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gstreamer/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
Philippe Normand 299a000917 webrtcbin: Allow session level setup attribute in SDP 
An SDP answer can declare its setup attribute at the session level or at the
media level. Until this patch we were validating only the latter case and an
assert was raised in the former case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6930>
2024-05-27 14:21:15 +00:00

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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 <gst/rtp/rtp.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)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)
#define TEST_GET_ANSWERER(t) (TEST_IS_OFFER_ELEMENT(t, t->webrtc1) ? (t)->webrtc2 : t->webrtc1)
#define TEST_SDP_IS_LOCAL(t, e, d) ((TEST_IS_OFFER_ELEMENT (t, e) ^ ((d)->type == GST_WEBRTC_SDP_TYPE_OFFER)) == 0)
typedef enum
{
STATE_NEW = 1,
STATE_NEGOTIATION_NEEDED,
STATE_OFFER_CREATED,
STATE_OFFER_SET,
STATE_ANSWER_CREATED,
STATE_ANSWER_SET,
STATE_EOS,
STATE_ERROR,
STATE_CUSTOM,
} TestState;
struct test_webrtc;
typedef void (*OnPadAdded) (struct test_webrtc * t, GstElement * element,
GstPad * pad, gpointer user_data);
/* basic premise of this is that webrtc1 and webrtc2 are attempting to connect
* to each other in various configurations */
struct test_webrtc
{
GList *harnesses;
GstTestClock *test_clock;
GThread *thread;
GMainLoop *loop;
GstBus *bus1;
GstBus *bus2;
GstElement *webrtc1;
GstElement *webrtc2;
GMutex lock;
GCond cond;
GArray *states;
guint offerror;
gulong error_signal_handler_id;
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;
void (*on_offer_created) (struct test_webrtc * t,
GstElement * element,
GstPromise * promise,
gpointer user_data);
GstWebRTCSessionDescription *offer_desc;
guint offer_set_count;
gpointer offer_data;
GDestroyNotify offer_notify;
void (*on_offer_set) (struct test_webrtc * t,
GstElement * element,
GstPromise * promise,
gpointer user_data);
gpointer offer_set_data;
GDestroyNotify offer_set_notify;
void (*on_answer_created) (struct test_webrtc * t,
GstElement * element,
GstPromise * promise,
gpointer user_data);
GstWebRTCSessionDescription *answer_desc;
guint answer_set_count;
gpointer answer_data;
GDestroyNotify answer_notify;
void (*on_answer_set) (struct test_webrtc * t,
GstElement * element,
GstPromise * promise,
gpointer user_data);
gpointer answer_set_data;
GDestroyNotify answer_set_notify;
void (*on_prepare_data_channel) (struct test_webrtc * t,
GstElement * element,
GObject * data_channel,
gboolean is_local,
gpointer user_data);
void (*on_data_channel) (struct test_webrtc * t,
GstElement * element,
GObject *data_channel,
gpointer user_data);
gpointer data_channel_data;
GDestroyNotify data_channel_notify;
OnPadAdded on_pad_added;
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
test_webrtc_signal_state_unlocked (struct test_webrtc *t, TestState state)
{
GST_TRACE ("signal state 0x%x", state);
g_array_append_val (t->states, 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 gboolean
test_webrtc_state_find_unlocked (struct test_webrtc *t, TestState state,
guint * idx)
{
guint i;
for (i = 0; i < t->states->len; i++) {
if (state == g_array_index (t->states, TestState, i))
return TRUE;
}
return FALSE;
}
#endif
static void
_on_answer_set (GstPromise * promise, gpointer user_data)
{
struct test_webrtc *t = user_data;
GstElement *answerer = TEST_GET_ANSWERER (t);
g_mutex_lock (&t->lock);
if (++t->answer_set_count >= 2) {
if (t->on_answer_set)
t->on_answer_set (t, answerer, promise, t->answer_set_data);
test_webrtc_signal_state_unlocked (t, STATE_ANSWER_SET);
g_cond_broadcast (&t->cond);
}
gst_promise_unref (promise);
g_mutex_unlock (&t->lock);
}
static void
_on_answer_received (GstPromise * promise, gpointer user_data)
{
struct test_webrtc *t = user_data;
GstElement *offeror = TEST_GET_OFFEROR (t);
GstElement *answerer = TEST_GET_ANSWERER (t);
const GstStructure *reply;
GstWebRTCSessionDescription *answer = NULL;
GError *error = NULL;
reply = gst_promise_get_reply (promise);
if (gst_structure_get (reply, "answer",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &answer, NULL)) {
gchar *desc = gst_sdp_message_as_text (answer->sdp);
GST_INFO ("Created Answer: %s", desc);
g_free (desc);
} else if (gst_structure_get (reply, "error", G_TYPE_ERROR, &error, NULL)) {
GST_INFO ("Creating answer resulted in error: %s", error->message);
} else {
g_assert_not_reached ();
}
g_mutex_lock (&t->lock);
g_assert (t->answer_desc == NULL);
t->answer_desc = answer;
if (t->on_answer_created) {
t->on_answer_created (t, answerer, promise, t->answer_data);
}
gst_promise_unref (promise);
if (error)
goto error;
if (t->answer_desc) {
promise = gst_promise_new_with_change_func (_on_answer_set, t, NULL);
g_signal_emit_by_name (answerer, "set-local-description", t->answer_desc,
promise);
promise = gst_promise_new_with_change_func (_on_answer_set, t, NULL);
g_signal_emit_by_name (offeror, "set-remote-description", t->answer_desc,
promise);
}
test_webrtc_signal_state_unlocked (t, STATE_ANSWER_CREATED);
g_mutex_unlock (&t->lock);
return;
error:
g_clear_error (&error);
test_webrtc_signal_state_unlocked (t, STATE_ERROR);
g_mutex_unlock (&t->lock);
return;
}
static void
_on_offer_set (GstPromise * promise, gpointer user_data)
{
struct test_webrtc *t = user_data;
GstElement *offeror = TEST_GET_OFFEROR (t);
g_mutex_lock (&t->lock);
if (++t->offer_set_count >= 2) {
if (t->on_offer_set)
t->on_offer_set (t, offeror, promise, t->offer_set_data);
test_webrtc_signal_state_unlocked (t, STATE_OFFER_SET);
g_cond_broadcast (&t->cond);
}
gst_promise_unref (promise);
g_mutex_unlock (&t->lock);
}
static void
_on_offer_received (GstPromise * promise, gpointer user_data)
{
struct test_webrtc *t = user_data;
GstElement *offeror = TEST_GET_OFFEROR (t);
GstElement *answerer = TEST_GET_ANSWERER (t);
const GstStructure *reply;
GstWebRTCSessionDescription *offer = NULL;
GError *error = NULL;
reply = gst_promise_get_reply (promise);
if (gst_structure_get (reply, "offer",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL)) {
gchar *desc = gst_sdp_message_as_text (offer->sdp);
GST_INFO ("Created offer: %s", desc);
g_free (desc);
} else if (gst_structure_get (reply, "error", G_TYPE_ERROR, &error, NULL)) {
GST_INFO ("Creating offer resulted in error: %s", error->message);
} else {
g_assert_not_reached ();
}
g_mutex_lock (&t->lock);
g_assert (t->offer_desc == NULL);
t->offer_desc = offer;
if (t->on_offer_created) {
t->on_offer_created (t, offeror, promise, t->offer_data);
}
gst_promise_unref (promise);
if (error)
goto error;
test_webrtc_signal_state_unlocked (t, STATE_OFFER_CREATED);
gst_object_ref (offeror);
gst_object_ref (answerer);
g_mutex_unlock (&t->lock);
if (t->offer_desc) {
promise = gst_promise_new_with_change_func (_on_offer_set, t, NULL);
g_signal_emit_by_name (offeror, "set-local-description", t->offer_desc,
promise);
promise = gst_promise_new_with_change_func (_on_offer_set, t, NULL);
g_signal_emit_by_name (answerer, "set-remote-description", t->offer_desc,
promise);
promise = gst_promise_new_with_change_func (_on_answer_received, t, NULL);
g_signal_emit_by_name (answerer, "create-answer", NULL, promise);
}
gst_clear_object (&offeror);
gst_clear_object (&answerer);
return;
error:
g_clear_error (&error);
test_webrtc_signal_state_unlocked (t, STATE_ERROR);
g_mutex_unlock (&t->lock);
return;
}
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 (GST_OBJECT_NAME (msg->src), "-state_changed-",
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 (GST_OBJECT_NAME (t->webrtc1), "-error", 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 (GST_OBJECT_NAME (t->webrtc2), "-error", 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",
GST_OBJECT_NAME (msg->src), err->message);
GST_WARNING ("Debugging info: %s", (dbg_info) ? dbg_info : "none");
g_error_free (err);
g_free (dbg_info);
test_webrtc_signal_state_unlocked (t, STATE_ERROR);
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");
test_webrtc_signal_state_unlocked (t, STATE_EOS);
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_channel_error_not_reached (GObject * channel, GError * error,
gpointer user_data)
{
g_assert_not_reached ();
}
static void on_message_string (GObject * channel, const gchar * str,
struct test_webrtc *t);
static void on_message_data (GObject * channel, GBytes * data,
struct test_webrtc *t);
static void
have_prepare_data_channel (struct test_webrtc *t,
GstElement * element,
GObject * data_channel, gboolean is_local, gpointer user_data)
{
t->error_signal_handler_id =
g_signal_connect (data_channel, "on-error",
G_CALLBACK (on_channel_error_not_reached), NULL);
g_signal_connect (data_channel, "on-message-string",
G_CALLBACK (on_message_string), t);
g_signal_connect (data_channel, "on-message-data",
G_CALLBACK (on_message_data), t);
}
static void
_on_prepare_data_channel (GstElement * webrtc, GObject * data_channel,
gboolean is_local, struct test_webrtc *t)
{
/* We can't lock the test_webrtc mutex here because this callback might be
* called from an already locked _on_data_channel thread. This is the case for
* the test_data_channel_create_after_negotiate test. */
if (t->on_prepare_data_channel)
t->on_prepare_data_channel (t, webrtc, data_channel, is_local,
t->data_channel_data);
}
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);
test_webrtc_signal_state_unlocked (t, STATE_NEGOTIATION_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:{
GError *err = NULL;
gchar *dbg = NULL;
gst_message_parse_error (msg, &err, &dbg);
g_error ("ERROR from element %s: %s (Debugging info: %s)",
GST_OBJECT_NAME (msg->src), err->message, (dbg) ? dbg : "none");
g_error_free (err);
g_free (dbg);
g_assert_not_reached ();
break;
}
default:
break;
}
}
static void
_offer_answer_not_reached (struct test_webrtc *t, GstElement * element,
GstPromise * promise, gpointer user_data)
{
g_assert_not_reached ();
}
static void
_on_prepare_data_channel_not_reached (struct test_webrtc *t,
GstElement * element, GObject * data_channel, gboolean is_local,
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_prepare_data_channel = _on_prepare_data_channel_not_reached;
ret->on_data_channel = _on_data_channel_not_reached;
ret->bus_message = _bus_no_errors;
ret->offerror = 1;
ret->error_signal_handler_id = -1;
g_mutex_init (&ret->lock);
g_cond_init (&ret->cond);
ret->states = g_array_new (FALSE, TRUE, sizeof (TestState));
ret->test_clock = GST_TEST_CLOCK (gst_test_clock_new ());
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);
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_clock (ret->webrtc1, GST_CLOCK (ret->test_clock));
gst_element_set_clock (ret->webrtc2, GST_CLOCK (ret->test_clock));
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, "prepare-data-channel",
G_CALLBACK (_on_prepare_data_channel), ret);
g_signal_connect (ret->webrtc2, "prepare-data-channel",
G_CALLBACK (_on_prepare_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);
return ret;
}
static void
test_webrtc_reset_negotiation (struct test_webrtc *t)
{
GST_DEBUG ("resetting negotiation");
if (t->offer_desc)
gst_webrtc_session_description_free (t->offer_desc);
t->offer_desc = NULL;
t->offer_set_count = 0;
if (t->answer_desc)
gst_webrtc_session_description_free (t->answer_desc);
t->answer_desc = NULL;
t->answer_set_count = 0;
test_webrtc_signal_state (t, STATE_NEGOTIATION_NEEDED);
}
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);
g_object_unref (t->test_clock);
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->offer_set_notify)
t->offer_set_notify (t->offer_set_data);
if (t->answer_notify)
t->answer_notify (t->answer_data);
if (t->answer_set_notify)
t->answer_set_notify (t->answer_set_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));
test_webrtc_reset_negotiation (t);
gst_object_unref (t->webrtc1);
gst_object_unref (t->webrtc2);
g_mutex_clear (&t->lock);
g_cond_clear (&t->cond);
g_array_free (t->states, TRUE);
t->states = NULL;
g_free (t);
}
static void
test_webrtc_create_offer (struct test_webrtc *t)
{
GstPromise *promise;
GstElement *offeror = TEST_GET_OFFEROR (t);
promise = gst_promise_new_with_change_func (_on_offer_received, t, NULL);
g_signal_emit_by_name (offeror, "create-offer", NULL, promise);
}
static TestState
test_webrtc_check_for_state_mask_unlocked (struct test_webrtc *t,
TestState state)
{
guint i;
GST_LOG ("attempting to check for state mask 0x%x", state);
for (i = 0; i < t->states->len; i++) {
TestState val = g_array_index (t->states, TestState, i);
if (((1 << val) & state) != 0) {
GST_DEBUG ("found state 0x%x in wait mask 0x%x at idx %u", val, state, i);
g_array_remove_range (t->states, 0, i + 1);
return val;
}
}
return 0;
}
static TestState
test_webrtc_check_for_state_mask (struct test_webrtc *t, TestState state)
{
TestState ret;
g_mutex_lock (&t->lock);
ret = test_webrtc_check_for_state_mask_unlocked (t, state);
g_mutex_unlock (&t->lock);
return ret;
}
static TestState
test_webrtc_wait_for_state_mask (struct test_webrtc *t, TestState state)
{
TestState ret = 0;
g_mutex_lock (&t->lock);
GST_LOG ("attempting to wait for state mask 0x%x", state);
while (TRUE) {
ret = test_webrtc_check_for_state_mask_unlocked (t, state);
if (ret)
break;
g_cond_wait (&t->cond, &t->lock);
}
g_mutex_unlock (&t->lock);
return ret;
}
static TestState
test_webrtc_wait_for_answer_error_eos (struct test_webrtc *t)
{
TestState states = 0;
states |= (1 << STATE_ANSWER_SET);
states |= (1 << STATE_EOS);
states |= (1 << STATE_ERROR);
return test_webrtc_wait_for_state_mask (t, states);
}
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);
}
#if 0
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 void
on_negotiation_needed_hit (struct test_webrtc *t, GstElement * element,
gpointer user_data)
{
guint *flag = (guint *) user_data;
*flag |= 1 << ((element == t->webrtc1) ? 1 : 2);
}
typedef void (*ValidateSDPFunc) (struct test_webrtc * t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data);
struct validate_sdp;
struct validate_sdp
{
ValidateSDPFunc validate;
gpointer user_data;
struct validate_sdp *next;
};
#define VAL_SDP_INIT(name,func,data,next) \
struct validate_sdp name = { func, data, next }
static void
_check_validate_sdp (struct test_webrtc *t, GstElement * element,
GstPromise * promise, gpointer user_data)
{
struct validate_sdp *validate = user_data;
GstWebRTCSessionDescription *desc = NULL;
if (TEST_IS_OFFER_ELEMENT (t, element))
desc = t->offer_desc;
else
desc = t->answer_desc;
while (validate) {
validate->validate (t, element, desc, validate->user_data);
validate = validate->next;
}
}
static void
test_validate_sdp_full (struct test_webrtc *t, struct validate_sdp *offer,
struct validate_sdp *answer, TestState wait_mask,
gboolean perform_state_change)
{
if (offer) {
t->offer_data = offer;
t->on_offer_created = _check_validate_sdp;
} else {
t->offer_data = NULL;
t->on_offer_created = NULL;
}
if (answer) {
t->answer_data = answer;
t->on_answer_created = _check_validate_sdp;
} else {
t->answer_data = NULL;
t->on_answer_created = NULL;
}
if (perform_state_change) {
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);
if (wait_mask == 0) {
fail_unless_equals_int (test_webrtc_wait_for_answer_error_eos (t),
STATE_ANSWER_SET);
} else {
test_webrtc_wait_for_state_mask (t, wait_mask);
}
}
static void
test_validate_sdp (struct test_webrtc *t, struct validate_sdp *offer,
struct validate_sdp *answer)
{
test_validate_sdp_full (t, offer, answer, 0, TRUE);
}
static void
_count_num_sdp_media (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
guint expected = GPOINTER_TO_UINT (user_data);
fail_unless_equals_int (gst_sdp_message_medias_len (desc->sdp), expected);
}
GST_START_TEST (test_sdp_no_media)
{
struct test_webrtc *t = test_webrtc_new ();
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (0), NULL);
/* check that a no stream connection creates 0 media sections */
t->on_negotiation_needed = NULL;
test_validate_sdp (t, &count, &count);
test_webrtc_free (t);
}
GST_END_TEST;
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);
if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0
|| g_strcmp0 (gst_sdp_media_get_media (media), "video") == 0) {
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", i);
have_direction = TRUE;
