#include <gst/check/gstcheck.h>
#include <gst/check/gstharness.h>
#include <gst/webrtc/webrtc.h>
+#include <gst/rtp/rtp.h>
#include "../../../ext/webrtc/webrtcsdp.h"
#include "../../../ext/webrtc/webrtcsdp.c"
#include "../../../ext/webrtc/utils.h"
GstPad * pad, gpointer user_data)
{
GstHarness *h;
- GstHarness **sink_harness = user_data;
+ GList **sink_harness = user_data;
if (GST_PAD_DIRECTION (pad) != GST_PAD_SRC)
return;
t->harnesses = g_list_prepend (t->harnesses, h);
if (sink_harness) {
- *sink_harness = h;
+ *sink_harness = g_list_prepend (*sink_harness, h);
g_cond_broadcast (&t->cond);
}
}
VAL_SDP_INIT (answer_non_reject, _count_non_rejected_media,
GUINT_TO_POINTER (0), &count);
GstHarness *h;
+ GList *sink_harnesses = NULL;
GstHarness *sink_harness = NULL;
guint i;
GstElement *rtpbin2;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_harness;
- t->pad_added_data = &sink_harness;
+ t->pad_added_data = &sink_harnesses;
rtpbin2 = gst_bin_get_by_name (GST_BIN (t->webrtc2), "rtpbin");
fail_unless (rtpbin2 != NULL);
gst_harness_push_from_src (h);
g_mutex_lock (&t->lock);
- while (sink_harness == NULL) {
+ while (sink_harnesses == NULL) {
gst_harness_push_from_src (h);
g_cond_wait_until (&t->cond, &t->lock, g_get_monotonic_time () + 5000);
}
+ fail_unless_equals_int (1, g_list_length (sink_harnesses));
+ sink_harness = (GstHarness *) sink_harnesses->data;
g_mutex_unlock (&t->lock);
fail_unless (sink_harness->element == t->webrtc2);
}
test_webrtc_free (t);
+
+ g_list_free (sink_harnesses);
}
GST_END_TEST;
GST_END_TEST;
+static void
+on_new_transceiver_set_rtx_fec (GstElement * webrtcbin, GObject * trans,
+ gpointer user_data)
+{
+ g_object_set (trans, "fec-type", GST_WEBRTC_FEC_TYPE_ULP_RED,
+ "fec-percentage", 100, "do-nack", TRUE, NULL);
+}
+
+GST_START_TEST (test_max_bundle_fec)
+{
+ struct test_webrtc *t = test_webrtc_new ();
+ guint media_format_count[] = { 5, 5, };
+ VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+ media_format_count, NULL);
+ VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL,
+ &media_formats);
+ VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2), &payloads);
+ VAL_SDP_INIT (offer_non_reject, _count_non_rejected_media,
+ GUINT_TO_POINTER (1), &count);
+ VAL_SDP_INIT (answer_non_reject, _count_non_rejected_media,
+ GUINT_TO_POINTER (2), &count);
+ const gchar *expected_offer_setup[] = { "actpass", "actpass", };
+ VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
+ &offer_non_reject);
+ const gchar *expected_answer_setup[] = { "active", "active", };
+ VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+ &answer_non_reject);
+ const gchar *expected_offer_direction[] = { "sendrecv", "sendrecv", };
+ VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
+ &offer_setup);
+ const gchar *expected_answer_direction[] = { "recvonly", "recvonly", };
+ VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
+ &answer_setup);
+ GstHarness *src0, *src1;
+ GList *sink_harnesses = NULL;
+ guint i;
+ GstElement *rtpbin2;
+ GstBuffer *buf;
+ guint ssrcs[] = { 123456789, 987654321 };
+ GArray *ssrcs_received;
+
+ t->on_negotiation_needed = NULL;
+ t->on_ice_candidate = NULL;
+ t->on_pad_added = _pad_added_harness;
+ t->pad_added_data = &sink_harnesses;
+
+ 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");
+
+ rtpbin2 = gst_bin_get_by_name (GST_BIN (t->webrtc2), "rtpbin");
+ fail_unless (rtpbin2 != NULL);
+ g_signal_connect (rtpbin2, "new-jitterbuffer",
+ G_CALLBACK (new_jitterbuffer_set_fast_start), NULL);
+ g_object_unref (rtpbin2);
+
+ g_signal_connect (t->webrtc1, "on-new-transceiver",
+ G_CALLBACK (on_new_transceiver_set_rtx_fec), NULL);
+ g_signal_connect (t->webrtc2, "on-new-transceiver",
+ G_CALLBACK (on_new_transceiver_set_rtx_fec), NULL);
+
+ src0 = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
+ add_audio_test_src_harness (src0, ssrcs[0]);
+ t->harnesses = g_list_prepend (t->harnesses, src0);
+
+ src1 = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
+ add_audio_test_src_harness (src1, ssrcs[1]);
+ t->harnesses = g_list_prepend (t->harnesses, src1);
+
+ 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);
+
+ for (i = 0; i < 10; i++) {
+ gst_harness_push_from_src (src0);
+ gst_harness_push_from_src (src1);
+ }
+
+ ssrcs_received = g_array_new (FALSE, TRUE, sizeof (guint32));
+
+ /* Get one buffer out for each ssrc sent.
