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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 &&
31 a.media_config.enable_dscp == b.media_config.enable_dscp &&
32 a.media_config.video.enable_cpu_overuse_detection ==
33 b.media_config.video.enable_cpu_overuse_detection &&
34 a.media_config.video.disable_prerenderer_smoothing ==
35 b.media_config.video.disable_prerenderer_smoothing &&
36 a.media_config.video.suspend_below_min_bitrate ==
37 b.media_config.video.suspend_below_min_bitrate;
htaa2a49d92016-03-04 02:51:39 -080038}
39
40TEST(MediaConstraintsInterface, CopyConstraintsIntoRtcConfiguration) {
41 FakeConstraints constraints;
42 PeerConnectionInterface::RTCConfiguration old_configuration;
43 PeerConnectionInterface::RTCConfiguration configuration;
44
45 CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
46 EXPECT_TRUE(Matches(old_configuration, configuration));
47
48 constraints.SetMandatory(MediaConstraintsInterface::kEnableIPv6, "true");
49 CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
50 EXPECT_FALSE(configuration.disable_ipv6);
51 constraints.SetMandatory(MediaConstraintsInterface::kEnableIPv6, "false");
52 CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
53 EXPECT_TRUE(configuration.disable_ipv6);
54
55 constraints.SetMandatory(MediaConstraintsInterface::kScreencastMinBitrate,
56 27);
57 CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
58 EXPECT_TRUE(configuration.screencast_min_bitrate);
59 EXPECT_EQ(27, *(configuration.screencast_min_bitrate));
60
61 // An empty set of constraints will not overwrite
62 // values that are already present.
63 constraints = FakeConstraints();
64 configuration = old_configuration;
Oskar Sundbom36f8f3e2017-11-16 10:54:27 +010065 configuration.enable_dtls_srtp = true;
htaa2a49d92016-03-04 02:51:39 -080066 configuration.audio_jitter_buffer_max_packets = 34;
67 CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
68 EXPECT_EQ(34, configuration.audio_jitter_buffer_max_packets);
69 ASSERT_TRUE(configuration.enable_dtls_srtp);
70 EXPECT_TRUE(*(configuration.enable_dtls_srtp));
71}
72
73} // namespace
74
75} // namespace webrtc