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 |