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gstreamer/tests/check/elements/webrtcbin.c
Matthew Waters 65c9c4b406 tests/webrtc: fix tests for no libnice 
webrtcbin will fail the state change to READY when libnice elements are
not available.
2018-11-26 17:13:08 +11:00

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/* GStreamer
*
* Unit tests for webrtcbin
*
* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/check/gstcheck.h>
#include <gst/check/gstharness.h>
#include <gst/webrtc/webrtc.h>
#include "../../../ext/webrtc/webrtcsdp.h"
#include "../../../ext/webrtc/webrtcsdp.c"
#include "../../../ext/webrtc/utils.h"
#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"
typedef enum
{
STATE_NEW,
STATE_NEGOTATION_NEEDED,
STATE_OFFER_CREATED,
STATE_ANSWER_CREATED,
STATE_EOS,
STATE_ERROR,
STATE_CUSTOM,
} TestState;
/* basic premise of this is that webrtc1 and webrtc2 are attempting to connect
* to each other in various configurations */
struct test_webrtc;
struct test_webrtc
{
GList *harnesses;
GThread *thread;
GMainLoop *loop;
GstBus *bus1;
GstBus *bus2;
GstElement *webrtc1;
GstElement *webrtc2;
GMutex lock;
GCond cond;
TestState state;
guint offerror;
gpointer user_data;
GDestroyNotify data_notify;
/* *INDENT-OFF* */
void (*on_negotiation_needed) (struct test_webrtc * t,
GstElement * element,
gpointer user_data);
gpointer negotiation_data;
GDestroyNotify negotiation_notify;
void (*on_ice_candidate) (struct test_webrtc * t,
GstElement * element,
guint mlineindex,
gchar * candidate,
GstElement * other,
gpointer user_data);
gpointer ice_candidate_data;
GDestroyNotify ice_candidate_notify;
GstWebRTCSessionDescription * (*on_offer_created) (struct test_webrtc * t,
GstElement * element,
GstPromise * promise,
gpointer user_data);
gpointer offer_data;
GDestroyNotify offer_notify;
GstWebRTCSessionDescription * (*on_answer_created) (struct test_webrtc * t,
GstElement * element,
GstPromise * promise,
gpointer user_data);
gpointer data_channel_data;
GDestroyNotify data_channel_notify;
void (*on_data_channel) (struct test_webrtc * t,
GstElement * element,
GObject *data_channel,
gpointer user_data);
gpointer answer_data;
GDestroyNotify answer_notify;
void (*on_pad_added) (struct test_webrtc * t,
GstElement * element,
GstPad * pad,
gpointer user_data);
gpointer pad_added_data;
GDestroyNotify pad_added_notify;
void (*bus_message) (struct test_webrtc * t,
GstBus * bus,
GstMessage * msg,
gpointer user_data);
gpointer bus_data;
GDestroyNotify bus_notify;
/* *INDENT-ON* */
};
static void
_on_answer_received (GstPromise * promise, gpointer user_data)
{
struct test_webrtc *t = user_data;
GstElement *offeror = t->offerror == 1 ? t->webrtc1 : t->webrtc2;
GstElement *answerer = t->offerror == 2 ? t->webrtc1 : t->webrtc2;
const GstStructure *reply;
GstWebRTCSessionDescription *answer = NULL;
gchar *desc;
reply = gst_promise_get_reply (promise);
gst_structure_get (reply, "answer",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &answer, NULL);
desc = gst_sdp_message_as_text (answer->sdp);
GST_INFO ("Created Answer: %s", desc);
g_free (desc);
g_mutex_lock (&t->lock);
if (t->on_answer_created) {
gst_webrtc_session_description_free (answer);
answer = t->on_answer_created (t, answerer, promise, t->answer_data);
}
gst_promise_unref (promise);
g_signal_emit_by_name (answerer, "set-local-description", answer, NULL);
g_signal_emit_by_name (offeror, "set-remote-description", answer, NULL);
t->state = STATE_ANSWER_CREATED;
g_cond_broadcast (&t->cond);
g_mutex_unlock (&t->lock);
gst_webrtc_session_description_free (answer);
}
static void
_on_offer_received (GstPromise * promise, gpointer user_data)
{
struct test_webrtc *t = user_data;
GstElement *offeror = t->offerror == 1 ? t->webrtc1 : t->webrtc2;
GstElement *answerer = t->offerror == 2 ? t->webrtc1 : t->webrtc2;
const GstStructure *reply;
GstWebRTCSessionDescription *offer = NULL;
gchar *desc;
reply = gst_promise_get_reply (promise);
gst_structure_get (reply, "offer",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
desc = gst_sdp_message_as_text (offer->sdp);
GST_INFO ("Created offer: %s", desc);
g_free (desc);
g_mutex_lock (&t->lock);
if (t->on_offer_created) {
gst_webrtc_session_description_free (offer);
offer = t->on_offer_created (t, offeror, promise, t->offer_data);
}
gst_promise_unref (promise);
g_signal_emit_by_name (offeror, "set-local-description", offer, NULL);
g_signal_emit_by_name (answerer, "set-remote-description", offer, NULL);
promise = gst_promise_new_with_change_func (_on_answer_received, t, NULL);
g_signal_emit_by_name (answerer, "create-answer", NULL, promise);
t->state = STATE_OFFER_CREATED;
g_cond_broadcast (&t->cond);
g_mutex_unlock (&t->lock);
gst_webrtc_session_description_free (offer);
}
static gboolean
_bus_watch (GstBus * bus, GstMessage * msg, struct test_webrtc *t)
{
g_mutex_lock (&t->lock);
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_STATE_CHANGED:
if (GST_ELEMENT (msg->src) == t->webrtc1
|| GST_ELEMENT (msg->src) == t->webrtc2) {
GstState old, new, pending;
gst_message_parse_state_changed (msg, &old, &new, &pending);
{
gchar *dump_name = g_strconcat ("%s-state_changed-",
GST_OBJECT_NAME (msg->src), gst_element_state_get_name (old), "_",
gst_element_state_get_name (new), NULL);
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (msg->src),
GST_DEBUG_GRAPH_SHOW_ALL, dump_name);
g_free (dump_name);
}
}
break;
case GST_MESSAGE_ERROR:{
GError *err = NULL;
gchar *dbg_info = NULL;
{
gchar *dump_name;
dump_name =
g_strconcat ("%s-error", GST_OBJECT_NAME (t->webrtc1), NULL);
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (t->webrtc1),
GST_DEBUG_GRAPH_SHOW_ALL, dump_name);
g_free (dump_name);
dump_name =
g_strconcat ("%s-error", GST_OBJECT_NAME (t->webrtc2), NULL);
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (t->webrtc2),
GST_DEBUG_GRAPH_SHOW_ALL, dump_name);
g_free (dump_name);
}
gst_message_parse_error (msg, &err, &dbg_info);
GST_WARNING ("ERROR from element %s: %s\n",
GST_OBJECT_NAME (msg->src), err->message);
GST_WARNING ("Debugging info: %s\n", (dbg_info) ? dbg_info : "none");
g_error_free (err);
g_free (dbg_info);
t->state = STATE_ERROR;
g_cond_broadcast (&t->cond);
break;
}
case GST_MESSAGE_EOS:{
{
gchar *dump_name;
dump_name = g_strconcat ("%s-eos", GST_OBJECT_NAME (t->webrtc1), NULL);
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (t->webrtc1),
GST_DEBUG_GRAPH_SHOW_ALL, dump_name);
g_free (dump_name);
dump_name = g_strconcat ("%s-eos", GST_OBJECT_NAME (t->webrtc2), NULL);
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (t->webrtc2),
GST_DEBUG_GRAPH_SHOW_ALL, dump_name);
g_free (dump_name);
}
GST_INFO ("EOS received\n");
t->state = STATE_EOS;
g_cond_broadcast (&t->cond);
break;
}
default:
break;
}
if (t->bus_message)
t->bus_message (t, bus, msg, t->bus_data);
g_mutex_unlock (&t->lock);
return TRUE;
}
static void
_on_negotiation_needed (GstElement * webrtc, struct test_webrtc *t)
{
g_mutex_lock (&t->lock);
if (t->on_negotiation_needed)
t->on_negotiation_needed (t, webrtc, t->negotiation_data);
if (t->state == STATE_NEW)
t->state = STATE_NEGOTATION_NEEDED;
g_cond_broadcast (&t->cond);
g_mutex_unlock (&t->lock);
}
static void
_on_ice_candidate (GstElement * webrtc, guint mlineindex, gchar * candidate,
struct test_webrtc *t)
{
GstElement *other;
g_mutex_lock (&t->lock);
other = webrtc == t->webrtc1 ? t->webrtc2 : t->webrtc1;
if (t->on_ice_candidate)
t->on_ice_candidate (t, webrtc, mlineindex, candidate, other,
t->ice_candidate_data);
