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-rw-r--r-- | third_party/libwebrtc/rtc_tools/rtp_generator/rtp_generator.h | 124 |
1 files changed, 124 insertions, 0 deletions
diff --git a/third_party/libwebrtc/rtc_tools/rtp_generator/rtp_generator.h b/third_party/libwebrtc/rtc_tools/rtp_generator/rtp_generator.h new file mode 100644 index 0000000000..9a56522c33 --- /dev/null +++ b/third_party/libwebrtc/rtc_tools/rtp_generator/rtp_generator.h @@ -0,0 +1,124 @@ +/* + * 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 RTC_TOOLS_RTP_GENERATOR_RTP_GENERATOR_H_ +#define RTC_TOOLS_RTP_GENERATOR_RTP_GENERATOR_H_ + +#include <memory> +#include <string> +#include <vector> + +#include "api/call/transport.h" +#include "api/media_types.h" +#include "api/rtc_event_log/rtc_event_log.h" +#include "api/task_queue/task_queue_factory.h" +#include "api/video/builtin_video_bitrate_allocator_factory.h" +#include "api/video_codecs/video_decoder_factory.h" +#include "api/video_codecs/video_encoder_factory.h" +#include "call/call.h" +#include "call/rtp_config.h" +#include "call/video_send_stream.h" +#include "media/engine/webrtc_video_engine.h" +#include "test/frame_generator_capturer.h" +#include "test/rtp_file_reader.h" +#include "test/rtp_file_writer.h" +#include "video/config/video_encoder_config.h" + +namespace webrtc { + +// Specifies all the configurable options to pass to the corpus generator. +// If modified please update the JSON parser as well as all. +struct RtpGeneratorOptions { + struct VideoSendStreamConfig { + // The time to record the RtpDump for. + int duration_ms = 10000; + // The video resolution width. + int video_width = 640; + // The video resolution height. + int video_height = 480; + // The video fps. + int video_fps = 24; + // The number of squares to render. + int num_squares = 128; + // The individual RTP configuration. + RtpConfig rtp; + }; + // Multiple senders can be active at once on an rtp channel. + std::vector<VideoSendStreamConfig> video_streams; +}; + +// Attempts to parse RtpGeneratorOptions from a JSON file. Any failures +// will result in absl::nullopt. +absl::optional<RtpGeneratorOptions> ParseRtpGeneratorOptionsFromFile( + const std::string& options_file); + +// The RtpGenerator allows generating of corpus material intended to be +// used by fuzzers. It accepts a simple Json configuration file that allows the +// user to configure the codec, extensions and error correction mechanisms. It +// will then proceed to generate an rtpdump for the specified duration using +// that configuration that can be replayed by the video_replayer. The receiver +// configuration JSON will also be output and can be replayed as follows: +// ./rtp_generator --config_file sender_config --output_rtpdump my.rtpdump +// --output_config receiver_config.json +// ./video_replay --config_file receiver_config.json --output_file my.rtpdump +// +// It achieves this by creating a VideoStreamSender, configuring it as requested +// by the user and then intercepting all outgoing RTP packets and writing them +// to a file instead of out of the network. It then uses this sender +// configuration to generate a mirror receiver configuration that can be read by +// the video_replay program. +class RtpGenerator final : public webrtc::Transport { + public: + // Construct a new RtpGenerator using the specified options. + explicit RtpGenerator(const RtpGeneratorOptions& options); + + RtpGenerator() = delete; + RtpGenerator(const RtpGenerator&) = delete; + RtpGenerator& operator=(const RtpGenerator&) = delete; + + // Cleans up the VideoSendStream. + ~RtpGenerator() override; + // Generates an rtp_dump that is written out to + void GenerateRtpDump(const std::string& rtp_dump_path); + + private: + // webrtc::Transport implementation + // Captured RTP packets are written to the RTPDump file instead of over the + // network. + bool SendRtp(const uint8_t* packet, + size_t length, + const webrtc::PacketOptions& options) override; + // RTCP packets are ignored for now. + bool SendRtcp(const uint8_t* packet, size_t length) override; + // Returns the maximum duration + int GetMaxDuration() const; + // Waits until all video streams have finished. + void WaitUntilAllVideoStreamsFinish(); + // Converts packet data into an RtpPacket. + test::RtpPacket DataToRtpPacket(const uint8_t* packet, size_t packet_len); + + const RtpGeneratorOptions options_; + std::unique_ptr<VideoEncoderFactory> video_encoder_factory_; + std::unique_ptr<VideoDecoderFactory> video_decoder_factory_; + std::unique_ptr<VideoBitrateAllocatorFactory> + video_bitrate_allocator_factory_; + std::unique_ptr<RtcEventLog> event_log_; + std::unique_ptr<Call> call_; + std::unique_ptr<test::RtpFileWriter> rtp_dump_writer_; + std::vector<std::unique_ptr<test::FrameGeneratorCapturer>> frame_generators_; + std::vector<VideoSendStream*> video_send_streams_; + std::vector<uint32_t> durations_ms_; + uint32_t start_ms_ = 0; + std::unique_ptr<TaskQueueFactory> task_queue_; +}; + +} // namespace webrtc + +#endif // RTC_TOOLS_RTP_GENERATOR_RTP_GENERATOR_H_ |