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webrtc: Fix racy unit test

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Don't reuse the same stats state structure across multiple
get-stats calls. Make each callback take a copy of the
non-changing fields it needs and use a local working copy
to avoid crashing.

Fixes problems with the unit test crashing sometimes for the
unit test introduced in MR !7338

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7387>
This commit is contained in:
Jan Schmidt 2024-08-20 19:20:34 +10:00 committed by GStreamer Marge Bot
parent 055b5af99e
commit 96c4bd8d9f
1 changed files with 18 additions and 11 deletions
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29
subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
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@ -1850,18 +1850,18 @@ validate_stats (struct stats_check_state *state)
static void
_on_stats (GstPromise * promise, gpointer user_data)
{
struct stats_check_state *state = user_data;
struct test_webrtc *t = state->t;
struct stats_check_state state = *(struct stats_check_state *) user_data;
struct test_webrtc *t = state.t;
const GstStructure *reply = gst_promise_get_reply (promise);
int i;
GST_LOG ("Got stats %" GST_PTR_FORMAT, reply);
state->stats = reply;
state.stats = reply;
validate_stats (state);
validate_stats (&state);
if (state->n_streams > 0 && state->saw_inbound_rtp && state->saw_outbound_rtp
&& state->saw_remote_inbound_rtp && state->saw_remote_outbound_rtp) {
if (state.n_streams > 0 && state.saw_inbound_rtp && state.saw_outbound_rtp
&& state.saw_remote_inbound_rtp && state.saw_remote_outbound_rtp) {
g_mutex_lock (&t->lock);
i = GPOINTER_TO_INT (t->user_data);
i++;
@ -1880,15 +1880,16 @@ _on_stats (GstPromise * promise, gpointer user_data)
GST_START_TEST (test_session_stats)
{
struct test_webrtc *t = test_webrtc_new ();
struct stats_check_state state = {.t = t, 0 };
GstPromise *p;
/* test that the stats generated without any streams are sane */
t->on_negotiation_needed = NULL;
test_validate_sdp (t, NULL, NULL);
struct stats_check_state state = {.t = t, 0 };
p = gst_promise_new_with_change_func (_on_stats, &state, NULL);
g_signal_emit_by_name (t->webrtc1, "get-stats", NULL, p);
p = gst_promise_new_with_change_func (_on_stats, &state, NULL);
g_signal_emit_by_name (t->webrtc2, "get-stats", NULL, p);
@ -1935,6 +1936,7 @@ GST_START_TEST (test_stats_with_stream)
p = gst_promise_new_with_change_func (_on_stats, &state, NULL);
g_signal_emit_by_name (t->webrtc1, "get-stats", NULL, p);
p = gst_promise_new_with_change_func (_on_stats, &state, NULL);
g_signal_emit_by_name (t->webrtc2, "get-stats", NULL, p);
@ -1982,12 +1984,14 @@ GST_START_TEST (test_stats_with_two_streams)
/* We need to push data until the connection is established and the
* statistics start reporting outbound-rtp stats before things will
* unblock below */
gst_harness_push_from_src (h1);
gst_harness_push_from_src (h2);
for (int i = 0; i < 5; i++) {
gst_harness_push_from_src (h1);
gst_harness_push_from_src (h2);
}
struct stats_check_state state = {.t = t,.n_streams = 2, 0 };
while (TRUE) {
struct stats_check_state state = {.t = t,.n_streams = 2, 0 };
g_usleep (100 * 1000);
p = gst_promise_new_with_change_func (_on_stats, &state, NULL);
g_signal_emit_by_name (t->webrtc1, "get-stats", NULL, p);
@ -1997,6 +2001,9 @@ GST_START_TEST (test_stats_with_two_streams)
if (test_webrtc_check_for_state_mask (t, 1 << STATE_CUSTOM))
break;
gst_harness_push_from_src (h1);
gst_harness_push_from_src (h2);
}
test_webrtc_free (t);