g_signal_emit_by_name (other, "add-ice-candidate", mlineindex, candidate);
g_mutex_unlock (&t->lock);
}
static void
_on_pad_added (GstElement * webrtc, GstPad * new_pad, struct test_webrtc *t)
{
g_mutex_lock (&t->lock);
if (t->on_pad_added)
t->on_pad_added (t, webrtc, new_pad, t->pad_added_data);
g_mutex_unlock (&t->lock);
}
static void
_on_data_channel (GstElement * webrtc, GObject * data_channel,
struct test_webrtc *t)
{
g_mutex_lock (&t->lock);
if (t->on_data_channel)
t->on_data_channel (t, webrtc, data_channel, t->data_channel_data);
g_mutex_unlock (&t->lock);
}
static void
_pad_added_not_reached (struct test_webrtc *t, GstElement * element,
GstPad * pad, gpointer user_data)
{
g_assert_not_reached ();
}
static void
_ice_candidate_not_reached (struct test_webrtc *t, GstElement * element,
guint mlineindex, gchar * candidate, GstElement * other, gpointer user_data)
{
g_assert_not_reached ();
}
static void
_negotiation_not_reached (struct test_webrtc *t, GstElement * element,
gpointer user_data)
{
g_assert_not_reached ();
}
static void
_bus_no_errors (struct test_webrtc *t, GstBus * bus, GstMessage * msg,
gpointer user_data)
{
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_ERROR:{
g_assert_not_reached ();
break;
}
default:
break;
}
}
static GstWebRTCSessionDescription *
_offer_answer_not_reached (struct test_webrtc *t, GstElement * element,
GstPromise * promise, gpointer user_data)
{
g_assert_not_reached ();
}
static void
_on_data_channel_not_reached (struct test_webrtc *t, GstElement * element,
GObject * data_channel, gpointer user_data)
{
g_assert_not_reached ();
}
static void
_broadcast (struct test_webrtc *t)
{
g_mutex_lock (&t->lock);
g_cond_broadcast (&t->cond);
g_mutex_unlock (&t->lock);
}
static gboolean
_unlock_create_thread (GMutex * lock)
{
g_mutex_unlock (lock);
return G_SOURCE_REMOVE;
}
static gpointer
_bus_thread (struct test_webrtc *t)
{
g_mutex_lock (&t->lock);
t->loop = g_main_loop_new (NULL, FALSE);
g_idle_add ((GSourceFunc) _unlock_create_thread, &t->lock);
g_cond_broadcast (&t->cond);
g_main_loop_run (t->loop);
g_mutex_lock (&t->lock);
g_main_loop_unref (t->loop);
t->loop = NULL;
g_cond_broadcast (&t->cond);
g_mutex_unlock (&t->lock);
return NULL;
}
static void
element_added_disable_sync (GstBin * bin, GstBin * sub_bin,
GstElement * element, gpointer user_data)
{
GObjectClass *class = G_OBJECT_GET_CLASS (element);
if (g_object_class_find_property (class, "async"))
g_object_set (element, "async", FALSE, NULL);
if (g_object_class_find_property (class, "sync"))
g_object_set (element, "sync", FALSE, NULL);
}
static struct test_webrtc *
test_webrtc_new (void)
{
struct test_webrtc *ret = g_new0 (struct test_webrtc, 1);
ret->on_negotiation_needed = _negotiation_not_reached;
ret->on_ice_candidate = _ice_candidate_not_reached;
ret->on_pad_added = _pad_added_not_reached;
ret->on_offer_created = _offer_answer_not_reached;
ret->on_answer_created = _offer_answer_not_reached;
ret->on_data_channel = _on_data_channel_not_reached;
ret->bus_message = _bus_no_errors;
g_mutex_init (&ret->lock);
g_cond_init (&ret->cond);
ret->bus1 = gst_bus_new ();
ret->bus2 = gst_bus_new ();
gst_bus_add_watch (ret->bus1, (GstBusFunc) _bus_watch, ret);
gst_bus_add_watch (ret->bus2, (GstBusFunc) _bus_watch, ret);
ret->webrtc1 = gst_element_factory_make ("webrtcbin", NULL);
ret->webrtc2 = gst_element_factory_make ("webrtcbin", NULL);
fail_unless (ret->webrtc1 != NULL && ret->webrtc2 != NULL);
gst_element_set_bus (ret->webrtc1, ret->bus1);
gst_element_set_bus (ret->webrtc2, ret->bus2);
g_signal_connect (ret->webrtc1, "deep-element-added",
G_CALLBACK (element_added_disable_sync), NULL);
g_signal_connect (ret->webrtc2, "deep-element-added",
G_CALLBACK (element_added_disable_sync), NULL);
g_signal_connect (ret->webrtc1, "on-negotiation-needed",
G_CALLBACK (_on_negotiation_needed), ret);
g_signal_connect (ret->webrtc2, "on-negotiation-needed",
G_CALLBACK (_on_negotiation_needed), ret);
g_signal_connect (ret->webrtc1, "on-ice-candidate",
G_CALLBACK (_on_ice_candidate), ret);
g_signal_connect (ret->webrtc2, "on-ice-candidate",
G_CALLBACK (_on_ice_candidate), ret);
g_signal_connect (ret->webrtc1, "on-data-channel",
G_CALLBACK (_on_data_channel), ret);
g_signal_connect (ret->webrtc2, "on-data-channel",
G_CALLBACK (_on_data_channel), ret);
g_signal_connect (ret->webrtc1, "pad-added", G_CALLBACK (_on_pad_added), ret);
g_signal_connect (ret->webrtc2, "pad-added", G_CALLBACK (_on_pad_added), ret);
g_signal_connect_swapped (ret->webrtc1, "notify::ice-gathering-state",
G_CALLBACK (_broadcast), ret);
g_signal_connect_swapped (ret->webrtc2, "notify::ice-gathering-state",
G_CALLBACK (_broadcast), ret);
g_signal_connect_swapped (ret->webrtc1, "notify::ice-connection-state",
G_CALLBACK (_broadcast), ret);
g_signal_connect_swapped (ret->webrtc2, "notify::ice-connection-state",
G_CALLBACK (_broadcast), ret);
ret->thread = g_thread_new ("test-webrtc", (GThreadFunc) _bus_thread, ret);
g_mutex_lock (&ret->lock);
while (!ret->loop)
g_cond_wait (&ret->cond, &ret->lock);
g_mutex_unlock (&ret->lock);
return ret;
}
static void
test_webrtc_free (struct test_webrtc *t)
{
/* Otherwise while one webrtcbin is being destroyed, the other could
* generate a signal that calls into the destroyed webrtcbin */
g_signal_handlers_disconnect_by_data (t->webrtc1, t);
g_signal_handlers_disconnect_by_data (t->webrtc2, t);
g_main_loop_quit (t->loop);
g_mutex_lock (&t->lock);
while (t->loop)
g_cond_wait (&t->cond, &t->lock);
g_mutex_unlock (&t->lock);
g_thread_join (t->thread);
gst_bus_remove_watch (t->bus1);
gst_bus_remove_watch (t->bus2);
gst_bus_set_flushing (t->bus1, TRUE);
gst_bus_set_flushing (t->bus2, TRUE);
gst_object_unref (t->bus1);
gst_object_unref (t->bus2);
g_list_free_full (t->harnesses, (GDestroyNotify) gst_harness_teardown);
if (t->data_notify)
t->data_notify (t->user_data);
if (t->negotiation_notify)
t->negotiation_notify (t->negotiation_data);
if (t->ice_candidate_notify)
t->ice_candidate_notify (t->ice_candidate_data);
if (t->offer_notify)
t->offer_notify (t->offer_data);
if (t->answer_notify)
t->answer_notify (t->answer_data);
if (t->pad_added_notify)
t->pad_added_notify (t->pad_added_data);
if (t->data_channel_notify)
t->data_channel_notify (t->data_channel_data);
fail_unless_equals_int (GST_STATE_CHANGE_SUCCESS,
gst_element_set_state (t->webrtc1, GST_STATE_NULL));
fail_unless_equals_int (GST_STATE_CHANGE_SUCCESS,
gst_element_set_state (t->webrtc2, GST_STATE_NULL));
gst_object_unref (t->webrtc1);
gst_object_unref (t->webrtc2);
g_mutex_clear (&t->lock);
g_cond_clear (&t->cond);
g_free (t);
}
static void
test_webrtc_create_offer (struct test_webrtc *t, GstElement * webrtc)
{
GstPromise *promise;
t->offerror = webrtc == t->webrtc1 ? 1 : 2;
promise = gst_promise_new_with_change_func (_on_offer_received, t, NULL);
g_signal_emit_by_name (webrtc, "create-offer", NULL, promise);
}
static void
test_webrtc_wait_for_state_mask (struct test_webrtc *t, TestState state)
{
g_mutex_lock (&t->lock);
while (((1 << t->state) & state) == 0) {
GST_INFO ("test state 0x%x, current 0x%x", state, (1 << t->state));
g_cond_wait (&t->cond, &t->lock);
}
GST_INFO ("have test state 0x%x, current 0x%x", state, 1 << t->state);
g_mutex_unlock (&t->lock);
}
static void
test_webrtc_wait_for_answer_error_eos (struct test_webrtc *t)
{
TestState states = 0;
states |= (1 << STATE_ANSWER_CREATED);
states |= (1 << STATE_EOS);
states |= (1 << STATE_ERROR);
test_webrtc_wait_for_state_mask (t, states);
}
static void
