| /* | 
 |  *  Copyright 2016 The WebRTC project authors. All Rights Reserved. | 
 |  * | 
 |  *  Use of this source code is governed by a BSD-style license | 
 |  *  that can be found in the LICENSE file in the root of the source | 
 |  *  tree. An additional intellectual property rights grant can be found | 
 |  *  in the file PATENTS.  All contributing project authors may | 
 |  *  be found in the AUTHORS file in the root of the source tree. | 
 |  */ | 
 |  | 
 | #include "webrtc/api/mediaconstraintsinterface.h" | 
 |  | 
 | #include "webrtc/api/test/fakeconstraints.h" | 
 | #include "webrtc/base/gunit.h" | 
 |  | 
 | namespace webrtc { | 
 |  | 
 | namespace { | 
 |  | 
 | // Checks all settings touched by CopyConstraintsIntoRtcConfiguration, | 
 | // plus audio_jitter_buffer_max_packets. | 
 | bool Matches(const PeerConnectionInterface::RTCConfiguration& a, | 
 |              const PeerConnectionInterface::RTCConfiguration& b) { | 
 |   return a.disable_ipv6 == b.disable_ipv6 && | 
 |          a.audio_jitter_buffer_max_packets == | 
 |              b.audio_jitter_buffer_max_packets && | 
 |          a.enable_rtp_data_channel == b.enable_rtp_data_channel && | 
 |          a.screencast_min_bitrate == b.screencast_min_bitrate && | 
 |          a.combined_audio_video_bwe == b.combined_audio_video_bwe && | 
 |          a.enable_dtls_srtp == b.enable_dtls_srtp && | 
 |          a.media_config.enable_dscp == b.media_config.enable_dscp && | 
 |          a.media_config.video.enable_cpu_overuse_detection == | 
 |              b.media_config.video.enable_cpu_overuse_detection && | 
 |          a.media_config.video.disable_prerenderer_smoothing == | 
 |              b.media_config.video.disable_prerenderer_smoothing && | 
 |          a.media_config.video.suspend_below_min_bitrate == | 
 |              b.media_config.video.suspend_below_min_bitrate; | 
 | } | 
 |  | 
 | TEST(MediaConstraintsInterface, CopyConstraintsIntoRtcConfiguration) { | 
 |   FakeConstraints constraints; | 
 |   PeerConnectionInterface::RTCConfiguration old_configuration; | 
 |   PeerConnectionInterface::RTCConfiguration configuration; | 
 |  | 
 |   CopyConstraintsIntoRtcConfiguration(&constraints, &configuration); | 
 |   EXPECT_TRUE(Matches(old_configuration, configuration)); | 
 |  | 
 |   constraints.SetMandatory(MediaConstraintsInterface::kEnableIPv6, "true"); | 
 |   CopyConstraintsIntoRtcConfiguration(&constraints, &configuration); | 
 |   EXPECT_FALSE(configuration.disable_ipv6); | 
 |   constraints.SetMandatory(MediaConstraintsInterface::kEnableIPv6, "false"); | 
 |   CopyConstraintsIntoRtcConfiguration(&constraints, &configuration); | 
 |   EXPECT_TRUE(configuration.disable_ipv6); | 
 |  | 
 |   constraints.SetMandatory(MediaConstraintsInterface::kScreencastMinBitrate, | 
 |                            27); | 
 |   CopyConstraintsIntoRtcConfiguration(&constraints, &configuration); | 
 |   EXPECT_TRUE(configuration.screencast_min_bitrate); | 
 |   EXPECT_EQ(27, *(configuration.screencast_min_bitrate)); | 
 |  | 
 |   // An empty set of constraints will not overwrite | 
 |   // values that are already present. | 
 |   constraints = FakeConstraints(); | 
 |   configuration = old_configuration; | 
 |   configuration.enable_dtls_srtp = rtc::Optional<bool>(true); | 
 |   configuration.audio_jitter_buffer_max_packets = 34; | 
 |   CopyConstraintsIntoRtcConfiguration(&constraints, &configuration); | 
 |   EXPECT_EQ(34, configuration.audio_jitter_buffer_max_packets); | 
 |   ASSERT_TRUE(configuration.enable_dtls_srtp); | 
 |   EXPECT_TRUE(*(configuration.enable_dtls_srtp)); | 
 | } | 
 |  | 
 | }  // namespace | 
 |  | 
 | }  // namespace webrtc |