fail_unless_equals_string (attr->key, expected_directions[i]);
} else if (g_strcmp0 (attr->key, "sendonly") == 0) {
fail_unless (have_direction == FALSE,
"duplicate/multiple directions for media %u", i);
have_direction = TRUE;
fail_unless_equals_string (attr->key, expected_directions[i]);
} else if (g_strcmp0 (attr->key, "recvonly") == 0) {
fail_unless (have_direction == FALSE,
"duplicate/multiple directions for media %u", i);
have_direction = TRUE;
fail_unless_equals_string (attr->key, expected_directions[i]);
} else if (g_strcmp0 (attr->key, "sendrecv") == 0) {
fail_unless (have_direction == FALSE,
"duplicate/multiple directions for media %u", i);
have_direction = TRUE;
fail_unless_equals_string (attr->key, expected_directions[i]);
}
}
fail_unless (have_direction, "no direction attribute in media %u", i);
}
}
}
static void
on_sdp_media_no_duplicate_payloads (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
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);
GArray *media_formats = g_array_new (FALSE, FALSE, sizeof (int));
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]);
}
}
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", i);
have_setup = TRUE;
fail_unless_equals_string (attr->value, expected_setup[i]);
}
}
fail_unless (have_setup, "no setup attribute in media %u", i);
}
}
static void
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, 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);
}
static struct test_webrtc *
create_audio_test (void)
{
struct test_webrtc *t = test_webrtc_new ();
GstHarness *h;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
t->on_prepare_data_channel = have_prepare_data_channel;
h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
return t;
}
static void
on_new_transceiver_expected_kind (GstWebRTCBin * webrtc,
GstWebRTCRTPTransceiver * trans, gpointer user_data)
{
GstWebRTCKind kind, expected = GPOINTER_TO_UINT (user_data);
g_object_get (trans, "kind", &kind, NULL);
fail_unless_equals_int (kind, expected);
}
GST_START_TEST (test_audio)
{
struct test_webrtc *t = create_audio_test ();
VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
NULL, NULL);
guint media_format_count[] = { 1 };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, &no_duplicate_payloads);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (1),
&media_formats);
const gchar *expected_offer_setup[] = { "actpass", };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &count);
const gchar *expected_answer_setup[] = { "active", };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&count);
const gchar *expected_offer_direction[] = { "sendrecv", };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { "recvonly", };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
GstWebRTCKind expected_kind = GST_WEBRTC_KIND_AUDIO;
/* check that a single stream connection creates the associated number
* of media sections */
g_signal_connect (t->webrtc1, "on-new-transceiver",
G_CALLBACK (on_new_transceiver_expected_kind),
GUINT_TO_POINTER (expected_kind));
g_signal_connect (t->webrtc2, "on-new-transceiver",
G_CALLBACK (on_new_transceiver_expected_kind),
GUINT_TO_POINTER (expected_kind));
test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_END_TEST;
static void
_check_ice_port_restriction (struct test_webrtc *t, GstElement * element,
guint mlineindex, gchar * candidate, GstElement * other, gpointer user_data)
{
GRegex *regex;
GMatchInfo *match_info;
gchar *candidate_port;
gchar *candidate_protocol;
gchar *candidate_typ;
guint port_as_int;
guint peer_number;
if (!candidate || candidate[0] == '\0')
return;
regex =
g_regex_new ("candidate:(\\d+) (1) (UDP|TCP) (\\d+) ([0-9.]+|[0-9a-f:]+)"
" (\\d+) typ ([a-z]+)", 0, 0, NULL);
g_regex_match (regex, candidate, 0, &match_info);
fail_unless (g_match_info_get_match_count (match_info) == 8, candidate);
candidate_protocol = g_match_info_fetch (match_info, 2);
candidate_port = g_match_info_fetch (match_info, 6);
candidate_typ = g_match_info_fetch (match_info, 7);
peer_number = t->webrtc1 == element ? 1 : 2;
port_as_int = atoi (candidate_port);
if (!g_strcmp0 (candidate_typ, "host") && port_as_int != 9) {
guint expected_min = peer_number * 10000 + 1000;
guint expected_max = expected_min + 999;
fail_unless (port_as_int >= expected_min);
fail_unless (port_as_int <= expected_max);
}
g_free (candidate_port);
g_free (candidate_protocol);
g_free (candidate_typ);
g_match_info_free (match_info);
g_regex_unref (regex);
}
GST_START_TEST (test_ice_port_restriction)
{
struct test_webrtc *t = create_audio_test ();
GObject *webrtcice;
VAL_SDP_INIT (offer, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
VAL_SDP_INIT (answer, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
/*
* Ports are defined as follows "{peer}{protocol}000"
* - peer number: "1" for t->webrtc1, "2" for t->webrtc2
*/
g_object_get (t->webrtc1, "ice-agent", &webrtcice, NULL);
g_object_set (webrtcice, "min-rtp-port", 11000, "max-rtp-port", 11999, NULL);
g_object_unref (webrtcice);
g_object_get (t->webrtc2, "ice-agent", &webrtcice, NULL);
g_object_set (webrtcice, "min-rtp-port", 21000, "max-rtp-port", 21999, NULL);
g_object_unref (webrtcice);
t->on_ice_candidate = _check_ice_port_restriction;
test_validate_sdp (t, &offer, &answer);
test_webrtc_wait_for_ice_gathering_complete (t);
test_webrtc_free (t);
}
GST_END_TEST;
static struct test_webrtc *
create_audio_video_test (void)
{
struct test_webrtc *t = create_audio_test ();
GstHarness *h;
h = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
add_fake_video_src_harness (h, 97, 0xBEEFDEAD);
t->harnesses = g_list_prepend (t->harnesses, h);
return t;
}
GST_START_TEST (test_audio_video)
{
struct test_webrtc *t = create_audio_video_test ();
VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
NULL, NULL);
guint media_format_count[] = { 1, 1 };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, &no_duplicate_payloads);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&media_formats);
const gchar *expected_offer_setup[] = { "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &count);
const gchar *expected_answer_setup[] = { "active", "active" };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&count);
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);
/* check that a dual stream connection creates the associated number
* of media sections */
test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_media_direction)
{
struct test_webrtc *t = create_audio_video_test ();
VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
NULL, NULL);
guint media_format_count[] = { 1, 1 };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, &no_duplicate_payloads);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&media_formats);
const gchar *expected_offer_setup[] = { "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &count);
const gchar *expected_answer_setup[] = { "active", "active" };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&count);
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[] = { "sendrecv", "recvonly" };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
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, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
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 video_mline = GPOINTER_TO_UINT (user_data);
guint j;
vmedia = gst_sdp_message_get_media (desc->sdp, video_mline);
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");
} else if (g_str_has_prefix (attr->value, "101")) {
fail_unless_equals_string (attr->value, "101 H264/90000");
}
}
}
}
/* 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 ();
VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
NULL, NULL);
guint media_format_count[] = { 1, 5, };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, &no_duplicate_payloads);
VAL_SDP_INIT (payloads, on_sdp_media_payload_types, GUINT_TO_POINTER (1),
&media_formats);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2), &payloads);
const gchar *expected_offer_setup[] = { "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &count);
const gchar *expected_offer_direction[] = { "sendrecv", "sendrecv" };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
GstWebRTCRTPTransceiver *trans;
GArray *transceivers;
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);
/* We don't really care about the answer here */
test_validate_sdp (t, &offer, NULL);
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_request_pad_simple (t->webrtc1, "sink_0");
fail_unless (pad == NULL);
fail_unless_equals_int (STATE_ERROR,
test_webrtc_wait_for_answer_error_eos (t));
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);
fail_unless_equals_int (STATE_ERROR,
test_webrtc_wait_for_answer_error_eos (t));
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, *kind;
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);
fail_unless (gst_structure_get (s, "kind", G_TYPE_STRING, &kind, NULL));
fail_unless (g_str_equal (kind, "audio") || g_str_equal (kind, "video"));
g_free (codec_id);
g_free (transport_id);
g_free (kind);
}
static void
validate_inbound_rtp_stats (const GstStructure * s, const GstStructure * stats)
{
guint ssrc, fir, pli, nack;
gint64 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_INT64,
&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;
gint64 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_INT64,
&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));
if (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 void
validate_candidate_stats (const GstStructure * s, const GstStructure * stats)
{
guint port;
guint64 priority;
gchar *address, *candidateType, *protocol;
fail_unless (gst_structure_get (s, "address", G_TYPE_STRING, &address, NULL));
fail_unless (gst_structure_get (s, "port", G_TYPE_UINT, &port, NULL));
fail_unless (gst_structure_get (s, "candidate-type", G_TYPE_STRING,
&candidateType, NULL));
fail_unless (gst_structure_get (s, "priority", G_TYPE_UINT64, &priority,
NULL));
fail_unless (gst_structure_get (s, "protocol", G_TYPE_STRING, &protocol,
NULL));
fail_unless (strcmp (protocol, "udp") || strcmp (protocol, "tcp"));
g_free (address);
g_free (candidateType);
g_free (protocol);
}
static void
validate_peer_connection_stats (const GstStructure * s)
{
guint opened, closed;
fail_unless (gst_structure_get (s, "data-channels-opened", G_TYPE_UINT,
&opened, NULL));
fail_unless (gst_structure_get (s, "data-channels-closed", G_TYPE_UINT,
&closed, NULL));
fail_unless (opened >= closed);
}
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) {
validate_peer_connection_stats (s);
} 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) {
validate_candidate_stats (s, stats);
} else if (type == GST_WEBRTC_STATS_REMOTE_CANDIDATE) {
validate_candidate_stats (s, stats);
} 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;
test_validate_sdp (t, NULL, NULL);
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_stats_with_stream)
{
struct test_webrtc *t = create_audio_test ();
GstPromise *p;
GstCaps *caps;
GstPad *pad;
/* test that the stats generated with stream are sane */
t->on_offer_created = NULL;
t->on_answer_created = NULL;
t->on_negotiation_needed = 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);
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);
/* set caps for webrtcbin sink to validate codec stats */
caps = gst_caps_from_string (OPUS_RTP_CAPS (96));
pad = gst_element_get_static_pad (t->webrtc1, "sink_0");
gst_pad_set_caps (pad, caps);
gst_caps_unref (caps);
test_webrtc_wait_for_answer_error_eos (t);
test_webrtc_signal_state (t, STATE_ANSWER_SET);
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);
gst_object_unref (pad);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_add_transceiver)
{
struct test_webrtc *t = test_webrtc_new ();
GstWebRTCRTPTransceiverDirection direction, trans_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);
g_object_get (trans, "direction", &trans_direction, 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;
static void
on_sdp_media_check_mid (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
const char **mid = user_data;
guint 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 seen_mid = FALSE;
guint 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, "mid") == 0) {
fail_unless (!seen_mid);
seen_mid = TRUE;
fail_unless_equals_string (attr->value, mid[i]);
}
}
}
}
GST_START_TEST (test_add_recvonly_transceiver)
{
struct test_webrtc *t = test_webrtc_new ();
GstWebRTCRTPTransceiverDirection direction;
GstWebRTCRTPTransceiver *trans;
VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
NULL, NULL);
guint media_format_count[] = { 1, 1, };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, &no_duplicate_payloads);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (1),
&media_formats);
const char *expected_mid[] = { "gst", };
VAL_SDP_INIT (mid, on_sdp_media_check_mid, expected_mid, &count);
const gchar *expected_offer_setup[] = { "actpass", };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &mid);
const gchar *expected_answer_setup[] = { "active", };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup, &mid);
const gchar *expected_offer_direction[] = { "recvonly", };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { "sendonly", };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
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;
/* setup recvonly transceiver */
caps = gst_caps_from_string (OPUS_RTP_CAPS (96) ", a-mid=(string)gst");
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, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_recvonly_sendonly)
{
struct test_webrtc *t = test_webrtc_new ();
GstWebRTCRTPTransceiverDirection direction;
GstWebRTCRTPTransceiver *trans;
VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
NULL, NULL);
guint media_format_count[] = { 1, 1, };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, &no_duplicate_payloads);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&media_formats);
const gchar *expected_offer_setup[] = { "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &count);
const gchar *expected_answer_setup[] = { "active", "active" };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&count);
const gchar *expected_offer_direction[] = { "recvonly", "sendonly" };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { "sendonly", "recvonly" };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
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_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
/* 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);
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, 0xBEEFDEAD);
t->harnesses = g_list_prepend (t->harnesses, h);
g_signal_emit_by_name (t->webrtc1, "get-transceivers", &transceivers);
fail_unless (transceivers != NULL);
fail_unless_equals_int (transceivers->len, 2);
trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 1);
g_object_set (trans, "direction",
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, NULL);
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, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
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);
}
GST_START_TEST (test_data_channel_create)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
gchar *label;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_prepare_data_channel = have_prepare_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);
g_object_get (channel, "label", &label, NULL);
g_assert_cmpstr (label, ==, "label");
test_validate_sdp (t, &offer, &offer);
g_object_unref (channel);
g_free (label);
test_webrtc_free (t);
}
GST_END_TEST;
static void
signal_data_channel (struct test_webrtc *t,
GstElement * element, GObject * our, gpointer user_data)
{
test_webrtc_signal_state_unlocked (t, STATE_CUSTOM);
}
GST_START_TEST (test_data_channel_create_two_channels)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
GObject *channel2 = NULL;
VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
gchar *label;
GstStructure *options = NULL;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_prepare_data_channel = have_prepare_data_channel;
t->on_data_channel = signal_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);
g_object_get (channel, "label", &label, NULL);
g_assert_cmpstr (label, ==, "label");
g_free (label);