test_webrtc_signal_state_unlocked (struct test_webrtc *t, TestState state)
{
t->state = state;
g_cond_broadcast (&t->cond);
}
static void
test_webrtc_signal_state (struct test_webrtc *t, TestState state)
{
g_mutex_lock (&t->lock);
test_webrtc_signal_state_unlocked (t, state);
g_mutex_unlock (&t->lock);
}
#if 0
static void
test_webrtc_wait_for_ice_gathering_complete (struct test_webrtc *t)
{
GstWebRTCICEGatheringState ice_state1, ice_state2;
g_mutex_lock (&t->lock);
g_object_get (t->webrtc1, "ice-gathering-state", &ice_state1, NULL);
g_object_get (t->webrtc2, "ice-gathering-state", &ice_state2, NULL);
while (ice_state1 != GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE &&
ice_state2 != GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE) {
g_cond_wait (&t->cond, &t->lock);
g_object_get (t->webrtc1, "ice-gathering-state", &ice_state1, NULL);
g_object_get (t->webrtc2, "ice-gathering-state", &ice_state2, NULL);
}
g_mutex_unlock (&t->lock);
}
static void
test_webrtc_wait_for_ice_connection (struct test_webrtc *t,
GstWebRTCICEConnectionState states)
{
GstWebRTCICEConnectionState ice_state1, ice_state2, current;
g_mutex_lock (&t->lock);
g_object_get (t->webrtc1, "ice-connection-state", &ice_state1, NULL);
g_object_get (t->webrtc2, "ice-connection-state", &ice_state2, NULL);
current = (1 << ice_state1) | (1 << ice_state2);
while ((current & states) == 0 || (current & ~states)) {
g_cond_wait (&t->cond, &t->lock);
g_object_get (t->webrtc1, "ice-connection-state", &ice_state1, NULL);
g_object_get (t->webrtc2, "ice-connection-state", &ice_state2, NULL);
current = (1 << ice_state1) | (1 << ice_state2);
}
g_mutex_unlock (&t->lock);
}
#endif
static void
_pad_added_fakesink (struct test_webrtc *t, GstElement * element,
GstPad * pad, gpointer user_data)
{
GstHarness *h;
if (GST_PAD_DIRECTION (pad) != GST_PAD_SRC)
return;
h = gst_harness_new_with_element (element, NULL, "src_%u");
gst_harness_add_sink_parse (h, "fakesink async=false sync=false");
t->harnesses = g_list_prepend (t->harnesses, h);
}
static GstWebRTCSessionDescription *
_count_num_sdp_media (struct test_webrtc *t, GstElement * element,
GstPromise * promise, gpointer user_data)
{
GstWebRTCSessionDescription *offer = NULL;
guint expected = GPOINTER_TO_UINT (user_data);
const GstStructure *reply;
const gchar *field;
field = t->offerror == 1 && t->webrtc1 == element ? "offer" : "answer";
reply = gst_promise_get_reply (promise);
gst_structure_get (reply, field,
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
fail_unless_equals_int (gst_sdp_message_medias_len (offer->sdp), expected);
return offer;
}
GST_START_TEST (test_sdp_no_media)
{
struct test_webrtc *t = test_webrtc_new ();
/* check that a no stream connection creates 0 media sections */
t->on_negotiation_needed = NULL;
t->offer_data = GUINT_TO_POINTER (0);
t->on_offer_created = _count_num_sdp_media;
t->answer_data = GUINT_TO_POINTER (0);
t->on_answer_created = _count_num_sdp_media;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_answer_error_eos (t);
fail_unless (t->state == STATE_ANSWER_CREATED);
test_webrtc_free (t);
}
GST_END_TEST;
static void
add_fake_audio_src_harness (GstHarness * h, gint pt)
{
GstCaps *caps = gst_caps_from_string (OPUS_RTP_CAPS (pt));
GstStructure *s = gst_caps_get_structure (caps, 0);
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)
{
GstCaps *caps = gst_caps_from_string (VP8_RTP_CAPS (pt));
GstStructure *s = gst_caps_get_structure (caps, 0);
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 struct test_webrtc *
create_audio_test (void)
{
struct test_webrtc *t = test_webrtc_new ();
GstHarness *h;
t->on_negotiation_needed = NULL;
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);
t->harnesses = g_list_prepend (t->harnesses, h);
return t;
}
GST_START_TEST (test_audio)
{
struct test_webrtc *t = create_audio_test ();
/* check that a single stream connection creates the associated number
* of media sections */
t->on_negotiation_needed = NULL;
t->offer_data = GUINT_TO_POINTER (1);
t->on_offer_created = _count_num_sdp_media;
t->answer_data = GUINT_TO_POINTER (1);
t->on_answer_created = _count_num_sdp_media;
t->on_ice_candidate = NULL;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_answer_error_eos (t);
fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
test_webrtc_free (t);
}
GST_END_TEST;
static struct test_webrtc *
create_audio_video_test (void)
{
struct test_webrtc *t = test_webrtc_new ();
GstHarness *h;
t->on_negotiation_needed = NULL;
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);
t->harnesses = g_list_prepend (t->harnesses, h);
h = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
add_fake_video_src_harness (h, 97);
t->harnesses = g_list_prepend (t->harnesses, h);
return t;
}
GST_START_TEST (test_audio_video)
{
struct test_webrtc *t = create_audio_video_test ();
/* check that a dual stream connection creates the associated number
* of media sections */
t->on_negotiation_needed = NULL;
t->offer_data = GUINT_TO_POINTER (2);
t->on_offer_created = _count_num_sdp_media;
t->answer_data = GUINT_TO_POINTER (2);
t->on_answer_created = _count_num_sdp_media;
t->on_ice_candidate = NULL;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_answer_error_eos (t);
fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
test_webrtc_free (t);
}
GST_END_TEST;
typedef void (*ValidateSDPFunc) (struct test_webrtc * t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data);
struct validate_sdp
{
ValidateSDPFunc validate;
gpointer user_data;
};
static GstWebRTCSessionDescription *
_check_validate_sdp (struct test_webrtc *t, GstElement * element,
GstPromise * promise, gpointer user_data)
{
struct validate_sdp *validate = user_data;
GstWebRTCSessionDescription *offer = NULL;
const GstStructure *reply;
const gchar *field;
field = t->offerror == 1 && t->webrtc1 == element ? "offer" : "answer";
reply = gst_promise_get_reply (promise);
gst_structure_get (reply, field,
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
validate->validate (t, element, offer, validate->user_data);
return offer;
}
static void
on_sdp_media_direction (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
gchar **expected_directions = 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);
gboolean have_direction = FALSE;
int j;
for (j = 0; j < gst_sdp_media_attributes_len (media); j++) {
const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, j);
if (g_strcmp0 (attr->key, "inactive") == 0) {
fail_unless (have_direction == FALSE,
"duplicate/multiple directions for media %u", j);
have_direction = TRUE;
fail_unless (g_strcmp0 (attr->key, expected_directions[i]) == 0);
} else if (g_strcmp0 (attr->key, "sendonly") == 0) {
fail_unless (have_direction == FALSE,
"duplicate/multiple directions for media %u", j);
have_direction = TRUE;
fail_unless (g_strcmp0 (attr->key, expected_directions[i]) == 0);
} else if (g_strcmp0 (attr->key, "recvonly") == 0) {
fail_unless (have_direction == FALSE,
"duplicate/multiple directions for media %u", j);
have_direction = TRUE;
fail_unless (g_strcmp0 (attr->key, expected_directions[i]) == 0);
} else if (g_strcmp0 (attr->key, "sendrecv") == 0) {
fail_unless (have_direction == FALSE,
"duplicate/multiple directions for media %u", j);
have_direction = TRUE;
fail_unless (g_strcmp0 (attr->key, expected_directions[i]) == 0);
}
}
fail_unless (have_direction, "no direction attribute in media %u", j);
}
}
GST_START_TEST (test_media_direction)
{
struct test_webrtc *t = create_audio_video_test ();
