec_preferences_negotiation_srcpad)
g_object_unref (rtpbin2);
h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
add_audio_test_src_harness (h);
add_audio_test_src_harness (h, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
@ -4380,7 +4395,7 @@ GST_START_TEST (test_codec_preferences_in_on_new_transceiver)
/* setup recvonly peer */
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_video_src_harness (h, 101);
add_fake_video_src_harness (h, 101, 0);
t->harnesses = g_list_prepend (t->harnesses, h);
/* connect to "on-new-transceiver" to set codec-preferences to H264 */
@ -4497,7 +4512,7 @@ GST_START_TEST (test_bundle_mid_header_extension)
/* setup sendonly peer */
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);