g_object_unref (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);
/* Wait SCTP transport creation */
test_validate_sdp_full (t, &offer, &offer, 1 << STATE_CUSTOM, FALSE);
/* Create another channel on an existing SCTP transport, forcing an ID that
should comply with the max-channels requiremennt, this should not raise a
critical warning, the id is beneath the required limits. */
options =
gst_structure_new ("options", "id", G_TYPE_INT, 2, "negotiated",
G_TYPE_BOOLEAN, TRUE, NULL);
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label2", options,
&channel2);
gst_structure_free (options);
g_assert_nonnull (channel2);
g_object_get (channel2, "label", &label, NULL);
g_assert_cmpstr (label, ==, "label2");
g_free (label);
g_object_unref (channel2);
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_assert_true (t->error_signal_handler_id > 0);
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;
VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_prepare_data_channel = have_prepare_data_channel;
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;
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_validate_sdp_full (t, &offer, &offer, 1 << STATE_CUSTOM, FALSE);
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)
{
GstWebRTCDataChannelState state;
gchar *expected;
g_object_get (channel, "ready-state", &state, NULL);
fail_unless_equals_int (GST_WEBRTC_DATA_CHANNEL_STATE_OPEN, state);
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;
GError *error = NULL;
g_object_get (our, "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);
fail_unless (gst_webrtc_data_channel_send_string_full (GST_WEBRTC_DATA_CHANNEL
(other), test_string, &error));
g_assert_null (error);
}
GST_START_TEST (test_data_channel_transfer_string)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_prepare_data_channel = have_prepare_data_channel;
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_validate_sdp_full (t, &offer, &offer, 1 << STATE_CUSTOM, FALSE);
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)
{
GstWebRTCDataChannelState state;
GBytes *expected;
g_object_get (channel, "ready-state", &state, NULL);
fail_unless_equals_int (GST_WEBRTC_DATA_CHANNEL_STATE_OPEN, state);
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;
GError *error = NULL;
g_object_get (our, "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);
fail_unless (gst_webrtc_data_channel_send_data_full (GST_WEBRTC_DATA_CHANNEL
(other), data, &error));
g_assert_null (error);
g_bytes_unref (data);
}
GST_START_TEST (test_data_channel_transfer_data)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_prepare_data_channel = have_prepare_data_channel;
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_validate_sdp_full (t, &offer, &offer, 1 << STATE_CUSTOM, FALSE);
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_prepare_data_channel = have_prepare_data_channel;
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;
t->data_channel_notify = (GDestroyNotify) g_object_unref;
}
GST_START_TEST (test_data_channel_create_after_negotiate)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_prepare_data_channel = have_prepare_data_channel;
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;
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_validate_sdp_full (t, &offer, &offer, 1 << STATE_CUSTOM, FALSE);
g_object_unref (channel);
test_webrtc_free (t);
}
GST_END_TEST;
struct test_data_channel
{
GObject *dc1;
GObject *dc2;
gint n_open;
gint n_closed;
};
static void
have_data_channel_mark_open (struct test_webrtc *t,
GstElement * element, GObject * our, gpointer user_data)
{
struct test_data_channel *tdc = t->data_channel_data;
tdc->dc2 = g_object_ref (our);
if (g_atomic_int_add (&tdc->n_open, 1) == 1) {
test_webrtc_signal_state_unlocked (t, STATE_CUSTOM);
}
}
static gboolean
is_data_channel_open (GObject * channel)
{
GstWebRTCDataChannelState ready_state = GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED;
if (channel) {
g_object_get (channel, "ready-state", &ready_state, NULL);
}
return ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_OPEN;
}
static void
on_data_channel_open (GObject * channel, GParamSpec * pspec,
struct test_webrtc *t)
{
struct test_data_channel *tdc = t->data_channel_data;
if (is_data_channel_open (channel)) {
if (g_atomic_int_add (&tdc->n_open, 1) == 1) {
test_webrtc_signal_state (t, STATE_CUSTOM);
}
}
}
static void
on_data_channel_close (GObject * channel, GParamSpec * pspec,
struct test_webrtc *t)
{
struct test_data_channel *tdc = t->data_channel_data;
GstWebRTCDataChannelState ready_state;
g_object_get (channel, "ready-state", &ready_state, NULL);
if (ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED) {
g_atomic_int_add (&tdc->n_closed, 1);
}
}
GST_START_TEST (test_data_channel_close)
{
#define NUM_CHANNELS 3
struct test_webrtc *t = test_webrtc_new ();
struct test_data_channel tdc = { NULL, };
guint channel_id[NUM_CHANNELS] = { 0, 1, 2 };
gulong sigid = 0;
int i;
VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_prepare_data_channel = have_prepare_data_channel;
t->on_data_channel = have_data_channel_mark_open;
t->data_channel_data = &tdc;
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);
/* open and close NUM_CHANNELS data channels to verify that we can reuse the
* stream id of a previously closed data channel and that we have the same
* behaviour no matter if we create the channel in READY or PLAYING state */
for (i = 0; i < NUM_CHANNELS; i++) {
GWeakRef dc1_ref, dc2_ref;
tdc.n_open = 0;
tdc.n_closed = 0;
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&tdc.dc1);
g_assert_nonnull (tdc.dc1);
g_weak_ref_init (&dc1_ref, tdc.dc1);
sigid = g_signal_connect (tdc.dc1, "notify::ready-state",
G_CALLBACK (on_data_channel_open), t);
if (i == 0) {
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_validate_sdp_full (t, &offer, &offer, 1 << STATE_CUSTOM, FALSE);
}
/* FIXME: Creating a data channel may result in "on-open" being sent
* before we even had a chance to register the signal. For this test we
* want to make sure that the channel is actually open before we try to
* close it. So if we didn't receive the signal we fall back to a 1s
* timeout where we explicitly check if both channels are open. */
while (!is_data_channel_open (tdc.dc1)
|| !is_data_channel_open (tdc.dc2))
g_usleep (100 * 1000);
g_object_get (tdc.dc1, "id", &channel_id[i], NULL);
g_signal_handler_disconnect (tdc.dc1, sigid);
g_weak_ref_init (&dc2_ref, tdc.dc2);
g_signal_connect (tdc.dc1, "notify::ready-state",
G_CALLBACK (on_data_channel_close), t);
g_signal_connect (tdc.dc2, "notify::ready-state",
G_CALLBACK (on_data_channel_close), t);
test_webrtc_signal_state (t, STATE_NEW);
/* currently we assume there is no renegotiation if the last data channel is
* removed but if it changes this test could be extended to verify both
* the behaviour of removing the last channel as well as the behaviour when
* there are still data channels remaining */
t->on_negotiation_needed = _negotiation_not_reached;
g_signal_emit_by_name (tdc.dc1, "close");
/* XXX: try to do something better here */
while (g_atomic_int_get (&tdc.n_closed) != 2)
g_usleep (100 * 1000);
g_clear_object (&tdc.dc1);
g_clear_object (&tdc.dc2);
/* XXX: try to do something better here */
while (g_weak_ref_get (&dc1_ref) != NULL
|| g_weak_ref_get (&dc2_ref) != NULL)
g_usleep (100 * 1000);
g_weak_ref_clear (&dc1_ref);
g_weak_ref_clear (&dc2_ref);
test_webrtc_signal_state (t, STATE_NEW);
test_webrtc_wait_for_state_mask (t, 1 << STATE_NEW);
}
/* verify the same stream id has been reused for each data channel */
assert_equals_int (channel_id[0], channel_id[1]);
assert_equals_int (channel_id[0], channel_id[2]);
test_webrtc_free (t);
#undef NUM_CHANNELS
}
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);
gst_webrtc_data_channel_send_string_full (GST_WEBRTC_DATA_CHANNEL (our), "A",
NULL);
}
GST_START_TEST (test_data_channel_low_threshold)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_prepare_data_channel = 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;
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_validate_sdp_full (t, &offer, &offer, 1 << STATE_CUSTOM, FALSE);
g_object_unref (channel);
test_webrtc_free (t);
}
GST_END_TEST;
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;
GError *error = NULL;
for (i = 0; i < size; i++)
random_data[i] = (guint8) (i & 0xff);
data = g_bytes_new_with_free_func (random_data, size,
(GDestroyNotify) g_free, random_data);
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);
fail_if (gst_webrtc_data_channel_send_data_full (GST_WEBRTC_DATA_CHANNEL
(other), data, &error));
g_assert_nonnull (error);
g_clear_error (&error);
g_bytes_unref (data);
test_webrtc_signal_state_unlocked (t, STATE_CUSTOM);
}
GST_START_TEST (test_data_channel_max_message_size)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_prepare_data_channel = 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_validate_sdp_full (t, &offer, &offer, 1 << STATE_CUSTOM, FALSE);
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 (g_atomic_int_add (n_ready, 1) >= 1) {
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;
VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
GstStructure *s;
gint n_ready = 0;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_prepare_data_channel = have_prepare_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);
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_validate_sdp_full (t, &offer, &offer, 0, FALSE);
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;
static void
_count_non_rejected_media (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * sd, gpointer user_data)
{
guint expected = GPOINTER_TO_UINT (user_data);
guint non_rejected_media;
guint i;
non_rejected_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);
if (gst_sdp_media_get_port (media) != 0)
non_rejected_media += 1;
}
fail_unless_equals_int (non_rejected_media, expected);
}
static void
_check_bundle_tag (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * sd, gpointer user_data)
{
gchar **bundled = NULL;
GStrv expected = user_data;
guint i;
fail_unless (_parse_bundle (sd->sdp, &bundled, NULL));
if (!bundled) {
fail_unless_equals_int (g_strv_length (expected), 0);
} else {
fail_unless_equals_int (g_strv_length (bundled), g_strv_length (expected));
}
for (i = 0; i < g_strv_length (expected); i++) {
fail_unless (g_strv_contains ((const gchar **) bundled, expected[i]));
}
g_strfreev (bundled);
}
static void
_check_bundle_only_media (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * sd, gpointer user_data)
{
gchar **expected_bundle_only = user_data;
guint i;
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 **) expected_bundle_only, mid))
fail_unless (_media_has_attribute_key (media, "bundle-only"));
}
}
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 };
guint media_format_count[] = { 1, 1, };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, NULL);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&media_formats);
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
VAL_SDP_INIT (bundle_tag, _check_bundle_tag, bundle, &payloads);
VAL_SDP_INIT (offer_non_reject, _count_non_rejected_media,
GUINT_TO_POINTER (1), &bundle_tag);
VAL_SDP_INIT (answer_non_reject, _count_non_rejected_media,
GUINT_TO_POINTER (2), &bundle_tag);
VAL_SDP_INIT (offer_bundle, _check_bundle_only_media, &offer_bundle_only,
&offer_non_reject);
VAL_SDP_INIT (answer_bundle, _check_bundle_only_media, &answer_bundle_only,
&answer_non_reject);
const gchar *expected_offer_setup[] = { "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&offer_bundle);
const gchar *expected_answer_setup[] = { "active", "active" };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&answer_bundle);
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);
/* 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
*/
gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
"max-bundle");
/* 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.
*/
gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy",
"max-bundle");
test_validate_sdp (t, &offer, &answer);
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 };
guint media_format_count[] = { 1, 1, };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, NULL);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&media_formats);
VAL_SDP_INIT (bundle_tag, _check_bundle_tag, bundle, &count);
VAL_SDP_INIT (count_non_reject, _count_non_rejected_media,
GUINT_TO_POINTER (2), &bundle_tag);
VAL_SDP_INIT (bundle_sdp, _check_bundle_only_media, &bundle_only,
&count_non_reject);
const gchar *expected_offer_setup[] = { "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&bundle_sdp);
const gchar *expected_answer_setup[] = { "active", "active" };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&bundle_sdp);
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);
/* 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
*/
gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
"max-compat");
/* 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.
*/
gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy",
"max-bundle");
test_validate_sdp (t, &offer, &answer);
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_mid[] = { "audio0", "video1", NULL };
const gchar *offer_bundle_only[] = { "video1", NULL };
const gchar *answer_mid[] = { NULL };
const gchar *answer_bundle_only[] = { NULL };
guint media_format_count[] = { 1, 1, };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, NULL);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&media_formats);
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
VAL_SDP_INIT (count_non_reject, _count_non_rejected_media,
GUINT_TO_POINTER (1), &payloads);
VAL_SDP_INIT (offer_bundle_tag, _check_bundle_tag, offer_mid,
&count_non_reject);
VAL_SDP_INIT (answer_bundle_tag, _check_bundle_tag, answer_mid,
&count_non_reject);
VAL_SDP_INIT (offer_bundle, _check_bundle_only_media, &offer_bundle_only,
&offer_bundle_tag);
VAL_SDP_INIT (answer_bundle, _check_bundle_only_media, &answer_bundle_only,
&answer_bundle_tag);
const gchar *expected_offer_setup[] = { "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&offer_bundle);
const gchar *expected_answer_setup[] = { "active", "active" };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&answer_bundle);
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);
/* 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
*/
gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
"max-bundle");
/* 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.
*/
gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy", "none");
test_validate_sdp (t, &offer, &answer);
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 *mids[] = { "audio0", "video1", "application2", NULL };
const gchar *offer_bundle_only[] = { "video1", "application2", NULL };
const gchar *answer_bundle_only[] = { NULL };
GObject *channel = NULL;
guint media_format_count[] = { 1, 1, 1 };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, NULL);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (3),
&media_formats);
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
VAL_SDP_INIT (bundle_tag, _check_bundle_tag, mids, &payloads);
VAL_SDP_INIT (offer_non_reject, _count_non_rejected_media,
GUINT_TO_POINTER (1), &bundle_tag);
VAL_SDP_INIT (answer_non_reject, _count_non_rejected_media,
GUINT_TO_POINTER (3), &bundle_tag);
VAL_SDP_INIT (offer_bundle, _check_bundle_only_media, &offer_bundle_only,
&offer_non_reject);
VAL_SDP_INIT (answer_bundle, _check_bundle_only_media, &answer_bundle_only,
&answer_non_reject);
const gchar *expected_offer_setup[] = { "actpass", "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&offer_bundle);
const gchar *expected_answer_setup[] = { "active", "active", "active" };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&answer_bundle);
const gchar *expected_offer_direction[] =
{ "sendrecv", "sendrecv", "sendrecv" };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] =
{ "recvonly", "recvonly", "recvonly" };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
/* 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
*/
gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
"max-bundle");
/* 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.