const gchar *expected_offer[] = { "sendrecv", "sendrecv" };
const gchar *expected_answer[] = { "sendrecv", "recvonly" };
struct validate_sdp offer = { on_sdp_media_direction, expected_offer };
struct validate_sdp answer = { on_sdp_media_direction, expected_answer };
GstHarness *h;
/* 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);
t->harnesses = g_list_prepend (t->harnesses, h);
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_answer_error_eos (t);
fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
test_webrtc_free (t);
}
GST_END_TEST;
static void
on_sdp_media_payload_types (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
const GstSDPMedia *vmedia;
guint j;
fail_unless_equals_int (gst_sdp_message_medias_len (desc->sdp), 2);
vmedia = gst_sdp_message_get_media (desc->sdp, 1);
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")) {
if (g_str_has_prefix (attr->value, "97")) {
fail_unless_equals_string (attr->value, "97 VP8/90000");
} else if (g_str_has_prefix (attr->value, "96")) {
fail_unless_equals_string (attr->value, "96 red/90000");
} else if (g_str_has_prefix (attr->value, "98")) {
fail_unless_equals_string (attr->value, "98 ulpfec/90000");
} else if (g_str_has_prefix (attr->value, "99")) {
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");
}
}
}
}
/* In this test we verify that webrtcbin will pick available payload
* types when it needs to, in that example for RTX and FEC */
GST_START_TEST (test_payload_types)
{
struct test_webrtc *t = create_audio_video_test ();
struct validate_sdp offer = { on_sdp_media_payload_types, NULL };
GstWebRTCRTPTransceiver *trans;
GArray *transceivers;
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
/* We don't really care about the answer here */
t->on_answer_created = NULL;
g_signal_emit_by_name (t->webrtc1, "get-transceivers", &transceivers);
fail_unless_equals_int (transceivers->len, 2);
trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 1);
g_object_set (trans, "fec-type", GST_WEBRTC_FEC_TYPE_ULP_RED, "do-nack", TRUE,
NULL);
g_array_unref (transceivers);
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_answer_error_eos (t);
fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
test_webrtc_free (t);
}
GST_END_TEST;
static void
on_sdp_media_setup (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
gchar **expected_setup = 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);
gboolean have_setup = FALSE;
int j;
for (j = 0; j < gst_sdp_media_attributes_len (media); j++) {
const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, j);
if (g_strcmp0 (attr->key, "setup") == 0) {
fail_unless (have_setup == FALSE,
"duplicate/multiple setup for media %u", j);
have_setup = TRUE;
fail_unless (g_strcmp0 (attr->value, expected_setup[i]) == 0);
}
}
fail_unless (have_setup, "no setup attribute in media %u", j);
}
}
GST_START_TEST (test_media_setup)
{
struct test_webrtc *t = create_audio_test ();
const gchar *expected_offer[] = { "actpass" };
const gchar *expected_answer[] = { "active" };
struct validate_sdp offer = { on_sdp_media_setup, expected_offer };
struct validate_sdp answer = { on_sdp_media_setup, expected_answer };
/* check the default dtls setup negotiation values */
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_answer_error_eos (t);
fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_no_nice_elements_request_pad)
{
struct test_webrtc *t = test_webrtc_new ();
GstPluginFeature *nicesrc, *nicesink;
GstRegistry *registry;
GstPad *pad;
/* check that the absence of libnice elements posts an error on the bus
* when requesting a pad */
registry = gst_registry_get ();
nicesrc = gst_registry_lookup_feature (registry, "nicesrc");
nicesink = gst_registry_lookup_feature (registry, "nicesink");
if (nicesrc)
gst_registry_remove_feature (registry, nicesrc);
if (nicesink)
gst_registry_remove_feature (registry, nicesink);
t->bus_message = NULL;
pad = gst_element_get_request_pad (t->webrtc1, "sink_0");
fail_unless (pad == NULL);
test_webrtc_wait_for_answer_error_eos (t);
fail_unless_equals_int (STATE_ERROR, t->state);
test_webrtc_free (t);
if (nicesrc)
gst_registry_add_feature (registry, nicesrc);
if (nicesink)
gst_registry_add_feature (registry, nicesink);
}
GST_END_TEST;
GST_START_TEST (test_no_nice_elements_state_change)
{
struct test_webrtc *t = test_webrtc_new ();
GstPluginFeature *nicesrc, *nicesink;
GstRegistry *registry;
/* check that the absence of libnice elements posts an error on the bus */
registry = gst_registry_get ();
nicesrc = gst_registry_lookup_feature (registry, "nicesrc");
nicesink = gst_registry_lookup_feature (registry, "nicesink");
if (nicesrc)
gst_registry_remove_feature (registry, nicesrc);
if (nicesink)
gst_registry_remove_feature (registry, nicesink);
t->bus_message = NULL;
gst_element_set_state (t->webrtc1, GST_STATE_READY);
test_webrtc_wait_for_answer_error_eos (t);
fail_unless_equals_int (STATE_ERROR, t->state);
test_webrtc_free (t);
if (nicesrc)
gst_registry_add_feature (registry, nicesrc);
if (nicesink)
gst_registry_add_feature (registry, nicesink);
}
GST_END_TEST;
static void
validate_rtc_stats (const GstStructure * s)
{
GstWebRTCStatsType type = 0;
double ts = 0.;
gchar *id = NULL;
fail_unless (gst_structure_get (s, "type", GST_TYPE_WEBRTC_STATS_TYPE, &type,
NULL));
fail_unless (gst_structure_get (s, "id", G_TYPE_STRING, &id, NULL));
fail_unless (gst_structure_get (s, "timestamp", G_TYPE_DOUBLE, &ts, NULL));
fail_unless (type != 0);
fail_unless (ts != 0.);
fail_unless (id != NULL);
g_free (id);
}
static void
validate_codec_stats (const GstStructure * s)
{
guint pt = 0, clock_rate = 0;
fail_unless (gst_structure_get (s, "payload-type", G_TYPE_UINT, &pt, NULL));
fail_unless (gst_structure_get (s, "clock-rate", G_TYPE_UINT, &clock_rate,
NULL));
fail_unless (pt >= 0 && pt <= 127);
fail_unless (clock_rate >= 0);
}
static void
validate_rtc_stream_stats (const GstStructure * s, const GstStructure * stats)
{
gchar *codec_id, *transport_id;
GstStructure *codec, *transport;
fail_unless (gst_structure_get (s, "codec-id", G_TYPE_STRING, &codec_id,
NULL));
fail_unless (gst_structure_get (s, "transport-id", G_TYPE_STRING,
&transport_id, NULL));
fail_unless (gst_structure_get (stats, codec_id, GST_TYPE_STRUCTURE, &codec,
NULL));
fail_unless (gst_structure_get (stats, transport_id, GST_TYPE_STRUCTURE,
&transport, NULL));
fail_unless (codec != NULL);
fail_unless (transport != NULL);
gst_structure_free (transport);
gst_structure_free (codec);
g_free (codec_id);
g_free (transport_id);
}
static void
validate_inbound_rtp_stats (const GstStructure * s, const GstStructure * stats)
{
guint ssrc, fir, pli, nack;
gint packets_lost;
guint64 packets_received, bytes_received;
double jitter;
gchar *remote_id;
GstStructure *remote;
validate_rtc_stream_stats (s, stats);
fail_unless (gst_structure_get (s, "ssrc", G_TYPE_UINT, &ssrc, NULL));
fail_unless (gst_structure_get (s, "fir-count", G_TYPE_UINT, &fir, NULL));
fail_unless (gst_structure_get (s, "pli-count", G_TYPE_UINT, &pli, NULL));
fail_unless (gst_structure_get (s, "nack-count", G_TYPE_UINT, &nack, NULL));
fail_unless (gst_structure_get (s, "packets-received", G_TYPE_UINT64,
&packets_received, NULL));
fail_unless (gst_structure_get (s, "bytes-received", G_TYPE_UINT64,
&bytes_received, NULL));
fail_unless (gst_structure_get (s, "jitter", G_TYPE_DOUBLE, &jitter, NULL));
fail_unless (gst_structure_get (s, "packets-lost", G_TYPE_INT, &packets_lost,
NULL));
