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Unified Diff: webrtc/api/mediaconstraintsinterface_unittest.cc

Issue 2514883002: Create //webrtc/api:libjingle_peerconnection_api + refactorings. (Closed)
Patch Set: Rebase Created 3 years, 11 months ago
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Index: webrtc/api/mediaconstraintsinterface_unittest.cc
diff --git a/webrtc/api/mediaconstraintsinterface_unittest.cc b/webrtc/api/mediaconstraintsinterface_unittest.cc
deleted file mode 100644
index dcf4bb7fde700207f4565ea1dafbd38f6824348c..0000000000000000000000000000000000000000
--- a/webrtc/api/mediaconstraintsinterface_unittest.cc
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- * 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
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