*/
gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy",
"max-bundle");
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_validate_sdp (t, &offer, &answer);
g_object_unref (channel);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_duplicate_nego)
{
struct test_webrtc *t = create_audio_video_test ();
guint media_format_count[] = { 1, 1, };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, NULL);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&media_formats);
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
const gchar *expected_offer_setup[] = { "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&payloads);
const gchar *expected_answer_setup[] = { "active", "active" };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&payloads);
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[] = { "sendrecv", "recvonly" };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
GstHarness *h;
guint negotiation_flag = 0;
/* check that negotiating twice succeeds */
t->on_negotiation_needed = on_negotiation_needed_hit;
t->negotiation_data = &negotiation_flag;
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
fail_unless (negotiation_flag & (1 << 2));
test_webrtc_reset_negotiation (t);
test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_dual_audio)
{
struct test_webrtc *t = create_audio_test ();
guint media_format_count[] = { 1, 1, };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, NULL);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&media_formats);
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
const gchar *expected_offer_setup[] = { "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&payloads);
const gchar *expected_answer_setup[] = { "active", "active" };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&payloads);
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[] = { "sendrecv", "recvonly" };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
GstHarness *h;
GstWebRTCRTPTransceiver *trans;
GArray *transceivers;
guint mline;
/* 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, 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, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
t->on_negotiation_needed = NULL;
test_validate_sdp (t, &offer, &answer);
g_signal_emit_by_name (t->webrtc1, "get-transceivers", &transceivers);
fail_unless (transceivers != NULL);
fail_unless_equals_int (2, transceivers->len);
trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 0);
fail_unless (trans != NULL);
g_object_get (trans, "mlineindex", &mline, NULL);
fail_unless_equals_int (mline, 0);
trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 1);
fail_unless (trans != NULL);
g_object_get (trans, "mlineindex", &mline, NULL);
fail_unless_equals_int (mline, 1);
g_array_unref (transceivers);
test_webrtc_free (t);
}
GST_END_TEST;
static void
sdp_increasing_session_version (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
GstWebRTCSessionDescription *previous;
const GstSDPOrigin *our_origin, *previous_origin;
const gchar *prop;
guint64 our_v, previous_v;
prop =
TEST_SDP_IS_LOCAL (t, element,
desc) ? "current-local-description" : "current-remote-description";
g_object_get (element, prop, &previous, NULL);
our_origin = gst_sdp_message_get_origin (desc->sdp);
previous_origin = gst_sdp_message_get_origin (previous->sdp);
our_v = g_ascii_strtoull (our_origin->sess_version, NULL, 10);
previous_v = g_ascii_strtoull (previous_origin->sess_version, NULL, 10);
ck_assert_int_lt (previous_v, our_v);
gst_webrtc_session_description_free (previous);
}
static void
sdp_equal_session_id (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
GstWebRTCSessionDescription *previous;
const GstSDPOrigin *our_origin, *previous_origin;
const gchar *prop;
prop =
TEST_SDP_IS_LOCAL (t, element,
desc) ? "current-local-description" : "current-remote-description";
g_object_get (element, prop, &previous, NULL);
our_origin = gst_sdp_message_get_origin (desc->sdp);
previous_origin = gst_sdp_message_get_origin (previous->sdp);
fail_unless_equals_string (previous_origin->sess_id, our_origin->sess_id);
gst_webrtc_session_description_free (previous);
}
static void
sdp_media_equal_attribute (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, GstWebRTCSessionDescription * previous,
const gchar * attr)
{
guint i, n;
n = MIN (gst_sdp_message_medias_len (previous->sdp),
gst_sdp_message_medias_len (desc->sdp));
for (i = 0; i < n; i++) {
const GstSDPMedia *our_media, *other_media;
const gchar *our_mid, *other_mid;
our_media = gst_sdp_message_get_media (desc->sdp, i);
other_media = gst_sdp_message_get_media (previous->sdp, i);
our_mid = gst_sdp_media_get_attribute_val (our_media, attr);
other_mid = gst_sdp_media_get_attribute_val (other_media, attr);
fail_unless_equals_string (our_mid, other_mid);
}
}
static void
sdp_media_equal_mid (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
GstWebRTCSessionDescription *previous;
const gchar *prop;
prop =
TEST_SDP_IS_LOCAL (t, element,
desc) ? "current-local-description" : "current-remote-description";
g_object_get (element, prop, &previous, NULL);
sdp_media_equal_attribute (t, element, desc, previous, "mid");
gst_webrtc_session_description_free (previous);
}
static void
sdp_media_equal_ice_params (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
GstWebRTCSessionDescription *previous;
const gchar *prop;
prop =
TEST_SDP_IS_LOCAL (t, element,
desc) ? "current-local-description" : "current-remote-description";
g_object_get (element, prop, &previous, NULL);
sdp_media_equal_attribute (t, element, desc, previous, "ice-ufrag");
sdp_media_equal_attribute (t, element, desc, previous, "ice-pwd");
gst_webrtc_session_description_free (previous);
}
static void
sdp_media_equal_fingerprint (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
GstWebRTCSessionDescription *previous;
const gchar *prop;
prop =
TEST_SDP_IS_LOCAL (t, element,
desc) ? "current-local-description" : "current-remote-description";
g_object_get (element, prop, &previous, NULL);
sdp_media_equal_attribute (t, element, desc, previous, "fingerprint");
gst_webrtc_session_description_free (previous);
}
GST_START_TEST (test_renego_add_stream)
{
struct test_webrtc *t = create_audio_video_test ();
guint media_format_count[] = { 1, 1, 1 };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, NULL);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&media_formats);
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
const gchar *expected_offer_setup[] = { "actpass", "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&payloads);
const gchar *expected_answer_setup[] = { "active", "active", "active" };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&payloads);
const gchar *expected_offer_direction[] =
{ "sendrecv", "sendrecv", "sendrecv" };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] =
{ "sendrecv", "recvonly", "recvonly" };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
VAL_SDP_INIT (renego_mid, sdp_media_equal_mid, NULL, NULL);
VAL_SDP_INIT (renego_ice_params, sdp_media_equal_ice_params, NULL,
&renego_mid);
VAL_SDP_INIT (renego_sess_id, sdp_equal_session_id, NULL, &renego_ice_params);
VAL_SDP_INIT (renego_sess_ver, sdp_increasing_session_version, NULL,
&renego_sess_id);
VAL_SDP_INIT (renego_fingerprint, sdp_media_equal_fingerprint, NULL,
&renego_sess_ver);
GstHarness *h;
/* 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, 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, 0xBEEFFFFF);
t->harnesses = g_list_prepend (t->harnesses, h);
media_formats.next = &renego_fingerprint;
count.user_data = GUINT_TO_POINTER (3);
/* renegotiate! */
test_webrtc_reset_negotiation (t);
test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_renego_stream_add_data_channel)
{
struct test_webrtc *t = create_audio_video_test ();
guint media_format_count[] = { 1, 1, 1 };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, NULL);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&media_formats);
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
const gchar *expected_offer_setup[] = { "actpass", "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&payloads);
const gchar *expected_answer_setup[] = { "active", "active", "active" };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&payloads);
const gchar *expected_offer_direction[] = { "sendrecv", "sendrecv", NULL };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { "sendrecv", "recvonly", NULL };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
VAL_SDP_INIT (renego_mid, sdp_media_equal_mid, NULL, NULL);
VAL_SDP_INIT (renego_ice_params, sdp_media_equal_ice_params, NULL,
&renego_mid);
VAL_SDP_INIT (renego_sess_id, sdp_equal_session_id, NULL, &renego_ice_params);
VAL_SDP_INIT (renego_sess_ver, sdp_increasing_session_version, NULL,
&renego_sess_id);
VAL_SDP_INIT (renego_fingerprint, sdp_media_equal_fingerprint, NULL,
&renego_sess_ver);
GObject *channel;
GstHarness *h;
/* 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, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
media_formats.next = &renego_fingerprint;
count.user_data = GUINT_TO_POINTER (3);
/* renegotiate! */
test_webrtc_reset_negotiation (t);
test_validate_sdp (t, &offer, &answer);
g_object_unref (channel);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_renego_data_channel_add_stream)
{
struct test_webrtc *t = test_webrtc_new ();
guint media_format_count[] = { 1, 1, 1 };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, NULL);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (1),
&media_formats);
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
const gchar *expected_offer_setup[] = { "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&payloads);
const gchar *expected_answer_setup[] = { "active", "active" };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&payloads);
const gchar *expected_offer_direction[] = { NULL, "sendrecv" };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { NULL, "recvonly" };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
VAL_SDP_INIT (renego_mid, sdp_media_equal_mid, NULL, NULL);
VAL_SDP_INIT (renego_ice_params, sdp_media_equal_ice_params, NULL,
&renego_mid);
VAL_SDP_INIT (renego_sess_id, sdp_equal_session_id, NULL, &renego_ice_params);
VAL_SDP_INIT (renego_sess_ver, sdp_increasing_session_version, NULL,
&renego_sess_id);
VAL_SDP_INIT (renego_fingerprint, sdp_media_equal_fingerprint, NULL,
&renego_sess_ver);
GObject *channel;
GstHarness *h;
/* negotiate an data channel and then renegotiate to add a av stream */
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_data_channel = NULL;
t->on_pad_added = _pad_added_fakesink;
t->on_prepare_data_channel = 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);
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
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, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
media_formats.next = &renego_fingerprint;
count.user_data = GUINT_TO_POINTER (2);
/* renegotiate! */
test_webrtc_reset_negotiation (t);
test_validate_sdp_full (t, &offer, &answer, 0, FALSE);
g_object_unref (channel);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_renego_stream_data_channel_add_stream)
{
struct test_webrtc *t = test_webrtc_new ();
guint media_format_count[] = { 1, 1, 1 };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, NULL);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&media_formats);
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
const gchar *expected_offer_setup[] = { "actpass", "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&payloads);
const gchar *expected_answer_setup[] = { "active", "active", "active" };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&payloads);
const gchar *expected_offer_direction[] = { "sendrecv", NULL, "sendrecv" };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { "recvonly", NULL, "recvonly" };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
VAL_SDP_INIT (renego_mid, sdp_media_equal_mid, NULL, NULL);
VAL_SDP_INIT (renego_ice_params, sdp_media_equal_ice_params, NULL,
&renego_mid);
VAL_SDP_INIT (renego_sess_id, sdp_equal_session_id, NULL, &renego_ice_params);
VAL_SDP_INIT (renego_sess_ver, sdp_increasing_session_version, NULL,
&renego_sess_id);
VAL_SDP_INIT (renego_fingerprint, sdp_media_equal_fingerprint, NULL,
&renego_sess_ver);
GObject *channel;
GstHarness *h;
/* Negotiate a stream and a data channel, then renogotiate with a new stream */
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_data_channel = NULL;
t->on_prepare_data_channel = 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, 97, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
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);
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
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, 0xBEEFDEAD);
t->harnesses = g_list_prepend (t->harnesses, h);
media_formats.next = &renego_fingerprint;
count.user_data = GUINT_TO_POINTER (3);
/* renegotiate! */
test_webrtc_reset_negotiation (t);
test_validate_sdp_full (t, &offer, &answer, 0, FALSE);
g_object_unref (channel);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_bundle_renego_add_stream)
{
struct test_webrtc *t = create_audio_video_test ();
const gchar *bundle[] = { "audio0", "video1", "audio2", NULL };
const gchar *offer_bundle_only[] = { "video1", "audio2", NULL };
const gchar *answer_bundle_only[] = { NULL };
guint media_format_count[] = { 1, 1, 1 };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, NULL);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&media_formats);
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
const gchar *expected_offer_setup[] = { "actpass", "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&payloads);
const gchar *expected_answer_setup[] = { "active", "active", "active" };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&payloads);
const gchar *expected_offer_direction[] =
{ "sendrecv", "sendrecv", "sendrecv" };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] =
{ "sendrecv", "recvonly", "recvonly" };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
VAL_SDP_INIT (renego_mid, sdp_media_equal_mid, NULL, &payloads);
VAL_SDP_INIT (renego_ice_params, sdp_media_equal_ice_params, NULL,
&renego_mid);
VAL_SDP_INIT (renego_sess_id, sdp_equal_session_id, NULL, &renego_ice_params);
VAL_SDP_INIT (renego_sess_ver, sdp_increasing_session_version, NULL,
&renego_sess_id);
VAL_SDP_INIT (renego_fingerprint, sdp_media_equal_fingerprint, NULL,
&renego_sess_ver);
VAL_SDP_INIT (bundle_tag, _check_bundle_tag, bundle, &renego_fingerprint);
VAL_SDP_INIT (offer_non_reject, _count_non_rejected_media,
GUINT_TO_POINTER (1), &bundle_tag);
VAL_SDP_INIT (answer_non_reject, _count_non_rejected_media,
GUINT_TO_POINTER (3), &bundle_tag);
VAL_SDP_INIT (offer_bundle_only_sdp, _check_bundle_only_media,
&offer_bundle_only, &offer_non_reject);
VAL_SDP_INIT (answer_bundle_only_sdp, _check_bundle_only_media,
&answer_bundle_only, &answer_non_reject);
GstHarness *h;
/* 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
*/
gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
"max-bundle");
/* 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.
*/
gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy",
"max-bundle");
/* 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, 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, 0xBEEFFFFF);
t->harnesses = g_list_prepend (t->harnesses, h);
offer_setup.next = &offer_bundle_only_sdp;
answer_setup.next = &answer_bundle_only_sdp;
count.user_data = GUINT_TO_POINTER (3);
/* renegotiate! */
test_webrtc_reset_negotiation (t);
test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_bundle_max_compat_max_bundle_renego_add_stream)
{
struct test_webrtc *t = create_audio_video_test ();
const gchar *bundle[] = { "audio0", "video1", "audio2", NULL };
const gchar *bundle_only[] = { NULL };
guint media_format_count[] = { 1, 1, 1 };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, NULL);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&media_formats);
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
const gchar *expected_offer_setup[] = { "actpass", "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&payloads);
const gchar *expected_answer_setup[] = { "active", "active", "active" };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&payloads);
const gchar *expected_offer_direction[] =
{ "sendrecv", "sendrecv", "sendrecv" };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] =
{ "sendrecv", "recvonly", "recvonly" };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
VAL_SDP_INIT (renego_mid, sdp_media_equal_mid, NULL, NULL);
VAL_SDP_INIT (renego_ice_params, sdp_media_equal_ice_params, NULL,
&renego_mid);
VAL_SDP_INIT (renego_sess_id, sdp_equal_session_id, NULL, &renego_ice_params);
VAL_SDP_INIT (renego_sess_ver, sdp_increasing_session_version, NULL,
&renego_sess_id);
VAL_SDP_INIT (renego_fingerprint, sdp_media_equal_fingerprint, NULL,
&renego_sess_ver);
VAL_SDP_INIT (bundle_tag, _check_bundle_tag, bundle, &renego_fingerprint);
VAL_SDP_INIT (count_non_reject, _count_non_rejected_media,
GUINT_TO_POINTER (3), &bundle_tag);
VAL_SDP_INIT (bundle_sdp, _check_bundle_only_media, &bundle_only,
&count_non_reject);
GstHarness *h;
/* 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
*/
gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
"max-compat");
/* 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.
