9db8b88bd4
This change adds a new way for test code to serialize the important information from a VideoReceiveStream::Config so that it can be stored as configuration data for WebRTC fuzzers. This code isn't included in the object itself as it is only going to be used to generate new configurations for the fuzzer each time a new error_correction or video format is added to WebRTC. Bug: webrtc:10117 Change-Id: I9b6fb8e0345890ab16f6d319d91e4e316d1f2888 Reviewed-on: https://webrtc-review.googlesource.com/c/116920 Commit-Queue: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26255}
69 lines
2.9 KiB
C++
69 lines
2.9 KiB
C++
/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "test/call_config_utils.h"
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#include "call/video_receive_stream.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace test {
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TEST(CallConfigUtils, MarshalUnmarshalProcessSameObject) {
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VideoReceiveStream::Config recv_config(nullptr);
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VideoReceiveStream::Decoder decoder;
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decoder.payload_type = 10;
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decoder.video_format.name = "test";
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decoder.video_format.parameters["99"] = 98;
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recv_config.decoders.push_back(decoder);
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recv_config.render_delay_ms = 10;
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recv_config.target_delay_ms = 15;
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recv_config.rtp.remote_ssrc = 100;
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recv_config.rtp.local_ssrc = 101;
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recv_config.rtp.rtcp_mode = RtcpMode::kCompound;
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recv_config.rtp.remb = false;
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recv_config.rtp.transport_cc = false;
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recv_config.rtp.nack.rtp_history_ms = 150;
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recv_config.rtp.red_payload_type = 50;
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recv_config.rtp.rtx_ssrc = 1000;
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recv_config.rtp.rtx_associated_payload_types[10] = 10;
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recv_config.rtp.extensions.emplace_back("uri", 128, true);
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VideoReceiveStream::Config unmarshaled_config =
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ParseVideoReceiveStreamJsonConfig(
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nullptr, GenerateVideoReceiveStreamJsonConfig(recv_config));
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EXPECT_EQ(recv_config.decoders[0].payload_type,
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unmarshaled_config.decoders[0].payload_type);
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EXPECT_EQ(recv_config.decoders[0].video_format.name,
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unmarshaled_config.decoders[0].video_format.name);
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EXPECT_EQ(recv_config.decoders[0].video_format.parameters,
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unmarshaled_config.decoders[0].video_format.parameters);
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EXPECT_EQ(recv_config.render_delay_ms, unmarshaled_config.render_delay_ms);
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EXPECT_EQ(recv_config.target_delay_ms, unmarshaled_config.target_delay_ms);
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EXPECT_EQ(recv_config.rtp.remote_ssrc, unmarshaled_config.rtp.remote_ssrc);
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EXPECT_EQ(recv_config.rtp.local_ssrc, unmarshaled_config.rtp.local_ssrc);
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EXPECT_EQ(recv_config.rtp.rtcp_mode, unmarshaled_config.rtp.rtcp_mode);
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EXPECT_EQ(recv_config.rtp.remb, unmarshaled_config.rtp.remb);
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EXPECT_EQ(recv_config.rtp.transport_cc, unmarshaled_config.rtp.transport_cc);
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EXPECT_EQ(recv_config.rtp.nack.rtp_history_ms,
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unmarshaled_config.rtp.nack.rtp_history_ms);
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EXPECT_EQ(recv_config.rtp.red_payload_type,
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unmarshaled_config.rtp.red_payload_type);
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EXPECT_EQ(recv_config.rtp.rtx_ssrc, unmarshaled_config.rtp.rtx_ssrc);
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EXPECT_EQ(recv_config.rtp.rtx_associated_payload_types,
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unmarshaled_config.rtp.rtx_associated_payload_types);
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EXPECT_EQ(recv_config.rtp.extensions, recv_config.rtp.extensions);
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}
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} // namespace test
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} // namespace webrtc
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