+ */
+ g_mutex_lock (&t->lock);
+ while (ssrcs_received->len < G_N_ELEMENTS (ssrcs)) {
+ GList *l;
+ guint i;
+
+ gst_harness_push_from_src (src0);
+ gst_harness_push_from_src (src1);
+ if (g_list_length (sink_harnesses) < 2) {
+ g_cond_wait_until (&t->cond, &t->lock, g_get_monotonic_time () + 5000);
+ if (g_list_length (sink_harnesses) < 2)
+ continue;
+ }
+
+ for (l = sink_harnesses; l; l = l->next) {
+ GstHarness *sink_harness = (GstHarness *) l->data;
+ GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
+ GstWebRTCRTPTransceiver *rtp_trans;
+ char *trans_mid;
+ GstPad *srcpad;
+ guint ssrc;
+ guint mlineindex;
+ char *expected_mid;
+
+ fail_unless (sink_harness->element == t->webrtc2);
+
+ buf = gst_harness_try_pull (sink_harness);
+ if (!buf)
+ continue;
+
+ /* ensure that the resulting pad has the correct mid set */
+ srcpad = gst_pad_get_peer (sink_harness->sinkpad);
+ fail_unless (srcpad != NULL);
+ g_object_get (srcpad, "transceiver", &rtp_trans, NULL);
+ gst_clear_object (&srcpad);
+ fail_unless (rtp_trans);
+ g_object_get (rtp_trans, "mid", &trans_mid, "mlineindex", &mlineindex,
+ NULL);
+ gst_clear_object (&rtp_trans);
+ expected_mid = g_strdup_printf ("audio%u", mlineindex);
+ fail_unless (trans_mid != NULL);
+ fail_unless_equals_string (trans_mid, expected_mid);
+ g_clear_pointer (&trans_mid, g_free);
+ g_clear_pointer (&expected_mid, g_free);
+
+ fail_unless (gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp));
+
+ ssrc = gst_rtp_buffer_get_ssrc (&rtp);
+ for (i = 0; i < ssrcs_received->len; i++) {
+ if (g_array_index (ssrcs_received, guint, i) == ssrc)
+ break;
+ }
+ if (i == ssrcs_received->len) {
+ g_array_append_val (ssrcs_received, ssrc);
+ }
+
+ gst_rtp_buffer_unmap (&rtp);
+
+ gst_buffer_unref (buf);
+ }
+ }
+ g_mutex_unlock (&t->lock);
+
+ GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN (t->webrtc1), GST_DEBUG_GRAPH_SHOW_ALL,
+ "webrtc1-fec-final");
+ GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN (t->webrtc2), GST_DEBUG_GRAPH_SHOW_ALL,
+ "webrtc2-fec-final");
+
+ test_webrtc_free (t);
+ g_list_free (sink_harnesses);
+
+ g_array_unref (ssrcs_received);
+}
+
+GST_END_TEST;
+
static Suite *
webrtcbin_suite (void)
{
tcase_add_test (tc, test_codec_preferences_invalid_extmap);
tcase_add_test (tc, test_renego_rtx);
tcase_add_test (tc, test_bundle_mid_header_extension);
+ tcase_add_test (tc, test_max_bundle_fec);
if (sctpenc && sctpdec) {
tcase_add_test (tc, test_data_channel_create);
tcase_add_test (tc, test_data_channel_remote_notify);
projects / platform / upstream / gstreamer.git / commitdiff