3e70781361
This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
72 lines
2.8 KiB
C++
72 lines
2.8 KiB
C++
/*
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* Copyright 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "api/mediaconstraintsinterface.h"
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#include "absl/types/optional.h"
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#include "api/test/fakeconstraints.h"
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#include "media/base/mediaconfig.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace {
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// Checks all settings touched by CopyConstraintsIntoRtcConfiguration,
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// plus audio_jitter_buffer_max_packets.
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bool Matches(const PeerConnectionInterface::RTCConfiguration& a,
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const PeerConnectionInterface::RTCConfiguration& b) {
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return a.disable_ipv6 == b.disable_ipv6 &&
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a.audio_jitter_buffer_max_packets ==
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b.audio_jitter_buffer_max_packets &&
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a.enable_rtp_data_channel == b.enable_rtp_data_channel &&
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a.screencast_min_bitrate == b.screencast_min_bitrate &&
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a.combined_audio_video_bwe == b.combined_audio_video_bwe &&
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a.enable_dtls_srtp == b.enable_dtls_srtp &&
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a.media_config == b.media_config;
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}
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TEST(MediaConstraintsInterface, CopyConstraintsIntoRtcConfiguration) {
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FakeConstraints constraints;
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PeerConnectionInterface::RTCConfiguration old_configuration;
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PeerConnectionInterface::RTCConfiguration configuration;
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CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
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EXPECT_TRUE(Matches(old_configuration, configuration));
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constraints.SetMandatory(MediaConstraintsInterface::kEnableIPv6, "true");
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CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
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EXPECT_FALSE(configuration.disable_ipv6);
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constraints.SetMandatory(MediaConstraintsInterface::kEnableIPv6, "false");
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CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
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EXPECT_TRUE(configuration.disable_ipv6);
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constraints.SetMandatory(MediaConstraintsInterface::kScreencastMinBitrate,
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27);
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CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
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EXPECT_TRUE(configuration.screencast_min_bitrate);
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EXPECT_EQ(27, *(configuration.screencast_min_bitrate));
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// An empty set of constraints will not overwrite
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// values that are already present.
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constraints = FakeConstraints();
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configuration = old_configuration;
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configuration.enable_dtls_srtp = true;
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configuration.audio_jitter_buffer_max_packets = 34;
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CopyConstraintsIntoRtcConfiguration(&constraints, &configuration);
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EXPECT_EQ(34, configuration.audio_jitter_buffer_max_packets);
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ASSERT_TRUE(configuration.enable_dtls_srtp);
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EXPECT_TRUE(*(configuration.enable_dtls_srtp));
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}
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} // namespace
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} // namespace webrtc
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