*/
gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy",
"max-bundle");
/* 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, 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, 0xBEEFFFFF);
t->harnesses = g_list_prepend (t->harnesses, h);
media_formats.next = &bundle_sdp;
count.user_data = GUINT_TO_POINTER (3);
/* renegotiate! */
test_webrtc_reset_negotiation (t);
test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_renego_transceiver_set_direction)
{
struct test_webrtc *t = create_audio_test ();
guint media_format_count[] = { 1, };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, NULL);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (1),
&media_formats);
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
const gchar *expected_offer_setup[] = { "actpass", };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&payloads);
const gchar *expected_answer_setup[] = { "active", };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&payloads);
const gchar *expected_offer_direction[] = { "sendrecv", };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { "sendrecv", };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
GstWebRTCRTPTransceiver *transceiver;
GstHarness *h;
GstPad *pad;
/* 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, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
/* renegotiate an inactive transceiver! */
pad = gst_element_get_static_pad (t->webrtc1, "sink_0");
g_object_get (pad, "transceiver", &transceiver, NULL);
fail_unless (transceiver != NULL);
g_object_set (transceiver, "direction",
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE, NULL);
expected_offer_direction[0] = "inactive";
expected_answer_direction[0] = "inactive";
/* TODO: also validate EOS events from the inactive change */
test_webrtc_reset_negotiation (t);
test_validate_sdp (t, &offer, &answer);
gst_object_unref (pad);
gst_object_unref (transceiver);
test_webrtc_free (t);
}
GST_END_TEST;
static void
offer_remove_last_media (struct test_webrtc *t, GstElement * element,
GstPromise * promise, gpointer user_data)
{
guint i, n;
GstSDPMessage *new, *old;
const GstSDPOrigin *origin;
const GstSDPConnection *conn;
old = t->offer_desc->sdp;
fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_new (&new));
origin = gst_sdp_message_get_origin (old);
conn = gst_sdp_message_get_connection (old);
fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_set_version (new,
gst_sdp_message_get_version (old)));
fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_set_origin (new,
origin->username, origin->sess_id, origin->sess_version,
origin->nettype, origin->addrtype, origin->addr));
fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_set_session_name (new,
gst_sdp_message_get_session_name (old)));
fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_set_information (new,
gst_sdp_message_get_information (old)));
fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_set_uri (new,
gst_sdp_message_get_uri (old)));
fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_set_connection (new,
conn->nettype, conn->addrtype, conn->address, conn->ttl,
conn->addr_number));
n = gst_sdp_message_attributes_len (old);
for (i = 0; i < n; i++) {
const GstSDPAttribute *a = gst_sdp_message_get_attribute (old, i);
fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_add_attribute (new,
a->key, a->value));
}
n = gst_sdp_message_medias_len (old);
fail_unless (n > 0);
for (i = 0; i < n - 1; i++) {
const GstSDPMedia *m = gst_sdp_message_get_media (old, i);
GstSDPMedia *new_m;
fail_unless_equals_int (GST_SDP_OK, gst_sdp_media_copy (m, &new_m));
fail_unless_equals_int (GST_SDP_OK, gst_sdp_message_add_media (new, new_m));
gst_sdp_media_init (new_m);
gst_sdp_media_free (new_m);
}
gst_webrtc_session_description_free (t->offer_desc);
t->offer_desc = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_OFFER,
new);
}
static void
offer_set_produced_error (struct test_webrtc *t, GstElement * element,
GstPromise * promise, gpointer user_data)
{
const GstStructure *reply;
GError *error = NULL;
reply = gst_promise_get_reply (promise);
fail_unless (gst_structure_get (reply, "error", G_TYPE_ERROR, &error, NULL));
GST_INFO ("error produced: %s", error->message);
g_clear_error (&error);
test_webrtc_signal_state_unlocked (t, STATE_CUSTOM);
}
static void
offer_created_produced_error (struct test_webrtc *t, GstElement * element,
GstPromise * promise, gpointer user_data)
{
const GstStructure *reply;
GError *error = NULL;
reply = gst_promise_get_reply (promise);
fail_unless (gst_structure_get (reply, "error", G_TYPE_ERROR, &error, NULL));
GST_INFO ("error produced: %s", error->message);
g_clear_error (&error);
}
GST_START_TEST (test_renego_lose_media_fails)
{
struct test_webrtc *t = create_audio_video_test ();
VAL_SDP_INIT (offer, _count_num_sdp_media, GUINT_TO_POINTER (2), NULL);
VAL_SDP_INIT (answer, _count_num_sdp_media, GUINT_TO_POINTER (2), NULL);
/* check that removing an m=line will produce an error */
test_validate_sdp (t, &offer, &answer);
test_webrtc_reset_negotiation (t);
t->on_offer_created = offer_remove_last_media;
t->on_offer_set = offer_set_produced_error;
t->on_answer_created = NULL;
test_webrtc_create_offer (t);
test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_bundle_codec_preferences_rtx_no_duplicate_payloads)
{
struct test_webrtc *t = test_webrtc_new ();
GstWebRTCRTPTransceiverDirection direction;
GstWebRTCRTPTransceiver *trans;
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);
const gchar *expected_offer_setup[] = { "actpass", };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&offer_media_formats);
const gchar *expected_answer_setup[] = { "active", };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&answer_media_formats);
const gchar *expected_offer_direction[] = { "recvonly", };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { "sendonly", };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
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, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_END_TEST;
static void
on_sdp_media_no_duplicate_extmaps (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
const GstSDPMedia *media = gst_sdp_message_get_media (desc->sdp, 0);
fail_unless (media != NULL);
fail_unless_equals_string (gst_sdp_media_get_attribute_val_n (media, "extmap",
0), "1 foobar");
fail_unless (gst_sdp_media_get_attribute_val_n (media, "extmap", 1) == NULL);
}
/* In this test, we validate that identical extmaps for multiple formats
* in the caps of a single transceiver are deduplicated. This is necessary
* because Firefox will complain about duplicate extmap ids and fail negotiation
* otherwise. */
GST_START_TEST (test_codec_preferences_no_duplicate_extmaps)
{
struct test_webrtc *t = test_webrtc_new ();
GstWebRTCRTPTransceiver *trans;
GstWebRTCRTPTransceiverDirection direction;
VAL_SDP_INIT (extmaps, on_sdp_media_no_duplicate_extmaps, NULL, NULL);
GstCaps *caps;
GstStructure *s;
caps = gst_caps_new_empty ();
s = gst_structure_from_string (VP8_RTP_CAPS (96), NULL);
gst_structure_set (s, "extmap-1", G_TYPE_STRING, "foobar", NULL);
gst_caps_append_structure (caps, s);
s = gst_structure_from_string (H264_RTP_CAPS (97), NULL);
gst_structure_set (s, "extmap-1", G_TYPE_STRING, "foobar", NULL);
gst_caps_append_structure (caps, s);
direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY;
g_signal_emit_by_name (t->webrtc1, "add-transceiver", direction, caps,
&trans);
gst_caps_unref (caps);
fail_unless (trans != NULL);
t->on_negotiation_needed = NULL;
t->on_pad_added = NULL;
t->on_ice_candidate = NULL;
test_validate_sdp (t, &extmaps, NULL);
test_webrtc_free (t);
}
GST_END_TEST;
/* In this test, we validate that trying to use different values
* for the same extmap id in multiple formats in the caps of a
* single transceiver errors out when creating the offer. */
GST_START_TEST (test_codec_preferences_incompatible_extmaps)
{
struct test_webrtc *t = test_webrtc_new ();
GstWebRTCRTPTransceiver *trans;
GstWebRTCRTPTransceiverDirection direction;
GstCaps *caps;
GstStructure *s;
caps = gst_caps_new_empty ();
s = gst_structure_from_string (VP8_RTP_CAPS (96), NULL);
gst_structure_set (s, "extmap-1", G_TYPE_STRING, "foobar", NULL);
gst_caps_append_structure (caps, s);
s = gst_structure_from_string (H264_RTP_CAPS (97), NULL);
gst_structure_set (s, "extmap-1", G_TYPE_STRING, "foobaz", NULL);
gst_caps_append_structure (caps, s);
direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY;
g_signal_emit_by_name (t->webrtc1, "add-transceiver", direction, caps,
&trans);
gst_caps_unref (caps);
fail_unless (trans != NULL);
t->on_negotiation_needed = NULL;
t->on_pad_added = NULL;
t->on_ice_candidate = NULL;
t->on_offer_created = offer_created_produced_error;
test_validate_sdp_full (t, NULL, NULL, 1 << STATE_ERROR, TRUE);
test_webrtc_free (t);
}
GST_END_TEST;
/* In this test, we validate that extmap values must be of the correct type */
GST_START_TEST (test_codec_preferences_invalid_extmap)
{
struct test_webrtc *t = test_webrtc_new ();
GstWebRTCRTPTransceiver *trans;
GstWebRTCRTPTransceiverDirection direction;
GstCaps *caps;
GstStructure *s;
caps = gst_caps_new_empty ();
s = gst_structure_from_string (VP8_RTP_CAPS (96), NULL);
gst_structure_set (s, "extmap-1", G_TYPE_INT, 42, NULL);
gst_caps_append_structure (caps, s);
direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY;
g_signal_emit_by_name (t->webrtc1, "add-transceiver", direction, caps,
&trans);
gst_caps_unref (caps);
fail_unless (trans != NULL);
t->on_negotiation_needed = NULL;
t->on_pad_added = NULL;
t->on_ice_candidate = NULL;
t->on_offer_created = offer_created_produced_error;
test_validate_sdp_full (t, NULL, NULL, 1 << STATE_ERROR, TRUE);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_reject_request_pad)
{
struct test_webrtc *t = test_webrtc_new ();
GstWebRTCRTPTransceiverDirection direction;
GstWebRTCRTPTransceiver *trans, *trans2;
guint offer_media_format_count[] = { 1, };
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);
const gchar *expected_offer_setup[] = { "actpass", };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&offer_media_formats);
const gchar *expected_answer_setup[] = { "active", };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&answer_media_formats);
const gchar *expected_offer_direction[] = { "recvonly", };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { "sendonly", };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
GstCaps *caps;
GstHarness *h;
GstPad *pad;
GstPadTemplate *templ;
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);
gst_caps_unref (caps);
fail_unless (trans != NULL);
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_video_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
/* This should fail because the direction is wrong */
pad = gst_element_request_pad_simple (t->webrtc1, "sink_0");
fail_unless (pad == NULL);
g_object_set (trans, "direction",
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV, NULL);
templ = gst_element_get_pad_template (t->webrtc1, "sink_%u");
fail_unless (templ != NULL);
/* This should fail because the caps are wrong */
caps = gst_caps_from_string (OPUS_RTP_CAPS (96));
pad = gst_element_request_pad (t->webrtc1, templ, "sink_0", caps);
fail_unless (pad == NULL);
g_object_set (trans, "codec-preferences", NULL, NULL);
/* This should fail because the kind doesn't match */
pad = gst_element_request_pad (t->webrtc1, templ, "sink_0", caps);
fail_unless (pad == NULL);
gst_caps_unref (caps);
/* This should succeed and give us sink_0 */
pad = gst_element_request_pad_simple (t->webrtc1, "sink_0");
fail_unless (pad != NULL);
g_object_get (pad, "transceiver", &trans2, NULL);
fail_unless (trans == trans2);
gst_object_unref (pad);
gst_object_unref (trans);
gst_object_unref (trans2);
test_webrtc_free (t);
}
GST_END_TEST;
static void
_verify_media_types (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
gchar **media_types = 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_string (gst_sdp_media_get_media (media), media_types[i]);
}
}
GST_START_TEST (test_reject_create_offer)
{
struct test_webrtc *t = test_webrtc_new ();
GstHarness *h;
GstPromise *promise;
GstPromiseResult res;
const GstStructure *s;
GError *error = NULL;
const gchar *media_types[] = { "video", "audio" };
VAL_SDP_INIT (media_type, _verify_media_types, &media_types, NULL);
guint media_format_count[] = { 1, 1 };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, &media_type);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&media_formats);
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
const gchar *expected_offer_setup[] = { "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&payloads);
const gchar *expected_answer_setup[] = { "active", "active" };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&payloads);
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);
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
/* setup sendonly peer */
h = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
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 */
promise = gst_promise_new ();
g_signal_emit_by_name (t->webrtc1, "create-offer", NULL, promise);
res = gst_promise_wait (promise);
fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
s = gst_promise_get_reply (promise);
fail_unless (s != NULL);
gst_structure_get (s, "error", G_TYPE_ERROR, &error, NULL);
fail_unless (g_error_matches (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE));
fail_unless_equals_string (error->message,
"Tranceiver <webrtctransceiver0> with mid (null) has locked mline 1 but the offer only has 0 sections");
g_clear_error (&error);
gst_promise_unref (promise);
h = gst_harness_new_with_element (t->webrtc1, "sink_%u", NULL);
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 */
test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_reject_create_offer_mline_locked_no_caps)
{
GstHarness *h;
GstPromise *promise;
const GstStructure *s;
GstPromiseResult res;
GError *error = NULL;
h = gst_harness_new_with_padnames ("webrtcbin", "sink_0", NULL);
promise = gst_promise_new ();
g_signal_emit_by_name (h->element, "create-offer", NULL, promise);
res = gst_promise_wait (promise);
fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
s = gst_promise_get_reply (promise);
fail_unless (s != NULL);
gst_structure_get (s, "error", G_TYPE_ERROR, &error, NULL);
fail_unless (g_error_matches (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE));
fail_unless_equals_string (error->message,
"Trying to add transceiver at line 0 but there is a transceiver with a"
" locked mline for this line which doesn't have caps");
g_clear_error (&error);
gst_promise_unref (promise);
gst_harness_teardown (h);
}
GST_END_TEST;
GST_START_TEST (test_reject_set_description)
{
struct test_webrtc *t = test_webrtc_new ();
GstHarness *h;
GstPromise *promise;
GstPromiseResult res;
const GstStructure *s;
GError *error = NULL;
GstWebRTCSessionDescription *desc = NULL;
GstPadTemplate *templ;
GstCaps *caps;
GstPad *pad;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
/* setup peer 1 */
h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
/* Create a second side with specific video caps */
templ = gst_element_get_pad_template (t->webrtc2, "sink_%u");
fail_unless (templ != NULL);
caps = gst_caps_from_string (VP8_RTP_CAPS (97));
pad = gst_element_request_pad (t->webrtc2, templ, "sink_0", caps);
fail_unless (pad != NULL);
gst_caps_unref (caps);
gst_object_unref (pad);
/* Create an offer */
promise = gst_promise_new ();
g_signal_emit_by_name (t->webrtc1, "create-offer", NULL, promise);
res = gst_promise_wait (promise);
fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
s = gst_promise_get_reply (promise);
fail_unless (s != NULL);
gst_structure_get (s, "offer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &desc,
NULL);
fail_unless (desc != NULL);
gst_promise_unref (promise);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
/* Verify that setting an offer where there is a forced m-line with
a different kind fails. */
promise = gst_promise_new ();
g_signal_emit_by_name (t->webrtc2, "set-remote-description", desc, promise);
res = gst_promise_wait (promise);
fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
s = gst_promise_get_reply (promise);
gst_structure_get (s, "error", G_TYPE_ERROR, &error, NULL);
fail_unless (g_error_matches (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE));
fail_unless_equals_string
(error->message,
"m-line 0 with transceiver <webrtctransceiver1> was locked to video, but SDP has audio media");
g_clear_error (&error);
fail_unless (s != NULL);
gst_promise_unref (promise);
gst_webrtc_session_description_free (desc);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_force_second_media)
{
struct test_webrtc *t = test_webrtc_new ();
const gchar *media_types[] = { "audio" };
VAL_SDP_INIT (media_type, _verify_media_types, &media_types, NULL);
guint media_format_count[] = { 1, };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, &media_type);
const gchar *expected_offer_setup[] = { "actpass", };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&media_formats);
const gchar *expected_answer_setup[] = { "active", };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&media_formats);
const gchar *expected_offer_direction[] = { "sendrecv", };
VAL_SDP_INIT (offer_direction, on_sdp_media_direction,
expected_offer_direction, &offer_setup);
const gchar *expected_answer_direction[] = { "recvonly", };
VAL_SDP_INIT (answer_direction, on_sdp_media_direction,
expected_answer_direction, &answer_setup);
VAL_SDP_INIT (answer_count, _count_num_sdp_media, GUINT_TO_POINTER (1),
&answer_direction);
VAL_SDP_INIT (offer_count, _count_num_sdp_media, GUINT_TO_POINTER (1),
&offer_direction);
const gchar *second_media_types[] = { "audio", "video" };
VAL_SDP_INIT (second_media_type, _verify_media_types, &second_media_types,
NULL);
guint second_media_format_count[] = { 1, 1 };
VAL_SDP_INIT (second_media_formats, on_sdp_media_count_formats,
second_media_format_count, &second_media_type);
const gchar *second_expected_offer_setup[] = { "active", "actpass" };
VAL_SDP_INIT (second_offer_setup, on_sdp_media_setup,
second_expected_offer_setup, &second_media_formats);
const gchar *second_expected_answer_setup[] = { "passive", "active" };
VAL_SDP_INIT (second_answer_setup, on_sdp_media_setup,
second_expected_answer_setup, &second_media_formats);
const gchar *second_expected_answer_direction[] = { "sendonly", "recvonly" };
VAL_SDP_INIT (second_answer_direction, on_sdp_media_direction,
second_expected_answer_direction, &second_answer_setup);
const gchar *second_expected_offer_direction[] = { "recvonly", "sendrecv" };
VAL_SDP_INIT (second_offer_direction, on_sdp_media_direction,
second_expected_offer_direction, &second_offer_setup);
VAL_SDP_INIT (second_answer_count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&second_answer_direction);
VAL_SDP_INIT (second_offer_count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&second_offer_direction);
GstHarness *h;
guint negotiation_flag = 0;
GstPadTemplate *templ;
GstCaps *caps;
GstPad *pad;
/* add a transceiver that will only receive an opus stream and check that
* the created offer is marked as recvonly */
t->on_negotiation_needed = on_negotiation_needed_hit;
t->negotiation_data = &negotiation_flag;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
/* setup peer */
h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
/* Create a second side with specific video caps */
templ = gst_element_get_pad_template (t->webrtc2, "sink_%u");
fail_unless (templ != NULL);
caps = gst_caps_from_string (VP8_RTP_CAPS (97));
pad = gst_element_request_pad (t->webrtc2, templ, NULL, caps);
gst_caps_unref (caps);
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, 0xBEEFDEAD);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer_count, &answer_count);
fail_unless (negotiation_flag & 1 << 2);
test_webrtc_reset_negotiation (t);
t->offerror = 2;
test_validate_sdp (t, &second_offer_count, &second_answer_count);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_codec_preferences_caps)
{
GstHarness *h;
GstPad *pad;
GstWebRTCRTPTransceiver *trans;
GstCaps *caps, *caps2;
h = gst_harness_new_with_padnames ("webrtcbin", "sink_0", NULL);
pad = gst_element_get_static_pad (h->element, "sink_0");
g_object_get (pad, "transceiver", &trans, NULL);
caps = gst_caps_from_string ("application/x-rtp, media=video,"
"encoding-name=VP8, payload=115; application/x-rtp, media=video,"
" encoding-name=H264, payload=104");
g_object_set (trans, "codec-preferences", caps, NULL);
caps2 = gst_pad_query_caps (pad, NULL);
fail_unless (gst_caps_is_equal (caps, caps2));
gst_caps_unref (caps2);
gst_caps_unref (caps);
caps = gst_caps_from_string (VP8_RTP_CAPS (115));
fail_unless (gst_pad_query_accept_caps (pad, caps));
gst_harness_set_src_caps (h, g_steal_pointer (&caps));
caps = gst_caps_from_string (VP8_RTP_CAPS (99));
fail_unless (!gst_pad_query_accept_caps (pad, caps));
gst_caps_unref (caps);
gst_object_unref (pad);
gst_object_unref (trans);
gst_harness_teardown (h);
}
GST_END_TEST;
GST_START_TEST (test_codec_preferences_negotiation_sinkpad)
{
struct test_webrtc *t = test_webrtc_new ();
guint media_format_count[] = { 1, };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, NULL);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (1),
&media_formats);
VAL_SDP_INIT (payloads2, on_sdp_media_payload_types, GUINT_TO_POINTER (0),
&count);
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &payloads2);
const gchar *expected_offer_setup[] = { "actpass", };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&payloads);
const gchar *expected_answer_setup[] = { "active", };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&payloads);
const gchar *expected_offer_direction[] = { "sendrecv", };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { "recvonly", };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
GstPad *pad;
GstWebRTCRTPTransceiver *transceiver;
GstHarness *h;
GstCaps *caps;
GstPromise *promise;
GstPromiseResult res;
const GstStructure *s;
GError *error = NULL;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
pad = gst_element_get_static_pad (t->webrtc1, "sink_0");
g_object_get (pad, "transceiver", &transceiver, NULL);
caps = gst_caps_from_string (VP8_RTP_CAPS (115) ";" VP8_RTP_CAPS (97));
g_object_set (transceiver, "codec-preferences", caps, NULL);
gst_caps_unref (caps);
gst_object_unref (transceiver);
gst_object_unref (pad);
add_fake_video_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
promise = gst_promise_new ();
g_signal_emit_by_name (t->webrtc1, "create-offer", NULL, promise);
res = gst_promise_wait (promise);
fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
s = gst_promise_get_reply (promise);
fail_unless (s != NULL);
gst_structure_get (s, "error", G_TYPE_ERROR, &error, NULL);
fail_unless (g_error_matches (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE));
fail_unless_equals_string
("Caps negotiation on pad sink_0 failed against codec preferences",
error->message);
g_clear_error (&error);
gst_promise_unref (promise);
caps = gst_caps_from_string (VP8_RTP_CAPS (97));
gst_harness_set_src_caps (h, caps);
test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_END_TEST;
static void
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);
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 ! "
"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
}
struct pad_added_harness_data
{
GList *sink_harnesses;
OnPadAdded on_pad_added;
gpointer on_pad_added_data;
};
static void
_pad_added_harness (struct test_webrtc *t, GstElement * element,
GstPad * pad, gpointer user_data)
{
struct pad_added_harness_data *data = user_data;
GstHarness *h;
if (GST_PAD_DIRECTION (pad) != GST_PAD_SRC)
return;
h = gst_harness_new_with_element (element, NULL, GST_OBJECT_NAME (pad));
t->harnesses = g_list_prepend (t->harnesses, h);
if (data) {
data->sink_harnesses = g_list_prepend (data->sink_harnesses, h);
g_cond_broadcast (&t->cond);
if (data->on_pad_added)
data->on_pad_added (t, element, pad, data->on_pad_added_data);
}
}
static void
new_jitterbuffer_set_fast_start (GstElement * rtpbin,
GstElement * rtpjitterbuffer, guint session_id, guint ssrc,
gpointer user_data)
{
g_object_set (rtpjitterbuffer, "faststart-min-packets", 1, NULL);
}
GST_START_TEST (test_codec_preferences_negotiation_srcpad)
{
struct test_webrtc *t = test_webrtc_new ();
guint media_format_count[] = { 1, };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, NULL);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (1),
&media_formats);
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
const gchar *expected_offer_setup[] = { "actpass", };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&payloads);
const gchar *expected_answer_setup[] = { "active", };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&payloads);
const gchar *expected_offer_direction[] = { "sendrecv", };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { "recvonly", };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
VAL_SDP_INIT (answer_non_reject, _count_non_rejected_media,
GUINT_TO_POINTER (0), &count);
GstHarness *h;
struct pad_added_harness_data pad_added_data = { NULL, };
GstHarness *sink_harness = NULL;
guint i;
GstElement *rtpbin2;
GstBuffer *buf;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_harness;
t->pad_added_data = &pad_added_data;
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);
h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
add_audio_test_src_harness (h, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
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 (h);
g_mutex_lock (&t->lock);
while (pad_added_data.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 (pad_added_data.sink_harnesses));
sink_harness = (GstHarness *) pad_added_data.sink_harnesses->data;
g_mutex_unlock (&t->lock);
fail_unless (sink_harness->element == t->webrtc2);
/* Get one buffer out, this makes sure the capsfilter is primed and
* avoids races.
*/
buf = gst_harness_pull (sink_harness);
fail_unless (buf != NULL);
gst_buffer_unref (buf);
gst_harness_set_sink_caps_str (sink_harness, OPUS_RTP_CAPS (100));
test_webrtc_reset_negotiation (t);
test_validate_sdp_full (t, &offer, &answer_non_reject, 0, FALSE);
/* check that the mid/mline is correct */
{
GstWebRTCRTPTransceiver *rtp_trans;
GstPad *srcpad;
guint mline;
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 != NULL);
g_object_get (rtp_trans, "mlineindex", &mline, NULL);
gst_clear_object (&rtp_trans);
fail_unless_equals_int (mline, 0);
}
test_webrtc_free (t);
g_list_free (pad_added_data.sink_harnesses);
}
GST_END_TEST;
static void
_on_new_transceiver_codec_preferences_h264 (GstElement * webrtcbin,
GstWebRTCRTPTransceiver * trans, gpointer * user_data)
{
GstCaps *caps;
caps = gst_caps_from_string ("application/x-rtp,encoding-name=(string)H264");
g_object_set (trans, "codec-preferences", caps, NULL);
gst_caps_unref (caps);
}
static void
on_sdp_media_payload_types_only_h264 (struct test_webrtc *t,
GstElement * element, GstWebRTCSessionDescription * desc,
gpointer user_data)
{
const GstSDPMedia *vmedia;
guint video_mline = GPOINTER_TO_UINT (user_data);
guint j;
vmedia = gst_sdp_message_get_media (desc->sdp, video_mline);
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")) {
fail_unless_equals_string (attr->value, "101 H264/90000");
}
}
}
GST_START_TEST (test_codec_preferences_in_on_new_transceiver)
{
struct test_webrtc *t = test_webrtc_new ();
GstWebRTCRTPTransceiverDirection direction;
GstWebRTCRTPTransceiver *trans;
VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
NULL, NULL);
guint offer_media_format_count[] = { 2 };
guint answer_media_format_count[] = { 1 };
VAL_SDP_INIT (offer_media_formats, on_sdp_media_count_formats,
offer_media_format_count, &no_duplicate_payloads);
VAL_SDP_INIT (answer_media_formats, on_sdp_media_count_formats,
answer_media_format_count, &no_duplicate_payloads);
VAL_SDP_INIT (offer_count, _count_num_sdp_media, GUINT_TO_POINTER (1),
&offer_media_formats);
VAL_SDP_INIT (answer_count, _count_num_sdp_media, GUINT_TO_POINTER (1),
&answer_media_formats);
VAL_SDP_INIT (offer_payloads, on_sdp_media_payload_types,
GUINT_TO_POINTER (0), &offer_count);
VAL_SDP_INIT (answer_payloads, on_sdp_media_payload_types_only_h264,
GUINT_TO_POINTER (0), &answer_count);
const gchar *expected_offer_setup[] = { "actpass", };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&offer_payloads);
const gchar *expected_answer_setup[] = { "active", };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&answer_payloads);
const gchar *expected_offer_direction[] = { "sendonly", };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { "recvonly", };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
GstCaps *caps;
GstHarness *h;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
/* setup sendonly transceiver with VP8 and H264 */
caps = gst_caps_from_string (VP8_RTP_CAPS (97) ";" H264_RTP_CAPS (101));
direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY;
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 recvonly peer */
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
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 */
g_signal_connect (t->webrtc2, "on-new-transceiver",
G_CALLBACK (_on_new_transceiver_codec_preferences_h264), NULL);
/* Answer SDP should now have H264 only. Without the codec-preferences it
* would only have VP8 because that comes first in the SDP */
test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_renego_rtx)
{
struct test_webrtc *t = create_audio_video_test ();
VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
NULL, NULL);
guint media_format_count[] = { 1, 1 };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, &no_duplicate_payloads);
VAL_SDP_INIT (count_media, _count_num_sdp_media, GUINT_TO_POINTER (2),
&media_formats);
VAL_SDP_INIT (payloads, on_sdp_media_payload_types,
GUINT_TO_POINTER (1), &count_media);
const gchar *expected_offer_direction[] = { "sendrecv", "sendrecv", };
VAL_SDP_INIT (offer_direction, on_sdp_media_direction,
expected_offer_direction, &payloads);
const gchar *expected_answer_direction[] = { "recvonly", "recvonly", };
VAL_SDP_INIT (answer_direction, on_sdp_media_direction,
expected_answer_direction, &payloads);
const gchar *expected_offer_setup[] = { "actpass", "actpass", };
VAL_SDP_INIT (offer, on_sdp_media_setup, expected_offer_setup,
&offer_direction);
const gchar *expected_answer_setup[] = { "active", "active", };
VAL_SDP_INIT (answer, on_sdp_media_setup, expected_answer_setup,
&answer_direction);
GstWebRTCRTPTransceiver *trans;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
test_validate_sdp (t, &offer, &answer);
test_webrtc_reset_negotiation (t);
g_signal_emit_by_name (t->webrtc1, "get-transceiver", 1, &trans);
g_object_set (trans, "do-nack", TRUE, "fec-type",
GST_WEBRTC_FEC_TYPE_ULP_RED, NULL);
g_clear_object (&trans);
g_signal_emit_by_name (t->webrtc2, "get-transceiver", 1, &trans);
g_object_set (trans, "do-nack", TRUE, "fec-type",
GST_WEBRTC_FEC_TYPE_ULP_RED, NULL);
g_clear_object (&trans);
/* adding RTX/RED/FEC increases the number of media formats */
media_format_count[1] = 5;
test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_bundle_mid_header_extension)
{
struct test_webrtc *t = test_webrtc_new ();
GstWebRTCRTPTransceiverDirection direction;
GstWebRTCRTPTransceiver *trans;
VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
NULL, NULL);
guint media_format_count[] = { 1, 1, };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, &no_duplicate_payloads);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (1),
&media_formats);
const char *expected_mid[] = { "gst", };
VAL_SDP_INIT (mid, on_sdp_media_check_mid, expected_mid, &count);
const gchar *expected_offer_setup[] = { "actpass", };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &mid);
const gchar *expected_answer_setup[] = { "active", };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup, &mid);
const gchar *expected_offer_direction[] = { "recvonly", };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { "sendonly", };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
GstCaps *caps;
GstHarness *h;
guint mline;
char *trans_mid;
/* 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;
/* setup recvonly transceiver */
