76 lines
3.2 KiB
C++
76 lines
3.2 KiB
C++
|
/*
|
||
|
|
* 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
|