fail_unless (gst_structure_get (s, "remote-id", G_TYPE_STRING, &remote_id,
NULL));
fail_unless (gst_structure_get (stats, remote_id, GST_TYPE_STRUCTURE, &remote,
NULL));
fail_unless (remote != NULL);
gst_structure_free (remote);
g_free (remote_id);
}
static void
validate_remote_inbound_rtp_stats (const GstStructure * s,
const GstStructure * stats)
{
guint ssrc;
gint packets_lost;
double jitter, rtt;
gchar *local_id;
GstStructure *local;
validate_rtc_stream_stats (s, stats);
fail_unless (gst_structure_get (s, "ssrc", G_TYPE_UINT, &ssrc, NULL));
fail_unless (gst_structure_get (s, "jitter", G_TYPE_DOUBLE, &jitter, NULL));
fail_unless (gst_structure_get (s, "packets-lost", G_TYPE_INT, &packets_lost,
NULL));
fail_unless (gst_structure_get (s, "round-trip-time", G_TYPE_DOUBLE, &rtt,
NULL));
fail_unless (gst_structure_get (s, "local-id", G_TYPE_STRING, &local_id,
NULL));
fail_unless (gst_structure_get (stats, local_id, GST_TYPE_STRUCTURE, &local,
NULL));
fail_unless (local != NULL);
gst_structure_free (local);
g_free (local_id);
}
static void
validate_outbound_rtp_stats (const GstStructure * s, const GstStructure * stats)
{
guint ssrc, fir, pli, nack;
guint64 packets_sent, bytes_sent;
gchar *remote_id;
GstStructure *remote;
validate_rtc_stream_stats (s, stats);
fail_unless (gst_structure_get (s, "ssrc", G_TYPE_UINT, &ssrc, NULL));
fail_unless (gst_structure_get (s, "fir-count", G_TYPE_UINT, &fir, NULL));
fail_unless (gst_structure_get (s, "pli-count", G_TYPE_UINT, &pli, NULL));
fail_unless (gst_structure_get (s, "nack-count", G_TYPE_UINT, &nack, NULL));
fail_unless (gst_structure_get (s, "packets-sent", G_TYPE_UINT64,
&packets_sent, NULL));
fail_unless (gst_structure_get (s, "bytes-sent", G_TYPE_UINT64, &bytes_sent,
NULL));
fail_unless (gst_structure_get (s, "remote-id", G_TYPE_STRING, &remote_id,
NULL));
fail_unless (gst_structure_get (stats, remote_id, GST_TYPE_STRUCTURE, &remote,
NULL));
fail_unless (remote != NULL);
gst_structure_free (remote);
g_free (remote_id);
}
static void
validate_remote_outbound_rtp_stats (const GstStructure * s,
const GstStructure * stats)
{
guint ssrc;
gchar *local_id;
GstStructure *local;
validate_rtc_stream_stats (s, stats);
fail_unless (gst_structure_get (s, "ssrc", G_TYPE_UINT, &ssrc, NULL));
fail_unless (gst_structure_get (s, "local-id", G_TYPE_STRING, &local_id,
NULL));
fail_unless (gst_structure_get (stats, local_id, GST_TYPE_STRUCTURE, &local,
NULL));
fail_unless (local != NULL);
gst_structure_free (local);
g_free (local_id);
}
static gboolean
validate_stats_foreach (GQuark field_id, const GValue * value,
const GstStructure * stats)
{
const gchar *field = g_quark_to_string (field_id);
GstWebRTCStatsType type;
const GstStructure *s;
fail_unless (GST_VALUE_HOLDS_STRUCTURE (value));
s = gst_value_get_structure (value);
GST_INFO ("validating field %s %" GST_PTR_FORMAT, field, s);
validate_rtc_stats (s);
gst_structure_get (s, "type", GST_TYPE_WEBRTC_STATS_TYPE, &type, NULL);
if (type == GST_WEBRTC_STATS_CODEC) {
validate_codec_stats (s);
} else if (type == GST_WEBRTC_STATS_INBOUND_RTP) {
validate_inbound_rtp_stats (s, stats);
} else if (type == GST_WEBRTC_STATS_OUTBOUND_RTP) {
validate_outbound_rtp_stats (s, stats);
} else if (type == GST_WEBRTC_STATS_REMOTE_INBOUND_RTP) {
validate_remote_inbound_rtp_stats (s, stats);
} else if (type == GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP) {
validate_remote_outbound_rtp_stats (s, stats);
} else if (type == GST_WEBRTC_STATS_CSRC) {
} else if (type == GST_WEBRTC_STATS_PEER_CONNECTION) {
} else if (type == GST_WEBRTC_STATS_DATA_CHANNEL) {
} else if (type == GST_WEBRTC_STATS_STREAM) {
} else if (type == GST_WEBRTC_STATS_TRANSPORT) {
} else if (type == GST_WEBRTC_STATS_CANDIDATE_PAIR) {
} else if (type == GST_WEBRTC_STATS_LOCAL_CANDIDATE) {
} else if (type == GST_WEBRTC_STATS_REMOTE_CANDIDATE) {
} else if (type == GST_WEBRTC_STATS_CERTIFICATE) {
} else {
g_assert_not_reached ();
}
return TRUE;
}
static void
validate_stats (const GstStructure * stats)
{
gst_structure_foreach (stats,
(GstStructureForeachFunc) validate_stats_foreach, (gpointer) stats);
}
static void
_on_stats (GstPromise * promise, gpointer user_data)
{
struct test_webrtc *t = user_data;
const GstStructure *reply = gst_promise_get_reply (promise);
int i;
validate_stats (reply);
i = GPOINTER_TO_INT (t->user_data);
i++;
t->user_data = GINT_TO_POINTER (i);
if (i >= 2)
test_webrtc_signal_state (t, STATE_CUSTOM);
gst_promise_unref (promise);
}
GST_START_TEST (test_session_stats)
{
struct test_webrtc *t = test_webrtc_new ();
GstPromise *p;
/* test that the stats generated without any streams are sane */
t->on_negotiation_needed = NULL;
t->on_offer_created = NULL;
t->on_answer_created = NULL;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_answer_error_eos (t);
fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
p = gst_promise_new_with_change_func (_on_stats, t, NULL);
g_signal_emit_by_name (t->webrtc1, "get-stats", NULL, p);
p = gst_promise_new_with_change_func (_on_stats, t, NULL);
g_signal_emit_by_name (t->webrtc2, "get-stats", NULL, p);
test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_add_transceiver)
{
struct test_webrtc *t = test_webrtc_new ();
GstWebRTCRTPTransceiverDirection direction;
GstWebRTCRTPTransceiver *trans;
direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV;
g_signal_emit_by_name (t->webrtc1, "add-transceiver", direction, NULL,
&trans);
fail_unless (trans != NULL);
fail_unless_equals_int (direction, trans->direction);
gst_object_unref (trans);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_get_transceivers)
{
struct test_webrtc *t = create_audio_test ();
GstWebRTCRTPTransceiver *trans;
GArray *transceivers;
g_signal_emit_by_name (t->webrtc1, "get-transceivers", &transceivers);
fail_unless (transceivers != NULL);
fail_unless_equals_int (1, transceivers->len);
trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 0);
fail_unless (trans != NULL);
g_array_unref (transceivers);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_add_recvonly_transceiver)
{
struct test_webrtc *t = test_webrtc_new ();
GstWebRTCRTPTransceiverDirection direction;
GstWebRTCRTPTransceiver *trans;
const gchar *expected_offer[] = { "recvonly" };
const gchar *expected_answer[] = { "sendonly" };
struct validate_sdp offer = { on_sdp_media_direction, expected_offer };
struct validate_sdp answer = { on_sdp_media_direction, expected_answer };
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_pad_added = _pad_added_fakesink;
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
/* setup recvonly transceiver */
caps = gst_caps_from_string (OPUS_RTP_CAPS (96));
direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY;
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 sendonly peer */
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
t->harnesses = g_list_prepend (t->harnesses, h);
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_answer_error_eos (t);
fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_recvonly_sendonly)
{
struct test_webrtc *t = test_webrtc_new ();
GstWebRTCRTPTransceiverDirection direction;
GstWebRTCRTPTransceiver *trans;
const gchar *expected_offer[] = { "recvonly", "sendonly" };
const gchar *expected_answer[] = { "sendonly", "recvonly" };
struct validate_sdp offer = { on_sdp_media_direction, expected_offer };
struct validate_sdp answer = { on_sdp_media_direction, expected_answer };
GstCaps *caps;
GstHarness *h;
GArray *transceivers;
/* 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_pad_added = _pad_added_fakesink;
