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webrtc: don't generate duplicate rtx payloads when bundle-policy is set

Browse source 
It was possible to generate a SDP that had an RTX payload type
that matched one of the media payload types when providing caps via
codec_preferences without any sink pads.

Fixes

m=video 9 UDP/TLS/RTP/SAVPF 96
...
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 nack pli
a=fmtp:96 apt=96

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2046>
This commit is contained in:
Matthew Waters 2021-03-01 20:53:53 +11:00 committed by GStreamer Marge Bot
parent 4ccad5336f
commit 2bed220771
2 changed files with 116 additions and 0 deletions
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22
ext/webrtc/gstwebrtcbin.c
View file

@ -2641,6 +2641,7 @@ gather_pad_pt (GstWebRTCBinPad * pad, GArray * reserved_pts)
gint pt;
if (gst_structure_get_int (s, "payload", &pt)) {
GST_TRACE_OBJECT (pad, "have reserved pt %u from received caps", pt);
g_array_append_val (reserved_pts, pt);
}
}
@ -2651,11 +2652,32 @@ gather_reserved_pts (GstWebRTCBin * webrtc)
{
GstElement *element = GST_ELEMENT (webrtc);
GArray *reserved_pts = g_array_new (FALSE, FALSE, sizeof (guint));
guint i;
GST_OBJECT_LOCK (webrtc);
g_list_foreach (element->sinkpads, (GFunc) gather_pad_pt, reserved_pts);
g_list_foreach (webrtc->priv->pending_pads, (GFunc) gather_pad_pt,
reserved_pts);
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
GstWebRTCRTPTransceiver *trans;
trans = g_ptr_array_index (webrtc->priv->transceivers, i);
if (trans->codec_preferences) {
guint j, n;
gint pt;
n = gst_caps_get_size (trans->codec_preferences);
for (j = 0; j < n; j++) {
GstStructure *s = gst_caps_get_structure (trans->codec_preferences, j);
if (gst_structure_get_int (s, "payload", &pt)) {
GST_TRACE_OBJECT (trans, "have reserved pt %u from codec preferences",
pt);
g_array_append_val (reserved_pts, pt);
}
}
}
}
GST_OBJECT_UNLOCK (webrtc);
return reserved_pts;

94
tests/check/elements/webrtcbin.c
View file

@ -2945,6 +2945,98 @@ GST_START_TEST (test_renego_lose_media_fails)
GST_END_TEST;
static void
on_sdp_media_no_duplicate_payloads (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
GArray *media_formats = g_array_new (FALSE, FALSE, sizeof (int));
int i, j, k;
for (i = 0; i < gst_sdp_message_medias_len (desc->sdp); i++) {
const GstSDPMedia *media = gst_sdp_message_get_media (desc->sdp, i);
for (j = 0; j < gst_sdp_media_formats_len (media); j++) {
int pt = atoi (gst_sdp_media_get_format (media, j));
for (k = 0; k < media_formats->len; k++) {
int val = g_array_index (media_formats, int, k);
if (pt == val)
fail ("found an unexpected duplicate payload type %u within media %u",
pt, i);
}
g_array_append_val (media_formats, pt);
}
}
g_array_free (media_formats, TRUE);
}
static void
on_sdp_media_count_formats (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
guint *expected_n_media_formats = user_data;
int i;
for (i = 0; i < gst_sdp_message_medias_len (desc->sdp); i++) {
const GstSDPMedia *media = gst_sdp_message_get_media (desc->sdp, i);
fail_unless_equals_int (gst_sdp_media_formats_len (media),
expected_n_media_formats[i]);
}
}
GST_START_TEST (test_bundle_codec_preferences_rtx_no_duplicate_payloads)
{
struct test_webrtc *t = test_webrtc_new ();
GstWebRTCRTPTransceiverDirection direction;
GstWebRTCRTPTransceiver *trans;
const gchar *expected_offer[] = { "recvonly" };
const gchar *expected_answer[] = { "sendonly" };
guint offer_media_format_count[] = { 2, };
guint answer_media_format_count[] = { 1, };
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, NULL);
VAL_SDP_INIT (offer_media_formats, on_sdp_media_count_formats,
offer_media_format_count, &payloads);
VAL_SDP_INIT (answer_media_formats, on_sdp_media_count_formats,
answer_media_format_count, &payloads);
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer,
&offer_media_formats);
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer,
&answer_media_formats);
GstCaps *caps;
GstHarness *h;
/* add a transceiver that will only receive an opus stream and check that
* the created offer is marked as recvonly */
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
"max-bundle");
gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy",
"max-bundle");
/* setup recvonly transceiver */
caps = gst_caps_from_string (VP8_RTP_CAPS (96));
direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY;
g_signal_emit_by_name (t->webrtc1, "add-transceiver", direction, caps,
&trans);
g_object_set (GST_OBJECT (trans), "do-nack", TRUE, NULL);
gst_caps_unref (caps);
fail_unless (trans != NULL);
gst_object_unref (trans);
/* setup sendonly peer */
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_video_src_harness (h, 96);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_END_TEST;
static Suite *
webrtcbin_suite (void)
{
@ -2987,6 +3079,8 @@ webrtcbin_suite (void)
tcase_add_test (tc, test_bundle_max_compat_max_bundle_renego_add_stream);
tcase_add_test (tc, test_renego_transceiver_set_direction);
tcase_add_test (tc, test_renego_lose_media_fails);
tcase_add_test (tc,
test_bundle_codec_preferences_rtx_no_duplicate_payloads);
if (sctpenc && sctpdec) {
tcase_add_test (tc, test_data_channel_create);
tcase_add_test (tc, test_data_channel_remote_notify);