From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 19:32:43 +0200 Subject: Adding upstream version 1:115.7.0. Signed-off-by: Daniel Baumann --- .../libwebrtc/sdk/media_constraints_unittest.cc | 56 ++++++++++++++++++++++ 1 file changed, 56 insertions(+) create mode 100644 third_party/libwebrtc/sdk/media_constraints_unittest.cc (limited to 'third_party/libwebrtc/sdk/media_constraints_unittest.cc') diff --git a/third_party/libwebrtc/sdk/media_constraints_unittest.cc b/third_party/libwebrtc/sdk/media_constraints_unittest.cc new file mode 100644 index 0000000000..2d25da03e7 --- /dev/null +++ b/third_party/libwebrtc/sdk/media_constraints_unittest.cc @@ -0,0 +1,56 @@ +/* + * 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 "sdk/media_constraints.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.audio_jitter_buffer_max_packets == + b.audio_jitter_buffer_max_packets && + a.screencast_min_bitrate == b.screencast_min_bitrate && + a.combined_audio_video_bwe == b.combined_audio_video_bwe && + a.media_config == b.media_config; +} + +TEST(MediaConstraints, CopyConstraintsIntoRtcConfiguration) { + const MediaConstraints constraints_empty; + PeerConnectionInterface::RTCConfiguration old_configuration; + PeerConnectionInterface::RTCConfiguration configuration; + + CopyConstraintsIntoRtcConfiguration(&constraints_empty, &configuration); + EXPECT_TRUE(Matches(old_configuration, configuration)); + + const MediaConstraints constraints_screencast( + {MediaConstraints::Constraint(MediaConstraints::kScreencastMinBitrate, + "27")}, + {}); + CopyConstraintsIntoRtcConfiguration(&constraints_screencast, &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. + configuration = old_configuration; + configuration.audio_jitter_buffer_max_packets = 34; + CopyConstraintsIntoRtcConfiguration(&constraints_empty, &configuration); + EXPECT_EQ(34, configuration.audio_jitter_buffer_max_packets); +} + +} // namespace + +} // namespace webrtc -- cgit v1.2.3