caps = gst_caps_from_string (OPUS_RTP_CAPS (96) ", a-mid=(string)gst");
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);
g_object_get (trans, "mlineindex", &mline, NULL);
fail_unless_equals_int (mline, -1);
/* setup sendonly peer */
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_audio_src_harness (h, 96, 0xDEADBEEF);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
g_object_get (trans, "mlineindex", &mline, "mid", &trans_mid, NULL);
fail_unless_equals_int (mline, 0);
fail_unless_equals_string (trans_mid, "gst");
g_clear_pointer (&trans_mid, g_free);
gst_object_unref (trans);
test_webrtc_free (t);
}
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);
}
struct pad_properties
{
const char *mid;
guint mlineindex;
};
struct new_pad_validate_properties
{
guint n_props;
struct pad_properties *props;
};
static void
on_pad_added_validate_props (struct test_webrtc *t, GstElement * element,
GstPad * pad, gpointer user_data)
{
GstWebRTCRTPTransceiver *rtp_trans;
struct new_pad_validate_properties *pad_props = user_data;
char *trans_mid;
guint mlineindex;
int i;
g_object_get (pad, "transceiver", &rtp_trans, NULL);
fail_unless (rtp_trans);
g_object_get (rtp_trans, "mid", &trans_mid, "mlineindex", &mlineindex, NULL);
fail_unless (trans_mid != NULL);
fail_unless (mlineindex != -1);
for (i = 0; i < pad_props->n_props; i++) {
struct pad_properties *expected = &pad_props->props[i];
if (g_strcmp0 (expected->mid, trans_mid) == 0) {
if (expected->mlineindex != -1) {
fail_unless_equals_int (mlineindex, expected->mlineindex);
break;
}
}
}
if (i == pad_props->n_props)
fail ("could not find a matching expected output pad for mid %s and mline %u", trans_mid, mlineindex);
g_clear_pointer (&trans_mid, g_free);
gst_clear_object (&rtp_trans);
}
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;
struct pad_properties pad_prop[] = {
{"audio0", 0},
{"audio1", 1},
};
struct new_pad_validate_properties validate_pad = { 2, pad_prop };
struct pad_added_harness_data pad_added_data =
{ NULL, on_pad_added_validate_props, &validate_pad };
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 = &pad_added_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");
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 *sink_harnesses = pad_added_data.sink_harnesses;
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;
}
g_mutex_unlock (&t->lock);
for (l = sink_harnesses; l; l = l->next) {
GstHarness *sink_harness = (GstHarness *) l->data;
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
guint ssrc;
fail_unless (sink_harness->element == t->webrtc2);
buf = gst_harness_try_pull (sink_harness);
if (!buf)
continue;
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_lock (&t->lock);
}
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 (pad_added_data.sink_harnesses);
g_array_unref (ssrcs_received);
}
GST_END_TEST;
#define RTPHDREXT_MID GST_RTP_HDREXT_BASE "sdes:mid"
#define RTPHDREXT_STREAM_ID GST_RTP_HDREXT_BASE "sdes:rtp-stream-id"
#define RTPHDREXT_REPAIRED_STREAM_ID GST_RTP_HDREXT_BASE "sdes:repaired-rtp-stream-id"
#define L16_CAPS "application/x-rtp, payload=11, media=audio," \
" encoding-name=L16, clock-rate=44100"
static GstCaps *
create_simulcast_audio_caps (GstWebRTCRTPTransceiverDirection direction,
guint n_rid, guint ssrc[], const char *mid, guint mid_ext_id,
const char *const *rids, guint stream_ext_id, guint repaired_ext_id)
{
GstStructure *s;
GstCaps *caps;
const char *dir_str;
if (direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY)
dir_str = "recv";
else if (direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY)
dir_str = "send";
else
g_assert_not_reached ();
caps = gst_caps_from_string (L16_CAPS);
s = gst_caps_get_structure (caps, 0);
if (mid && mid_ext_id != G_MAXUINT) {
char *extmap_key = g_strdup_printf ("extmap-%u", mid_ext_id);
gst_structure_set (s, "a-mid", G_TYPE_STRING, mid, extmap_key,
G_TYPE_STRING, RTPHDREXT_MID, NULL);
g_free (extmap_key);
}
if (rids && n_rid > 0 && stream_ext_id != G_MAXUINT) {
GString *simulcast_value = g_string_new (dir_str);
char *extmap_key, *value;
int i;
g_string_append_c (simulcast_value, ' ');
for (i = 0; i < n_rid; i++) {
char *rid_key = g_strdup_printf ("rid-%s", rids[i]);
gst_structure_set (s, rid_key, G_TYPE_STRING, dir_str, NULL);
if (i > 0)
g_string_append_c (simulcast_value, ';');
g_string_append (simulcast_value, rids[i]);
g_free (rid_key);
}
value = g_string_free (simulcast_value, FALSE);
simulcast_value = NULL;
extmap_key = g_strdup_printf ("extmap-%u", stream_ext_id);
gst_structure_set (s, extmap_key, G_TYPE_STRING, RTPHDREXT_STREAM_ID,
"a-simulcast", G_TYPE_STRING, value, NULL);
g_clear_pointer (&extmap_key, g_free);
g_clear_pointer (&value, g_free);
if (repaired_ext_id != G_MAXUINT) {
extmap_key = g_strdup_printf ("extmap-%u", repaired_ext_id);
gst_structure_set (s, extmap_key, G_TYPE_STRING,
RTPHDREXT_REPAIRED_STREAM_ID, NULL);
g_clear_pointer (&extmap_key, g_free);
}
}
return caps;
}
static void
add_simulcast_audio_test_src_harness (GstHarness * h, guint n_rid,
guint ssrc[], const char *mid, guint mid_ext_id,
const char *const *rids, guint stream_ext_id, guint repaired_ext_id)
{
GstRTPHeaderExtension *ext;
GstElement *capsfilter;
char *launch_str;
GstCaps *caps;
int i;
caps =
create_simulcast_audio_caps
(GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, n_rid, ssrc, mid,
mid_ext_id, rids, stream_ext_id, repaired_ext_id);
gst_harness_set_src_caps (h, gst_caps_ref (caps));
if (n_rid == 0) {
launch_str =
g_strdup ("audiotestsrc is-live=true ! " L16_CAPS
",ssrc=(uint)3384078954 ! rtpL16pay name=payloader0");
} else {
GString *launch = g_string_new (NULL);
for (i = 0; i < n_rid; i++) {
const char *rtpfunnel = "funnel.";
if (i == 0)
rtpfunnel = "rtpfunnel name=funnel ! capsfilter name=capsfilter";
g_string_append_printf (launch, "audiotestsrc is-live=true ! "
"rtpL16pay name=payloader%u ! " L16_CAPS ", ssrc=(uint)%u ! %s ", i,
ssrc[i], rtpfunnel);
}
launch_str = g_string_free (launch, FALSE);
}
GST_INFO ("generated launch string %s", launch_str);
gst_harness_add_src_parse (h, launch_str, TRUE);
g_clear_pointer (&launch_str, g_free);
capsfilter =
gst_bin_get_by_name (GST_BIN (h->src_harness->element), "capsfilter");
g_object_set (capsfilter, "caps", caps, NULL);
gst_clear_object (&capsfilter);
gst_clear_caps (&caps);
for (i = 0; i == 0 || i < n_rid; i++) {
const char *rid = n_rid > 0 ? rids[i] : NULL;
char *pay_name = g_strdup_printf ("payloader%u", i);
GstElement *payloader =
gst_bin_get_by_name (GST_BIN (h->src_harness->element), pay_name);
fail_unless (payloader);
g_clear_pointer (&pay_name, g_free);
if (mid_ext_id != G_MAXUINT) {
ext = gst_rtp_header_extension_create_from_uri (RTPHDREXT_MID);
fail_unless (ext);
gst_rtp_header_extension_set_id (ext, mid_ext_id);
g_object_set (ext, "mid", mid, NULL);
g_signal_emit_by_name (payloader, "add-extension", ext);
gst_clear_object (&ext);
}
if (n_rid > 0 && stream_ext_id != G_MAXUINT) {
ext = gst_rtp_header_extension_create_from_uri (RTPHDREXT_STREAM_ID);
fail_unless (ext);
gst_rtp_header_extension_set_id (ext, stream_ext_id);
g_object_set (ext, "rid", rid, NULL);
g_signal_emit_by_name (payloader, "add-extension", ext);
gst_clear_object (&ext);
}
if (n_rid > 0 && repaired_ext_id != G_MAXUINT) {
ext =
gst_rtp_header_extension_create_from_uri
(RTPHDREXT_REPAIRED_STREAM_ID);
fail_unless (ext);
gst_rtp_header_extension_set_id (ext, repaired_ext_id);
g_object_set (ext, "rid", rid, NULL);
g_signal_emit_by_name (payloader, "add-extension", ext);
gst_clear_object (&ext);
}
gst_clear_object (&payloader);
}
}
#undef L16_CAPS
static gboolean
gst_g_ptr_array_find_str (GPtrArray * ptr, const char *needle, guint * index)
{
guint i;
for (i = 0; i < ptr->len; i++) {
const char *test = g_ptr_array_index (ptr, i);
if (g_strcmp0 (test, needle) == 0) {
if (index)
*index = i;
return TRUE;
}
}
return FALSE;
}
struct ExpectedRid
{
guint n_rid;
const char *const *rid;
};
static void
on_sdp_media_rid (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
struct ExpectedRid *expected_rids = 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);
struct ExpectedRid *expected_rid = &expected_rids[i];
GPtrArray *seen_rid = g_ptr_array_new_with_free_func (g_free);
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, "rid") == 0) {
const char *p;
char *v;
guint k;
p = attr->value;
/* take up to either space or nul-terminator */
while (p && *p && *p == ' ')
p++;
v = (char *) p;
/* take up to either space or nul-terminator */
while (p && *p && *p != ' ')
p++;
g_assert (v != p);
v = g_strndup (v, p - v);
GST_INFO ("rid = %s", v);
fail_unless (FALSE == gst_g_ptr_array_find_str (seen_rid, v, NULL),
"duplicate/multiple rid for media %u", i);
for (k = 0; k < expected_rid->n_rid; k++) {
GST_LOG ("expected %u = %s", k, expected_rid->rid[k]);
if (g_strcmp0 (v, expected_rid->rid[k]) == 0)
break;
}
fail_unless (k < expected_rid->n_rid, "rid %s not found in media %u",
v, i);
g_ptr_array_add (seen_rid, v);
}
}
fail_unless (seen_rid->len == expected_rid->n_rid,
"mismatch in number of rid's in media %u, seen %u, expected %u", i,
seen_rid->len, expected_rid->n_rid);
g_ptr_array_unref (seen_rid);
}
}
static void
do_test_simulcast (gboolean enable_fec_rtx)
{
struct test_webrtc *t = test_webrtc_new ();
guint media_format_count[] = { enable_fec_rtx ? 5 : 1, };
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);
const char *expected_rids0[] = { "a", "z" };
struct ExpectedRid expected_rids = { G_N_ELEMENTS (expected_rids0),
expected_rids0
};
VAL_SDP_INIT (rids, on_sdp_media_rid, &expected_rids, &payloads);
VAL_SDP_INIT (non_reject, _count_non_rejected_media,
GUINT_TO_POINTER (1), &rids);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (1), &non_reject);
const gchar *expected_offer_setup[] = { "actpass", };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &count);
const gchar *expected_answer_setup[] = { "active", };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&count);
const gchar *expected_offer_direction[] = { "sendrecv", };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { "recvonly", };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
GstHarness *h;
GObject *trans;
guint i;
GstElement *rtpbin2;
GstBuffer *buf;
guint mid_ext_id = 1;
guint stream_ext_id = 2;
guint repaired_ext_id = 3;
const char *mid = "5";
guint ssrcs[] = { 123456789, 987654321 };
GArray *ssrcs_received;
GstCaps *caps;
struct pad_properties pad_prop = { mid, 0 };
struct new_pad_validate_properties validate_pad = { 1, &pad_prop };
struct pad_added_harness_data pad_added_data =
{ NULL, on_pad_added_validate_props, &validate_pad };
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_harness;
t->pad_added_data = &pad_added_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");
if (enable_fec_rtx) {
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);
}
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);
h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
add_simulcast_audio_test_src_harness (h, expected_rids.n_rid, ssrcs, mid,
mid_ext_id, expected_rids.rid, stream_ext_id, repaired_ext_id);
t->harnesses = g_list_prepend (t->harnesses, h);
/* setup recvonly transceiver as answer */
caps =
create_simulcast_audio_caps
(GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, expected_rids.n_rid,
ssrcs, mid, mid_ext_id, expected_rids.rid, stream_ext_id,
repaired_ext_id);
g_signal_emit_by_name (t->webrtc2, "add-transceiver",
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, caps, &trans);
gst_clear_caps (&caps);
fail_unless (trans != NULL);
g_clear_object (&trans);
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 (h);
}
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 *sink_harnesses = pad_added_data.sink_harnesses;
GList *l;
guint i;
gst_harness_push_from_src (h);
if (g_list_length (pad_added_data.sink_harnesses) < 2) {
g_cond_wait_until (&t->cond, &t->lock, g_get_monotonic_time () + 5000);
if (g_list_length (pad_added_data.sink_harnesses) < 2)
continue;
}
g_mutex_unlock (&t->lock);
for (l = sink_harnesses; l; l = l->next) {
GstHarness *sink_harness = (GstHarness *) l->data;
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
guint ssrc;
fail_unless (sink_harness->element == t->webrtc2);
buf = gst_harness_try_pull (sink_harness);
if (!buf)
continue;
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_lock (&t->lock);
}
g_mutex_unlock (&t->lock);
test_webrtc_free (t);
g_list_free (pad_added_data.sink_harnesses);
g_array_unref (ssrcs_received);
}
GST_START_TEST (test_simulcast)
{
do_test_simulcast (FALSE);
}
GST_END_TEST;
GST_START_TEST (test_simulcast_fec_rtx)
{
do_test_simulcast (TRUE);
}
GST_END_TEST;
GST_START_TEST (test_bundle_multiple_media_rtx_payload_mapping)
{
struct test_webrtc *t = test_webrtc_new ();
guint offer_media_format_count[] = { 5, 5, };
VAL_SDP_INIT (payloads0, on_sdp_media_payload_types, GUINT_TO_POINTER (0),
NULL);
VAL_SDP_INIT (payloads1, on_sdp_media_payload_types, GUINT_TO_POINTER (1),
&payloads0);
VAL_SDP_INIT (no_dup_payloads, on_sdp_media_no_duplicate_payloads, NULL,
&payloads1);
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
offer_media_format_count, &no_dup_payloads);
const gchar *expected_offer_setup[] = { "actpass", "actpass", };
VAL_SDP_INIT (setup, on_sdp_media_setup, expected_offer_setup,
&media_formats);
const gchar *expected_offer_direction[] = { "recvonly", "recvonly", };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&setup);
GstWebRTCRTPTransceiverDirection direction;
GstWebRTCRTPTransceiver *trans;
GstCaps *caps;
/* add two identical transceivers that will only receive a vp8 stream and check that
* the created offer has the same rtx/red mappings */
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
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 (97));
direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY;
g_signal_emit_by_name (t->webrtc1, "add-transceiver", direction, caps,
&trans);
fail_unless (trans != NULL);
g_object_set (GST_OBJECT (trans), "do-nack", TRUE, "fec-type",
GST_WEBRTC_FEC_TYPE_ULP_RED, NULL);
gst_object_unref (trans);
direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY;
g_signal_emit_by_name (t->webrtc1, "add-transceiver", direction, caps,
&trans);
fail_unless (trans != NULL);
g_object_set (GST_OBJECT (trans), "do-nack", TRUE, "fec-type",
GST_WEBRTC_FEC_TYPE_ULP_RED, NULL);
gst_object_unref (trans);
gst_caps_unref (caps);
/* don't really care about the answer */
test_validate_sdp (t, &offer, NULL);
test_webrtc_free (t);
}
GST_END_TEST;
static void
add_media_line (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
GstSDPMedia *media = NULL;
const GstSDPMedia *existing_media;
GstSDPResult res;
existing_media = gst_sdp_message_get_media (desc->sdp, 0);
res = gst_sdp_media_copy (existing_media, &media);
fail_unless (res == GST_SDP_OK);
res = gst_sdp_message_add_media (desc->sdp, media);
fail_unless (res == GST_SDP_OK);
gst_sdp_media_free (media);
}
static void
on_answer_set_rejected (struct test_webrtc *t, GstElement * element,
GstPromise * promise, gpointer user_data)
{
const GstStructure *s;
GError *error = NULL;
GError *compare_error = user_data;
s = gst_promise_get_reply (promise);
fail_unless (s != NULL);
gst_structure_get (s, "error", G_TYPE_ERROR, &error, NULL);
fail_unless (g_error_matches (error, compare_error->domain,
compare_error->code));
fail_unless_equals_string (compare_error->message, error->message);
g_clear_error (&error);
}
GST_START_TEST (test_invalid_add_media_in_answer)
{
struct test_webrtc *t = create_audio_test ();
VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
NULL, NULL);
guint media_format_count[] = { 1 };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, &no_duplicate_payloads);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (1),
&media_formats);
const gchar *expected_offer_setup[] = { "actpass", };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &count);
const gchar *expected_offer_direction[] = { "sendrecv", };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
VAL_SDP_INIT (answer, add_media_line, NULL, NULL);
GError answer_set_error = { GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
(gchar *) "Answer doesn't have the same number of m-lines as the offer."
