| /* |
| * 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. |
| */ |
| |
| #ifndef TEST_FUZZERS_UTILS_RTP_REPLAYER_H_ |
| #define TEST_FUZZERS_UTILS_RTP_REPLAYER_H_ |
| |
| #include <stdio.h> |
| |
| #include <map> |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "api/test/video/function_video_decoder_factory.h" |
| #include "api/video_codecs/video_decoder.h" |
| #include "call/call.h" |
| #include "logging/rtc_event_log/rtc_event_log.h" |
| #include "media/engine/internal_decoder_factory.h" |
| #include "rtc_base/time_utils.h" |
| #include "test/null_transport.h" |
| #include "test/rtp_file_reader.h" |
| #include "test/test_video_capturer.h" |
| #include "test/video_renderer.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| // The RtpReplayer is a utility for fuzzing the RTP/RTCP receiver stack in |
| // WebRTC. It achieves this by accepting a set of Receiver configurations and |
| // an RtpDump (consisting of both RTP and RTCP packets). The |rtp_dump| is |
| // passed in as a buffer to allow simple mutation fuzzing directly on the dump. |
| class RtpReplayer final { |
| public: |
| // Holds all the important stream information required to emulate the WebRTC |
| // rtp receival code path. |
| struct StreamState { |
| test::NullTransport transport; |
| std::vector<std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>> sinks; |
| std::vector<VideoReceiveStream*> receive_streams; |
| std::unique_ptr<VideoDecoderFactory> decoder_factory; |
| }; |
| |
| // Construct an RtpReplayer from a JSON replay configuration file. |
| static void Replay(const std::string& replay_config_filepath, |
| const uint8_t* rtp_dump_data, |
| size_t rtp_dump_size); |
| |
| // Construct an RtpReplayer from a set of VideoReceiveStream::Configs. Note |
| // the stream_state.transport must be set for each receiver stream. |
| static void Replay( |
| std::unique_ptr<StreamState> stream_state, |
| std::vector<VideoReceiveStream::Config> receive_stream_config, |
| const uint8_t* rtp_dump_data, |
| size_t rtp_dump_size); |
| |
| private: |
| // Reads the replay configuration from Json. |
| static std::vector<VideoReceiveStream::Config> ReadConfigFromFile( |
| const std::string& replay_config, |
| Transport* transport); |
| |
| // Configures the stream state based on the receiver configurations. |
| static void SetupVideoStreams( |
| std::vector<VideoReceiveStream::Config>* receive_stream_configs, |
| StreamState* stream_state, |
| Call* call); |
| |
| // Creates a new RtpReader which can read the RtpDump |
| static std::unique_ptr<test::RtpFileReader> CreateRtpReader( |
| const uint8_t* rtp_dump_data, |
| size_t rtp_dump_size); |
| |
| // Replays each packet to from the RtpDump. |
| static void ReplayPackets(Call* call, test::RtpFileReader* rtp_reader); |
| }; // class RtpReplayer |
| |
| } // namespace test |
| } // namespace webrtc |
| |
| #endif // TEST_FUZZERS_UTILS_RTP_REPLAYER_H_ |