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
/* setup recvonly transceiver */
caps = gst_caps_from_string (OPUS_RTP_CAPS (96));
direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY;
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 sendonly stream */
h = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
add_fake_audio_src_harness (h, 96);
t->harnesses = g_list_prepend (t->harnesses, h);
g_signal_emit_by_name (t->webrtc1, "get-transceivers", &transceivers);
fail_unless (transceivers != NULL);
trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 1);
trans->direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY;
g_array_unref (transceivers);
/* setup sendonly peer */
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
t->harnesses = g_list_prepend (t->harnesses, h);
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_answer_error_eos (t);
fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
test_webrtc_free (t);
}
GST_END_TEST;
static void
on_sdp_has_datachannel (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
gboolean have_data_channel = FALSE;
int i;
for (i = 0; i < gst_sdp_message_medias_len (desc->sdp); i++) {
if (_message_media_is_datachannel (desc->sdp, i)) {
/* there should only be one data channel m= section */
fail_unless_equals_int (FALSE, have_data_channel);
have_data_channel = TRUE;
}
}
fail_unless_equals_int (TRUE, have_data_channel);
}
static void
on_channel_error_not_reached (GObject * channel, GError * error,
gpointer user_data)
{
g_assert_not_reached ();
}
GST_START_TEST (test_data_channel_create)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
struct validate_sdp offer = { on_sdp_has_datachannel, NULL };
struct validate_sdp answer = { on_sdp_has_datachannel, NULL };
gchar *label;
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
g_assert_nonnull (channel);
g_object_get (channel, "label", &label, NULL);
g_assert_cmpstr (label, ==, "label");
g_signal_connect (channel, "on-error",
G_CALLBACK (on_channel_error_not_reached), NULL);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_answer_error_eos (t);
fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
g_object_unref (channel);
g_free (label);
test_webrtc_free (t);
}
GST_END_TEST;
static void
have_data_channel (struct test_webrtc *t, GstElement * element,
GObject * our, gpointer user_data)
{
GObject *other = user_data;
gchar *our_label, *other_label;
g_signal_connect (our, "on-error", G_CALLBACK (on_channel_error_not_reached),
NULL);
g_object_get (our, "label", &our_label, NULL);
g_object_get (other, "label", &other_label, NULL);
g_assert_cmpstr (our_label, ==, other_label);
g_free (our_label);
g_free (other_label);
test_webrtc_signal_state_unlocked (t, STATE_CUSTOM);
}
GST_START_TEST (test_data_channel_remote_notify)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
struct validate_sdp offer = { on_sdp_has_datachannel, NULL };
struct validate_sdp answer = { on_sdp_has_datachannel, NULL };
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
t->on_data_channel = have_data_channel;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
g_assert_nonnull (channel);
t->data_channel_data = channel;
g_signal_connect (channel, "on-error",
G_CALLBACK (on_channel_error_not_reached), NULL);
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);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
g_object_unref (channel);
test_webrtc_free (t);
}
GST_END_TEST;
static const gchar *test_string = "GStreamer WebRTC is awesome!";
static void
on_message_string (GObject * channel, const gchar * str, struct test_webrtc *t)
{
gchar *expected = g_object_steal_data (channel, "expected");
g_assert_cmpstr (expected, ==, str);
g_free (expected);
test_webrtc_signal_state (t, STATE_CUSTOM);
}
static void
have_data_channel_transfer_string (struct test_webrtc *t, GstElement * element,
GObject * our, gpointer user_data)
{
GObject *other = user_data;
GstWebRTCDataChannelState state;
g_object_get (our, "ready-state", &state, NULL);
fail_unless_equals_int (GST_WEBRTC_DATA_CHANNEL_STATE_OPEN, state);
g_object_get (other, "ready-state", &state, NULL);
fail_unless_equals_int (GST_WEBRTC_DATA_CHANNEL_STATE_OPEN, state);
g_object_set_data_full (our, "expected", g_strdup (test_string), g_free);
g_signal_connect (our, "on-message-string", G_CALLBACK (on_message_string),
t);
g_signal_connect (other, "on-error",
G_CALLBACK (on_channel_error_not_reached), NULL);
g_signal_emit_by_name (other, "send-string", test_string);
}
GST_START_TEST (test_data_channel_transfer_string)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
struct validate_sdp offer = { on_sdp_has_datachannel, NULL };
struct validate_sdp answer = { on_sdp_has_datachannel, NULL };
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
t->on_data_channel = have_data_channel_transfer_string;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
g_assert_nonnull (channel);
t->data_channel_data = channel;
g_signal_connect (channel, "on-error",
G_CALLBACK (on_channel_error_not_reached), NULL);
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);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
g_object_unref (channel);
test_webrtc_free (t);
}
GST_END_TEST;
#define g_assert_cmpbytes(b1, b2) \
G_STMT_START { \
gsize l1, l2; \
const guint8 *d1 = g_bytes_get_data (b1, &l1); \
const guint8 *d2 = g_bytes_get_data (b2, &l2); \
g_assert_cmpmem (d1, l1, d2, l2); \
} G_STMT_END;
static void
on_message_data (GObject * channel, GBytes * data, struct test_webrtc *t)
{
GBytes *expected = g_object_steal_data (channel, "expected");
g_assert_cmpbytes (data, expected);
g_bytes_unref (expected);
test_webrtc_signal_state (t, STATE_CUSTOM);
}
static void
have_data_channel_transfer_data (struct test_webrtc *t, GstElement * element,
GObject * our, gpointer user_data)
{
GObject *other = user_data;
GBytes *data = g_bytes_new_static (test_string, strlen (test_string));
GstWebRTCDataChannelState state;
g_object_get (our, "ready-state", &state, NULL);
fail_unless_equals_int (GST_WEBRTC_DATA_CHANNEL_STATE_OPEN, state);
g_object_get (other, "ready-state", &state, NULL);
fail_unless_equals_int (GST_WEBRTC_DATA_CHANNEL_STATE_OPEN, state);
g_object_set_data_full (our, "expected", g_bytes_ref (data),
(GDestroyNotify) g_bytes_unref);
g_signal_connect (our, "on-message-data", G_CALLBACK (on_message_data), t);
g_signal_connect (other, "on-error",
G_CALLBACK (on_channel_error_not_reached), NULL);
g_signal_emit_by_name (other, "send-data", data);
}
GST_START_TEST (test_data_channel_transfer_data)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
struct validate_sdp offer = { on_sdp_has_datachannel, NULL };
struct validate_sdp answer = { on_sdp_has_datachannel, NULL };
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
t->on_data_channel = have_data_channel_transfer_data;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
g_assert_nonnull (channel);
t->data_channel_data = channel;
g_signal_connect (channel, "on-error",
G_CALLBACK (on_channel_error_not_reached), NULL);
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);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
g_object_unref (channel);
test_webrtc_free (t);
}
GST_END_TEST;
static void
have_data_channel_create_data_channel (struct test_webrtc *t,
GstElement * element, GObject * our, gpointer user_data)
{
GObject *another;
t->on_data_channel = have_data_channel_transfer_string;
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&another);
