ce9da1636a
Do not fuzz with real renderer because it is merely frame copying and doesn't exercise different control flows. This CL also improved fuzzing performance and fixed a memory leak. Bug: chromium:952606, chromium:1009077, chromium:1009073 Change-Id: I77c6f2581db82bfd95edb18e5f0e541a94c78208 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156620 Commit-Queue: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29522}
90 lines
3.2 KiB
C++
90 lines
3.2 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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#ifndef TEST_FUZZERS_UTILS_RTP_REPLAYER_H_
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#define TEST_FUZZERS_UTILS_RTP_REPLAYER_H_
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#include <stdio.h>
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#include <map>
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/rtc_event_log/rtc_event_log.h"
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#include "api/test/video/function_video_decoder_factory.h"
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#include "api/video_codecs/video_decoder.h"
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#include "call/call.h"
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#include "media/engine/internal_decoder_factory.h"
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#include "rtc_base/fake_clock.h"
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#include "rtc_base/time_utils.h"
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#include "test/null_transport.h"
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#include "test/rtp_file_reader.h"
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#include "test/test_video_capturer.h"
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namespace webrtc {
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namespace test {
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// The RtpReplayer is a utility for fuzzing the RTP/RTCP receiver stack in
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// WebRTC. It achieves this by accepting a set of Receiver configurations and
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// an RtpDump (consisting of both RTP and RTCP packets). The |rtp_dump| is
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// passed in as a buffer to allow simple mutation fuzzing directly on the dump.
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class RtpReplayer final {
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public:
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// Holds all the important stream information required to emulate the WebRTC
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// rtp receival code path.
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struct StreamState {
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test::NullTransport transport;
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std::vector<std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>> sinks;
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std::vector<VideoReceiveStream*> receive_streams;
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std::unique_ptr<VideoDecoderFactory> decoder_factory;
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};
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// Construct an RtpReplayer from a JSON replay configuration file.
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static void Replay(const std::string& replay_config_filepath,
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const uint8_t* rtp_dump_data,
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size_t rtp_dump_size);
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// Construct an RtpReplayer from a set of VideoReceiveStream::Configs. Note
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// the stream_state.transport must be set for each receiver stream.
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static void Replay(
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std::unique_ptr<StreamState> stream_state,
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std::vector<VideoReceiveStream::Config> receive_stream_config,
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const uint8_t* rtp_dump_data,
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size_t rtp_dump_size);
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private:
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// Reads the replay configuration from Json.
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static std::vector<VideoReceiveStream::Config> ReadConfigFromFile(
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const std::string& replay_config,
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Transport* transport);
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// Configures the stream state based on the receiver configurations.
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static void SetupVideoStreams(
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std::vector<VideoReceiveStream::Config>* receive_stream_configs,
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StreamState* stream_state,
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Call* call);
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// Creates a new RtpReader which can read the RtpDump
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static std::unique_ptr<test::RtpFileReader> CreateRtpReader(
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const uint8_t* rtp_dump_data,
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size_t rtp_dump_size);
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// Replays each packet to from the RtpDump.
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static void ReplayPackets(rtc::FakeClock* clock,
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Call* call,
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test::RtpFileReader* rtp_reader);
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}; // class RtpReplayer
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} // namespace test
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} // namespace webrtc
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#endif // TEST_FUZZERS_UTILS_RTP_REPLAYER_H_
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