Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(449)

Side by Side Diff: webrtc/api/mediaconstraintsinterface_unittest.cc

Issue 1717583002: Non-constraint interfaces for all constrainable interfaces (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Review comments Created 4 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/api/mediaconstraintsinterface.cc ('k') | webrtc/api/peerconnection.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
(Empty)
1 /*
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
11 #include "webrtc/api/mediaconstraintsinterface.h"
12
13 #include "webrtc/api/test/fakeconstraints.h"
14 #include "webrtc/base/gunit.h"
15
16 namespace webrtc {
17
18 namespace {
19
20 bool Matches(const PeerConnectionInterface::RTCConfiguration& a,
21 const PeerConnectionInterface::RTCConfiguration& b) {
22 return a.audio_jitter_buffer_max_packets ==
23 b.audio_jitter_buffer_max_packets &&
24 a.disable_prerenderer_smoothing == b.disable_prerenderer_smoothing;
25 }
26
27 TEST(MediaConstraintsInterface, CopyConstraintsIntoRtcConfiguration) {
28 FakeConstraints constraints;
29 PeerConnectionInterface::RTCConfiguration old_configuration;
30 PeerConnectionInterface::RTCConfiguration configuration;
31
32 CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
33 EXPECT_TRUE(Matches(old_configuration, configuration));
34
35 constraints.SetMandatory(MediaConstraintsInterface::kEnableIPv6, "true");
36 CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
37 EXPECT_FALSE(configuration.disable_ipv6);
38 constraints.SetMandatory(MediaConstraintsInterface::kEnableIPv6, "false");
39 CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
40 EXPECT_TRUE(configuration.disable_ipv6);
41
42 constraints.SetMandatory(MediaConstraintsInterface::kScreencastMinBitrate,
43 27);
44 CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
45 EXPECT_TRUE(configuration.screencast_min_bitrate);
46 EXPECT_EQ(27, *(configuration.screencast_min_bitrate));
47
48 // An empty set of constraints will not overwrite
49 // values that are already present.
50 constraints = FakeConstraints();
51 configuration = old_configuration;
52 configuration.enable_dtls_srtp = rtc::Optional<bool>(true);
53 configuration.audio_jitter_buffer_max_packets = 34;
54 CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
55 EXPECT_EQ(34, configuration.audio_jitter_buffer_max_packets);
56 ASSERT_TRUE(configuration.enable_dtls_srtp);
57 EXPECT_TRUE(*(configuration.enable_dtls_srtp));
58 }
59
60 } // namespace
61
62 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/api/mediaconstraintsinterface.cc ('k') | webrtc/api/peerconnection.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698