From 9db8b88bd44a43118c73302bac883ce55abfe933 Mon Sep 17 00:00:00 2001 From: Benjamin Wright Date: Mon, 14 Jan 2019 15:30:20 -0800 Subject: [PATCH] Added JSON generator for VideoReceiveStream::Config objects. 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 Reviewed-by: Sebastian Jansson Cr-Commit-Position: refs/heads/master@{#26255} --- test/BUILD.gn | 3 ++ test/call_config_utils.cc | 54 ++++++++++++++++++++++++ test/call_config_utils.h | 4 ++ test/call_config_utils_unittest.cc | 68 ++++++++++++++++++++++++++++++ 4 files changed, 129 insertions(+) create mode 100644 test/call_config_utils_unittest.cc diff --git a/test/BUILD.gn b/test/BUILD.gn index 773e25f66c..a21637fd7c 100644 --- a/test/BUILD.gn +++ b/test/BUILD.gn @@ -327,6 +327,7 @@ if (rtc_include_tests) { rtc_test("test_support_unittests") { deps = [ + ":call_config_utils", ":direct_transport", ":fake_video_codecs", ":fileutils", @@ -345,6 +346,7 @@ if (rtc_include_tests) { "../api/video:builtin_video_bitrate_allocator_factory", "../api/video:video_frame", "../api/video:video_frame_i420", + "../call:video_stream_api", "../modules/rtp_rtcp:rtp_rtcp", "../modules/video_capture", "../modules/video_coding:simulcast_test_fixture_impl", @@ -358,6 +360,7 @@ if (rtc_include_tests) { "//third_party/abseil-cpp/absl/strings", ] sources = [ + "call_config_utils_unittest.cc", "direct_transport_unittest.cc", "fake_vp8_encoder_unittest.cc", "frame_generator_unittest.cc", diff --git a/test/call_config_utils.cc b/test/call_config_utils.cc index 48d4849a5c..671220b800 100644 --- a/test/call_config_utils.cc +++ b/test/call_config_utils.cc @@ -68,5 +68,59 @@ VideoReceiveStream::Config ParseVideoReceiveStreamJsonConfig( return receive_config; } +Json::Value GenerateVideoReceiveStreamJsonConfig( + const VideoReceiveStream::Config& config) { + Json::Value root_json; + + root_json["decoders"] = Json::Value(Json::arrayValue); + for (const auto& decoder : config.decoders) { + Json::Value decoder_json; + decoder_json["payload_type"] = decoder.payload_type; + decoder_json["payload_name"] = decoder.video_format.name; + decoder_json["codec_params"] = Json::Value(Json::arrayValue); + for (const auto& codec_param_entry : decoder.video_format.parameters) { + Json::Value codec_param_json; + codec_param_json[codec_param_entry.first] = codec_param_entry.second; + decoder_json["codec_params"].append(codec_param_json); + } + root_json["decoders"].append(decoder_json); + } + + Json::Value rtp_json; + rtp_json["remote_ssrc"] = config.rtp.remote_ssrc; + rtp_json["local_ssrc"] = config.rtp.local_ssrc; + rtp_json["rtcp_mode"] = config.rtp.rtcp_mode == RtcpMode::kCompound + ? "RtcpMode::kCompound" + : "RtcpMode::kReducedSize"; + rtp_json["remb"] = config.rtp.remb; + rtp_json["transport_cc"] = config.rtp.transport_cc; + rtp_json["nack"]["rtp_history_ms"] = config.rtp.nack.rtp_history_ms; + rtp_json["ulpfec_payload_type"] = config.rtp.ulpfec_payload_type; + rtp_json["red_payload_type"] = config.rtp.red_payload_type; + rtp_json["rtx_ssrc"] = config.rtp.rtx_ssrc; + rtp_json["rtx_payload_types"] = Json::Value(Json::arrayValue); + + for (auto& kv : config.rtp.rtx_associated_payload_types) { + Json::Value val; + val[std::to_string(kv.first)] = kv.second; + rtp_json["rtx_payload_types"].append(val); + } + + rtp_json["extensions"] = Json::Value(Json::arrayValue); + for (auto& ext : config.rtp.extensions) { + Json::Value ext_json; + ext_json["uri"] = ext.uri; + ext_json["id"] = ext.id; + ext_json["encrypt"] = ext.encrypt; + rtp_json["extensions"].append(ext_json); + } + root_json["rtp"] = rtp_json; + + root_json["render_delay_ms"] = config.render_delay_ms; + root_json["target_delay_ms"] = config.target_delay_ms; + + return root_json; +} + } // namespace test. } // namespace webrtc. diff --git a/test/call_config_utils.h b/test/call_config_utils.h index 246c7f810d..186c67fd86 100644 --- a/test/call_config_utils.h +++ b/test/call_config_utils.h @@ -23,6 +23,10 @@ VideoReceiveStream::Config ParseVideoReceiveStreamJsonConfig( webrtc::Transport* transport, const Json::Value& json); +// Serialize a VideoReceiveStream::Config into a Json object. +Json::Value GenerateVideoReceiveStreamJsonConfig( + const VideoReceiveStream::Config& config); + } // namespace test } // namespace webrtc diff --git a/test/call_config_utils_unittest.cc b/test/call_config_utils_unittest.cc new file mode 100644 index 0000000000..afc48c4c9b --- /dev/null +++ b/test/call_config_utils_unittest.cc @@ -0,0 +1,68 @@ +/* + * 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) { + VideoReceiveStream::Config recv_config(nullptr); + + VideoReceiveStream::Decoder decoder; + decoder.payload_type = 10; + decoder.video_format.name = "test"; + decoder.video_format.parameters["99"] = 98; + recv_config.decoders.push_back(decoder); + recv_config.render_delay_ms = 10; + recv_config.target_delay_ms = 15; + recv_config.rtp.remote_ssrc = 100; + recv_config.rtp.local_ssrc = 101; + recv_config.rtp.rtcp_mode = RtcpMode::kCompound; + recv_config.rtp.remb = false; + recv_config.rtp.transport_cc = 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); + + VideoReceiveStream::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.target_delay_ms, unmarshaled_config.target_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.remb, unmarshaled_config.rtp.remb); + EXPECT_EQ(recv_config.rtp.transport_cc, unmarshaled_config.rtp.transport_cc); + 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