Files
platform-external-webrtc/pc/media_constraints_interface_unittest.cc
Steve Anton 10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00

72 lines
2.8 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 "api/media_constraints_interface.h"
#include "absl/types/optional.h"
#include "api/test/fake_constraints.h"
#include "media/base/media_config.h"
#include "test/gtest.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 == b.media_config;
}
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 = 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