webrtc/api/mediaconstraintsinterface_unittest.cc - Issue 2514883002: Create //webrtc/api:libjingle_peerconnection_api + refactorings.

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Issue 2514883002: Create //webrtc/api:libjingle_peerconnection_api + refactorings. (Closed)

Patch Set: Rebase Created 3 years, 11 months ago

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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 // Checks all settings touched by CopyConstraintsIntoRtcConfiguration,

21 // plus audio_jitter_buffer_max_packets.

22 bool Matches(const PeerConnectionInterface::RTCConfiguration& a,

23 const PeerConnectionInterface::RTCConfiguration& b) {

24 return a.disable_ipv6 == b.disable_ipv6 &&

25 a.audio_jitter_buffer_max_packets ==

26 b.audio_jitter_buffer_max_packets &&

27 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;

38 }

39

40 TEST(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;

65 configuration.enable_dtls_srtp = rtc::Optional<bool>(true);

66 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

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