g_assert_nonnull (another);
t->data_channel_data = another;
g_signal_connect (another, "on-error",
G_CALLBACK (on_channel_error_not_reached), NULL);
}
GST_START_TEST (test_data_channel_create_after_negotiate)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
struct validate_sdp offer = { on_sdp_has_datachannel, NULL };
struct validate_sdp answer = { on_sdp_has_datachannel, NULL };
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
t->on_data_channel = have_data_channel_create_data_channel;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "prev-label", NULL,
&channel);
g_assert_nonnull (channel);
t->data_channel_data = channel;
g_signal_connect (channel, "on-error",
G_CALLBACK (on_channel_error_not_reached), NULL);
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);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
g_object_unref (channel);
test_webrtc_free (t);
}
GST_END_TEST;
static void
on_buffered_amount_low_emitted (GObject * channel, struct test_webrtc *t)
{
test_webrtc_signal_state (t, STATE_CUSTOM);
}
static void
have_data_channel_check_low_threshold_emitted (struct test_webrtc *t,
GstElement * element, GObject * our, gpointer user_data)
{
g_signal_connect (our, "on-buffered-amount-low",
G_CALLBACK (on_buffered_amount_low_emitted), t);
g_object_set (our, "buffered-amount-low-threshold", 1, NULL);
g_signal_connect (our, "on-error", G_CALLBACK (on_channel_error_not_reached),
NULL);
g_signal_emit_by_name (our, "send-string", "DATA");
}
GST_START_TEST (test_data_channel_low_threshold)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
struct validate_sdp offer = { on_sdp_has_datachannel, NULL };
struct validate_sdp answer = { on_sdp_has_datachannel, NULL };
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
t->on_data_channel = have_data_channel_check_low_threshold_emitted;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
g_assert_nonnull (channel);
t->data_channel_data = channel;
g_signal_connect (channel, "on-error",
G_CALLBACK (on_channel_error_not_reached), NULL);
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);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
g_object_unref (channel);
test_webrtc_free (t);
}
GST_END_TEST;
static void
on_channel_error (GObject * channel, GError * error, struct test_webrtc *t)
{
g_assert_nonnull (error);
test_webrtc_signal_state (t, STATE_CUSTOM);
}
static void
have_data_channel_transfer_large_data (struct test_webrtc *t,
GstElement * element, GObject * our, gpointer user_data)
{
GObject *other = user_data;
const gsize size = 1024 * 1024;
guint8 *random_data = g_new (guint8, size);
GBytes *data;
gsize i;
for (i = 0; i < size; i++)
random_data[i] = (guint8) (i & 0xff);
data = g_bytes_new_static (random_data, size);
g_object_set_data_full (our, "expected", g_bytes_ref (data),
(GDestroyNotify) g_bytes_unref);
g_signal_connect (our, "on-message-data", G_CALLBACK (on_message_data), t);
g_signal_connect (other, "on-error", G_CALLBACK (on_channel_error), t);
g_signal_emit_by_name (other, "send-data", data);
}
GST_START_TEST (test_data_channel_max_message_size)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
struct validate_sdp offer = { on_sdp_has_datachannel, NULL };
struct validate_sdp answer = { on_sdp_has_datachannel, NULL };
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
t->on_data_channel = have_data_channel_transfer_large_data;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
g_assert_nonnull (channel);
t->data_channel_data = channel;
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);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
g_object_unref (channel);
test_webrtc_free (t);
}
GST_END_TEST;
static void
_on_ready_state_notify (GObject * channel, GParamSpec * pspec,
struct test_webrtc *t)
{
gint *n_ready = t->data_channel_data;
GstWebRTCDataChannelState ready_state;
g_object_get (channel, "ready-state", &ready_state, NULL);
if (ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_OPEN) {
if (++(*n_ready) >= 2)
test_webrtc_signal_state (t, STATE_CUSTOM);
}
}
GST_START_TEST (test_data_channel_pre_negotiated)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel1 = NULL, *channel2 = NULL;
struct validate_sdp offer = { on_sdp_has_datachannel, NULL };
struct validate_sdp answer = { on_sdp_has_datachannel, NULL };
GstStructure *s;
gint n_ready = 0;
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
s = gst_structure_new ("application/data-channel", "negotiated",
G_TYPE_BOOLEAN, TRUE, "id", G_TYPE_INT, 1, NULL);
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", s,
&channel1);
g_assert_nonnull (channel1);
g_signal_emit_by_name (t->webrtc2, "create-data-channel", "label", s,
&channel2);
g_assert_nonnull (channel2);
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);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_answer_error_eos (t);
fail_unless (t->state == STATE_ANSWER_CREATED);
t->data_channel_data = &n_ready;
g_signal_connect (channel1, "notify::ready-state",
G_CALLBACK (_on_ready_state_notify), t);
g_signal_connect (channel2, "notify::ready-state",
G_CALLBACK (_on_ready_state_notify), t);
test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
test_webrtc_signal_state (t, STATE_NEW);
have_data_channel_transfer_string (t, t->webrtc1, channel1, channel2);
test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
g_object_unref (channel1);
g_object_unref (channel2);
gst_structure_free (s);
test_webrtc_free (t);
}
GST_END_TEST;
typedef struct
{
guint num_media;
guint num_active_media;
const gchar **bundled;
const gchar **bundled_only;
} BundleCheckData;
static void
_check_bundled_sdp_media (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * sd, gpointer user_data)
{
gchar **bundled = NULL;
BundleCheckData *data = (BundleCheckData *) user_data;
guint i;
guint active_media;
fail_unless_equals_int (gst_sdp_message_medias_len (sd->sdp),
data->num_media);
fail_unless (_parse_bundle (sd->sdp, &bundled));
if (!bundled) {
fail_unless_equals_int (g_strv_length ((GStrv) data->bundled), 0);
} else {
fail_unless_equals_int (g_strv_length (bundled),
g_strv_length ((GStrv) data->bundled));
}
for (i = 0; data->bundled[i]; i++) {
fail_unless (g_strv_contains ((const gchar **) bundled, data->bundled[i]));
}
active_media = 0;
for (i = 0; i < gst_sdp_message_medias_len (sd->sdp); i++) {
const GstSDPMedia *media = gst_sdp_message_get_media (sd->sdp, i);
const gchar *mid = gst_sdp_media_get_attribute_val (media, "mid");
if (g_strv_contains ((const gchar **) data->bundled_only, mid))
fail_unless (_media_has_attribute_key (media, "bundle-only"));
if (gst_sdp_media_get_port (media) != 0)
active_media += 1;
}
fail_unless_equals_int (active_media, data->num_active_media);
if (bundled)
g_strfreev (bundled);
}
GST_START_TEST (test_bundle_audio_video_max_bundle_max_bundle)
{
struct test_webrtc *t = create_audio_video_test ();
const gchar *bundle[] = { "audio0", "video1", NULL };
const gchar *offer_bundle_only[] = { "video1", NULL };
const gchar *answer_bundle_only[] = { NULL };
/* We set a max-bundle policy on the offering webrtcbin,
* this means that all the offered medias should be part
* of the group:BUNDLE attribute, and they should be marked
* as bundle-only
*/
BundleCheckData offer_data = {
2,
1,
bundle,
offer_bundle_only,
};
/* We also set a max-bundle policy on the answering webrtcbin,
* this means that all the offered medias should be part
* of the group:BUNDLE attribute, but need not be marked
* as bundle-only.
