blob: cf6716b7376f443b23a4e12373e68010805e4413 [file] [log] [blame]
htaa2a49d92016-03-04 02:51:39 -08001/*
2 * Copyright 2016 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020011#include "api/mediaconstraintsinterface.h"
htaa2a49d92016-03-04 02:51:39 -080012
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020013#include "api/test/fakeconstraints.h"
14#include "rtc_base/gunit.h"
htaa2a49d92016-03-04 02:51:39 -080015
16namespace webrtc {
17
18namespace {
19
nissec36b31b2016-04-11 23:25:29 -070020// Checks all settings touched by CopyConstraintsIntoRtcConfiguration,
21// plus audio_jitter_buffer_max_packets.
htaa2a49d92016-03-04 02:51:39 -080022bool Matches(const PeerConnectionInterface::RTCConfiguration& a,
23 const PeerConnectionInterface::RTCConfiguration& b) {
nissec36b31b2016-04-11 23:25:29 -070024 return a.disable_ipv6 == b.disable_ipv6 &&
25 a.audio_jitter_buffer_max_packets ==
htaa2a49d92016-03-04 02:51:39 -080026 b.audio_jitter_buffer_max_packets &&
nissec36b31b2016-04-11 23:25:29 -070027 a.enable_rtp_data_channel == b.enable_rtp_data_channel &&
28 a.screencast_min_bitrate == b.screencast_min_bitrate &&
29 a.combined_audio_video_bwe == b.combined_audio_video_bwe &&
30 a.enable_dtls_srtp == b.enable_dtls_srtp &&
Niels Möller1d7ecd22018-01-18 15:25:12 +010031 a.media_config == b.media_config;
htaa2a49d92016-03-04 02:51:39 -080032}
33
34TEST(MediaConstraintsInterface, CopyConstraintsIntoRtcConfiguration) {
35 FakeConstraints constraints;
36 PeerConnectionInterface::RTCConfiguration old_configuration;
37 PeerConnectionInterface::RTCConfiguration configuration;
38
39 CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
40 EXPECT_TRUE(Matches(old_configuration, configuration));
41
42 constraints.SetMandatory(MediaConstraintsInterface::kEnableIPv6, "true");
43 CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
44 EXPECT_FALSE(configuration.disable_ipv6);
45 constraints.SetMandatory(MediaConstraintsInterface::kEnableIPv6, "false");
46 CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
47 EXPECT_TRUE(configuration.disable_ipv6);
48
49 constraints.SetMandatory(MediaConstraintsInterface::kScreencastMinBitrate,
50 27);
51 CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
52 EXPECT_TRUE(configuration.screencast_min_bitrate);
53 EXPECT_EQ(27, *(configuration.screencast_min_bitrate));
54
55 // An empty set of constraints will not overwrite
56 // values that are already present.
57 constraints = FakeConstraints();
58 configuration = old_configuration;
Oskar Sundbom36f8f3e2017-11-16 10:54:27 +010059 configuration.enable_dtls_srtp = true;
htaa2a49d92016-03-04 02:51:39 -080060 configuration.audio_jitter_buffer_max_packets = 34;
61 CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
62 EXPECT_EQ(34, configuration.audio_jitter_buffer_max_packets);
63 ASSERT_TRUE(configuration.enable_dtls_srtp);
64 EXPECT_TRUE(*(configuration.enable_dtls_srtp));
65}
66
67} // namespace
68
69} // namespace webrtc