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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-21 11:44:51 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-21 11:44:51 +0000
commit9e3c08db40b8916968b9f30096c7be3f00ce9647 (patch)
treea68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/sdk/media_constraints_unittest.cc
parentInitial commit. (diff)
downloadthunderbird-9e3c08db40b8916968b9f30096c7be3f00ce9647.tar.xz
thunderbird-9e3c08db40b8916968b9f30096c7be3f00ce9647.zip
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/sdk/media_constraints_unittest.cc')
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diff --git a/third_party/libwebrtc/sdk/media_constraints_unittest.cc b/third_party/libwebrtc/sdk/media_constraints_unittest.cc
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+++ b/third_party/libwebrtc/sdk/media_constraints_unittest.cc
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+/*
+ * 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