*/
BundleCheckData answer_data = {
2,
2,
bundle,
answer_bundle_only,
};
struct validate_sdp offer = { _check_bundled_sdp_media, &offer_data };
struct validate_sdp answer = { _check_bundled_sdp_media, &answer_data };
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");
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_answer_error_eos (t);
fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_bundle_audio_video_max_compat_max_bundle)
{
struct test_webrtc *t = create_audio_video_test ();
const gchar *bundle[] = { "audio0", "video1", NULL };
const gchar *bundle_only[] = { NULL };
/* We set a max-compat policy on the offering webrtcbin,
* this means that all the offered medias should be part
* of the group:BUNDLE attribute, and they should *not* be marked
* as bundle-only
*/
BundleCheckData offer_data = {
2,
2,
bundle,
bundle_only,
};
/* We set a max-bundle policy on the answering webrtcbin,
* this means that all the offered medias should be part
* of the group:BUNDLE attribute, but need not be marked
* as bundle-only.
*/
BundleCheckData answer_data = {
2,
2,
bundle,
bundle_only,
};
struct validate_sdp offer = { _check_bundled_sdp_media, &offer_data };
struct validate_sdp answer = { _check_bundled_sdp_media, &answer_data };
gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
"max-compat");
gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy",
"max-bundle");
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_answer_error_eos (t);
fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_bundle_audio_video_max_bundle_none)
{
struct test_webrtc *t = create_audio_video_test ();
const gchar *offer_bundle[] = { "audio0", "video1", NULL };
const gchar *offer_bundle_only[] = { "video1", NULL };
const gchar *answer_bundle[] = { NULL };
const gchar *answer_bundle_only[] = { NULL };
/* We set a max-bundle policy on the offering webrtcbin,
* this means that all the offered medias should be part
* of the group:BUNDLE attribute, and they should be marked
* as bundle-only
*/
BundleCheckData offer_data = {
2,
1,
offer_bundle,
offer_bundle_only,
};
/* We set a none policy on the answering webrtcbin,
* this means that the answer should contain no bundled
* medias, and as the bundle-policy of the offering webrtcbin
* is set to max-bundle, only one media should be active.
*/
BundleCheckData answer_data = {
2,
1,
answer_bundle,
answer_bundle_only,
};
struct validate_sdp offer = { _check_bundled_sdp_media, &offer_data };
struct validate_sdp answer = { _check_bundled_sdp_media, &answer_data };
gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
"max-bundle");
gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy", "none");
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_answer_error_eos (t);
fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_bundle_audio_video_data)
{
struct test_webrtc *t = create_audio_video_test ();
const gchar *bundle[] = { "audio0", "video1", "application2", NULL };
const gchar *offer_bundle_only[] = { "video1", "application2", NULL };
const gchar *answer_bundle_only[] = { NULL };
GObject *channel = NULL;
/* We set a max-bundle policy on the offering webrtcbin,
* this means that all the offered medias should be part
* of the group:BUNDLE attribute, and they should be marked
* as bundle-only
*/
BundleCheckData offer_data = {
3,
1,
bundle,
offer_bundle_only,
};
/* We also set a max-bundle policy on the answering webrtcbin,
* this means that all the offered medias should be part
* of the group:BUNDLE attribute, but need not be marked
* as bundle-only.
*/
BundleCheckData answer_data = {
3,
3,
bundle,
answer_bundle_only,
};
struct validate_sdp offer = { _check_bundled_sdp_media, &offer_data };
struct validate_sdp answer = { _check_bundled_sdp_media, &answer_data };
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");
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = _check_validate_sdp;
t->answer_data = &answer;
t->on_answer_created = _check_validate_sdp;
t->on_ice_candidate = NULL;
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
test_webrtc_create_offer (t, t->webrtc1);
test_webrtc_wait_for_answer_error_eos (t);
fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
g_object_unref (channel);
test_webrtc_free (t);
}
GST_END_TEST;
static Suite *
webrtcbin_suite (void)
{
Suite *s = suite_create ("webrtcbin");
TCase *tc = tcase_create ("general");
GstPluginFeature *nicesrc, *nicesink, *dtlssrtpdec, *dtlssrtpenc;
GstPluginFeature *sctpenc, *sctpdec;
GstRegistry *registry;
registry = gst_registry_get ();
nicesrc = gst_registry_lookup_feature (registry, "nicesrc");
nicesink = gst_registry_lookup_feature (registry, "nicesink");
dtlssrtpenc = gst_registry_lookup_feature (registry, "dtlssrtpenc");
dtlssrtpdec = gst_registry_lookup_feature (registry, "dtlssrtpdec");
sctpenc = gst_registry_lookup_feature (registry, "sctpenc");
sctpdec = gst_registry_lookup_feature (registry, "sctpdec");
tcase_add_test (tc, test_no_nice_elements_request_pad);
tcase_add_test (tc, test_no_nice_elements_state_change);
if (nicesrc && nicesink && dtlssrtpenc && dtlssrtpdec) {
tcase_add_test (tc, test_sdp_no_media);
tcase_add_test (tc, test_session_stats);
tcase_add_test (tc, test_audio);
tcase_add_test (tc, test_audio_video);
tcase_add_test (tc, test_media_direction);
tcase_add_test (tc, test_media_setup);
tcase_add_test (tc, test_add_transceiver);
tcase_add_test (tc, test_get_transceivers);
tcase_add_test (tc, test_add_recvonly_transceiver);
tcase_add_test (tc, test_recvonly_sendonly);
tcase_add_test (tc, test_payload_types);
tcase_add_test (tc, test_bundle_audio_video_max_bundle_max_bundle);
tcase_add_test (tc, test_bundle_audio_video_max_bundle_none);
tcase_add_test (tc, test_bundle_audio_video_max_compat_max_bundle);
if (sctpenc && sctpdec) {
tcase_add_test (tc, test_data_channel_create);
tcase_add_test (tc, test_data_channel_remote_notify);
tcase_add_test (tc, test_data_channel_transfer_string);
tcase_add_test (tc, test_data_channel_transfer_data);
tcase_add_test (tc, test_data_channel_create_after_negotiate);
tcase_add_test (tc, test_data_channel_low_threshold);
tcase_add_test (tc, test_data_channel_max_message_size);
tcase_add_test (tc, test_data_channel_pre_negotiated);
tcase_add_test (tc, test_bundle_audio_video_data);
} else {
GST_WARNING ("Some required elements were not found. "
"All datachannel are disabled. sctpenc %p, sctpdec %p", sctpenc,
sctpdec);
}
} else {
GST_WARNING ("Some required elements were not found. "
"All media tests are disabled. nicesrc %p, nicesink %p, "
"dtlssrtpenc %p, dtlssrtpdec %p", nicesrc, nicesink, dtlssrtpenc,
dtlssrtpdec);
}
if (nicesrc)
gst_object_unref (nicesrc);
if (nicesink)
gst_object_unref (nicesink);
if (dtlssrtpdec)
gst_object_unref (dtlssrtpdec);
if (dtlssrtpenc)
gst_object_unref (dtlssrtpenc);
if (sctpenc)
gst_object_unref (sctpenc);
if (sctpdec)
gst_object_unref (sctpdec);
suite_add_tcase (s, tc);
return s;
}
GST_CHECK_MAIN (webrtcbin);
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