};
/* Ensure that if the answer has more m-lines than the offer, it gets
* rejected.
*/
t->on_answer_set = on_answer_set_rejected;
t->answer_set_data = &answer_set_error;
test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_END_TEST;
#define VALID_TURN_SERVER_URL1 "turn://testuser:testpass@test.com:1234"
#define VALID_TURN_SERVER_URL2 "turns://1665056262%3Atestuser:T4VwcehYgPAa5bpFAO14gVE19so=@test.com:1234"
#define INVALID_TURN_SERVER_URL1 "testuser@testpass@test.com:1234" /* protocol of uri is missing */
#define INVALID_TURN_SERVER_URL2 "turns://testuser:testpass/@test.com:1234" /* unescaped character in password */
#define INVALID_TURN_SERVER_URL3 "turns://test.com:1234" /* 'user:pass' is missing */
GST_START_TEST (test_add_turn_server)
{
struct test_webrtc *t = test_webrtc_new ();
gboolean ret = FALSE;
g_signal_emit_by_name (t->webrtc1, "add-turn-server",
VALID_TURN_SERVER_URL1, &ret);
fail_unless (ret != FALSE);
g_signal_emit_by_name (t->webrtc1, "add-turn-server",
VALID_TURN_SERVER_URL2, &ret);
fail_unless (ret != FALSE);
g_signal_emit_by_name (t->webrtc1, "add-turn-server",
INVALID_TURN_SERVER_URL1, &ret);
fail_unless (ret != TRUE);
g_signal_emit_by_name (t->webrtc1, "add-turn-server",
INVALID_TURN_SERVER_URL2, &ret);
fail_unless (ret != TRUE);
g_signal_emit_by_name (t->webrtc1, "add-turn-server",
INVALID_TURN_SERVER_URL3, &ret);
fail_unless (ret != TRUE);
test_webrtc_free (t);
}
GST_END_TEST;
GST_START_TEST (test_data_channel_recreate_offer)
{
GstHarness *h;
GstWebRTCDataChannel *channel;
GstPromise *promise;
const GstStructure *s;
GstPromiseResult res;
GstPad *pad;
h = gst_harness_new_with_padnames ("webrtcbin", "sink_0", NULL);
add_audio_test_src_harness (h, 0xDEADBEEF);
g_signal_emit_by_name (h->element, "create-data-channel", "label", NULL,
&channel);
fail_unless (GST_IS_WEBRTC_DATA_CHANNEL (channel));
pad = gst_element_get_static_pad (h->element, "sink_0");
fail_unless (pad != NULL);
promise = gst_promise_new ();
g_signal_emit_by_name (h->element, "create-offer", NULL, promise);
res = gst_promise_wait (promise);
fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
s = gst_promise_get_reply (promise);
fail_unless (s != NULL);
gst_promise_unref (promise);
promise = gst_promise_new ();
g_signal_emit_by_name (h->element, "create-offer", NULL, promise);
res = gst_promise_wait (promise);
fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
s = gst_promise_get_reply (promise);
fail_unless (s != NULL);
gst_promise_unref (promise);
gst_object_unref (pad);
gst_object_unref (channel);
gst_harness_teardown (h);
}
GST_END_TEST;
static void
validate_msid (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
char **expected_msid = 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_msid = FALSE;
char *prev_msid = NULL;
int j;
for (j = 0; j < gst_sdp_media_attributes_len (media); j++) {
const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, j);
const char *start;
if (!attr->value)
continue;
start = strstr (attr->value, "msid:");
if (start) {
const char *end;
char *msid;
start += strlen ("msid:");
end = strstr (start, " ");
msid = g_strndup (start, end - start);
fail_unless (end, "Invalid msid attribute");
fail_if (have_msid && g_strcmp0 (prev_msid, msid) != 0,
"different values for multiple msid values at mline %u, "
"prev msid %s, msid %s", i, prev_msid, msid);
have_msid = TRUE;
fail_unless_equals_string (msid, expected_msid[i]);
g_clear_pointer (&prev_msid, g_free);
prev_msid = msid;
}
}
g_clear_pointer (&prev_msid, g_free);
fail_unless (have_msid, "no msid attribute in media %u", i);
}
}
static void
_pad_added_src_check_msid (struct test_webrtc *t, GstElement * element,
GstPad * pad, gpointer user_data)
{
const char *expected_msid = user_data;
char *msid;
if (GST_PAD_DIRECTION (pad) != GST_PAD_SRC)
return;
g_object_get (pad, "msid", &msid, NULL);
fail_unless_equals_string (msid, expected_msid);
g_clear_pointer (&msid, g_free);
test_webrtc_signal_state_unlocked (t, STATE_CUSTOM);
}
GST_START_TEST (test_msid)
{
struct test_webrtc *t = create_audio_test ();
VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
NULL, NULL);
guint media_format_count[] = { 1, 5 };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, &no_duplicate_payloads);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&media_formats);
const gchar *expected_offer_msid[] = { "a1", "a1", };
VAL_SDP_INIT (offer_msid, validate_msid, expected_offer_msid, &count);
const gchar *expected_offer_setup[] = { "actpass", "actpass", };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&offer_msid);
const gchar *expected_offer_direction[] = { "sendrecv", "sendrecv", };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_setup[] = { "active", "active", };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&count);
const gchar *expected_answer_direction[] = { "recvonly", "recvonly", };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
GstPad *pad;
GstHarness *src;
GstElement *rtpbin2;
t->on_pad_added = _pad_added_src_check_msid;
t->pad_added_data = (gpointer) "a1";
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);
src = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
add_audio_test_src_harness (src, 0x12345678);
t->harnesses = g_list_prepend (t->harnesses, src);
pad = gst_element_get_static_pad (t->webrtc1, "sink_0");
g_object_set (pad, "msid", "a1", NULL);
gst_clear_object (&pad);
pad = gst_element_get_static_pad (t->webrtc1, "sink_1");
g_object_set (pad, "msid", "a1", NULL);
gst_clear_object (&pad);
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);
while (TRUE) {
gst_harness_push_from_src (src);
if (test_webrtc_check_for_state_mask (t, 1 << STATE_CUSTOM))
break;
g_usleep (10 * 1000);
}
test_webrtc_free (t);
}
GST_END_TEST;
static void
_check_ice_end_of_candidates (struct test_webrtc *t, GstElement * element,
guint mlineindex, gchar * candidate, GstElement * other, gpointer user_data)
{
gint *end_count = user_data;
if (!candidate || candidate[0] == '\0') {
g_atomic_int_inc (end_count);
}
}
static void
sdp_media_has_end_of_candidates (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
guint 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_string (gst_sdp_media_get_attribute_val_n (media,
"end-of-candidates", 0), "");
fail_unless (gst_sdp_media_get_attribute_val_n (media, "end-of-candidates",
1) == NULL);
}
}
GST_START_TEST (test_ice_end_of_candidates)
{
struct test_webrtc *t = create_audio_test ();
GstWebRTCSessionDescription *local_desc;
gint end_candidate_count = 0;
VAL_SDP_INIT (offer, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
VAL_SDP_INIT (answer, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
t->on_ice_candidate = _check_ice_end_of_candidates;
t->ice_candidate_data = &end_candidate_count;
test_validate_sdp (t, &offer, &answer);
test_webrtc_wait_for_ice_gathering_complete (t);
fail_unless_equals_int (end_candidate_count, 2);
g_object_get (t->webrtc1, "current-local-description", &local_desc, NULL);
sdp_media_has_end_of_candidates (t, t->webrtc1, local_desc, NULL);
gst_webrtc_session_description_free (local_desc);
test_webrtc_free (t);
}
GST_END_TEST;
static void
_set_setup_session_attr_on_answer (struct test_webrtc *t, GstElement * element,
GstPromise * promise, gpointer user_data)
{
GstSDPMessage *sdp;
GstSDPMessage *modified_sdp = NULL;
const GstSDPMedia *media;
GstSDPMedia *modified_media;
const gchar *attr;
if (TEST_IS_OFFER_ELEMENT (t, element))
return;
sdp = t->answer_desc->sdp;
media = gst_sdp_message_get_media (sdp, 0);
attr = gst_sdp_media_get_attribute_val (media, "setup");
/* Remove the setup attribute from first media */
gst_sdp_media_copy (media, &modified_media);
for (unsigned index = 0;
index < gst_sdp_media_attributes_len (modified_media); index++) {
const GstSDPAttribute *current =
gst_sdp_media_get_attribute (modified_media, index);
if (!g_str_equal (current->key, "setup"))
continue;
gst_sdp_media_remove_attribute (modified_media, index);
break;
}
gst_sdp_message_copy (sdp, &modified_sdp);
/* Add session-level setup attribute to modified answer */
gst_sdp_message_add_attribute (modified_sdp, "setup", attr);
/* Replace first media of answer with a media without session attribute */
gst_sdp_message_remove_media (modified_sdp, 0);
gst_sdp_message_add_media (modified_sdp, modified_media);
gst_sdp_media_free (modified_media);
gst_sdp_message_free (sdp);
t->answer_desc->sdp = modified_sdp;
}
static void
_offer_created_do_nothing (struct test_webrtc *t, GstElement * element,
GstPromise * promise, gpointer user_data)
{
}
GST_START_TEST (test_sdp_session_setup_attribute)
{
struct test_webrtc *t = create_audio_test ();
t->on_offer_created = _offer_created_do_nothing;
t->on_answer_created = _set_setup_session_attr_on_answer;
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);
test_webrtc_wait_for_state_mask (t, 1 << STATE_ANSWER_SET);
test_webrtc_wait_for_ice_gathering_complete (t);
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_stats_with_stream);
tcase_add_test (tc, test_audio);
tcase_add_test (tc, test_ice_port_restriction);
tcase_add_test (tc, test_audio_video);
tcase_add_test (tc, test_media_direction);
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);
tcase_add_test (tc, test_dual_audio);
tcase_add_test (tc, test_duplicate_nego);
tcase_add_test (tc, test_renego_add_stream);
tcase_add_test (tc, test_bundle_renego_add_stream);
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);
tcase_add_test (tc, test_reject_request_pad);
tcase_add_test (tc, test_reject_create_offer);
tcase_add_test (tc, test_reject_create_offer_mline_locked_no_caps);
tcase_add_test (tc, test_reject_set_description);
tcase_add_test (tc, test_force_second_media);
tcase_add_test (tc, test_codec_preferences_caps);
tcase_add_test (tc, test_codec_preferences_negotiation_sinkpad);
tcase_add_test (tc, test_codec_preferences_negotiation_srcpad);
tcase_add_test (tc, test_codec_preferences_in_on_new_transceiver);
tcase_add_test (tc, test_codec_preferences_no_duplicate_extmaps);
tcase_add_test (tc, test_codec_preferences_incompatible_extmaps);
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);
tcase_add_test (tc, test_simulcast);
tcase_add_test (tc, test_simulcast_fec_rtx);
tcase_add_test (tc, test_bundle_multiple_media_rtx_payload_mapping);
tcase_add_test (tc, test_invalid_add_media_in_answer);
tcase_add_test (tc, test_add_turn_server);
tcase_add_test (tc, test_msid);
tcase_add_test (tc, test_ice_end_of_candidates);
tcase_add_test (tc, test_sdp_session_setup_attribute);
if (sctpenc && sctpdec) {
tcase_add_test (tc, test_data_channel_create);
tcase_add_test (tc, test_data_channel_create_two_channels);
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_close);
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);
tcase_add_test (tc, test_renego_stream_add_data_channel);
tcase_add_test (tc, test_renego_data_channel_add_stream);
tcase_add_test (tc, test_renego_stream_data_channel_add_stream);
tcase_add_test (tc, test_data_channel_recreate_offer);
} else {
GST_WARNING ("Some required elements were not found. "
"All datachannel tests 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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