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-rw-r--r--third_party/libwebrtc/test/call_config_utils_unittest.cc64
1 files changed, 64 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/call_config_utils_unittest.cc b/third_party/libwebrtc/test/call_config_utils_unittest.cc
new file mode 100644
index 0000000000..e010ab6707
--- /dev/null
+++ b/third_party/libwebrtc/test/call_config_utils_unittest.cc
@@ -0,0 +1,64 @@
+/*
+ * Copyright (c) 2019 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 "test/call_config_utils.h"
+
+#include "call/video_receive_stream.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+TEST(CallConfigUtils, MarshalUnmarshalProcessSameObject) {
+ VideoReceiveStreamInterface::Config recv_config(nullptr);
+
+ VideoReceiveStreamInterface::Decoder decoder;
+ decoder.payload_type = 10;
+ decoder.video_format.name = "test";
+ decoder.video_format.parameters["99"] = "b";
+ recv_config.decoders.push_back(decoder);
+ recv_config.render_delay_ms = 10;
+ recv_config.rtp.remote_ssrc = 100;
+ recv_config.rtp.local_ssrc = 101;
+ recv_config.rtp.rtcp_mode = RtcpMode::kCompound;
+ recv_config.rtp.lntf.enabled = false;
+ recv_config.rtp.nack.rtp_history_ms = 150;
+ recv_config.rtp.red_payload_type = 50;
+ recv_config.rtp.rtx_ssrc = 1000;
+ recv_config.rtp.rtx_associated_payload_types[10] = 10;
+ recv_config.rtp.extensions.emplace_back("uri", 128, true);
+
+ VideoReceiveStreamInterface::Config unmarshaled_config =
+ ParseVideoReceiveStreamJsonConfig(
+ nullptr, GenerateVideoReceiveStreamJsonConfig(recv_config));
+
+ EXPECT_EQ(recv_config.decoders[0].payload_type,
+ unmarshaled_config.decoders[0].payload_type);
+ EXPECT_EQ(recv_config.decoders[0].video_format.name,
+ unmarshaled_config.decoders[0].video_format.name);
+ EXPECT_EQ(recv_config.decoders[0].video_format.parameters,
+ unmarshaled_config.decoders[0].video_format.parameters);
+ EXPECT_EQ(recv_config.render_delay_ms, unmarshaled_config.render_delay_ms);
+ EXPECT_EQ(recv_config.rtp.remote_ssrc, unmarshaled_config.rtp.remote_ssrc);
+ EXPECT_EQ(recv_config.rtp.local_ssrc, unmarshaled_config.rtp.local_ssrc);
+ EXPECT_EQ(recv_config.rtp.rtcp_mode, unmarshaled_config.rtp.rtcp_mode);
+ EXPECT_EQ(recv_config.rtp.lntf.enabled, unmarshaled_config.rtp.lntf.enabled);
+ EXPECT_EQ(recv_config.rtp.nack.rtp_history_ms,
+ unmarshaled_config.rtp.nack.rtp_history_ms);
+ EXPECT_EQ(recv_config.rtp.red_payload_type,
+ unmarshaled_config.rtp.red_payload_type);
+ EXPECT_EQ(recv_config.rtp.rtx_ssrc, unmarshaled_config.rtp.rtx_ssrc);
+ EXPECT_EQ(recv_config.rtp.rtx_associated_payload_types,
+ unmarshaled_config.rtp.rtx_associated_payload_types);
+ EXPECT_EQ(recv_config.rtp.extensions, recv_config.rtp.extensions);
+}
+
+} // namespace test
+} // namespace webrtc