mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-04 23:46:43 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
Philippe Normand 424a78c9b9 webrtcbin: Prevent critical warning when creating an additional data channel 
The max_channels value wasn't clamped to 65534 in all situations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5001>
2023-07-10 14:08:09 +00:00

5957 lines
199 KiB
C
Raw Blame History

/* 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 && 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);
}
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 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);
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);
Reference in a new issue View git blame Copy permalink