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https://github.com/mollyim/webrtc.git
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2ee83c1784
Environment includes propagated field trials that can be later passed to RemoteBitrateEstimators member, and would allow not to rely on the global field trial string Bug: webrtc:42220378 Change-Id: Icf75a433c20352b2c22829c2148c92f69a2517aa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349645 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42242}
64 lines
2.3 KiB
C++
64 lines
2.3 KiB
C++
/*
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* Copyright (c) 2015 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 <algorithm>
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#include <cstddef>
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#include <cstdint>
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#include "api/array_view.h"
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#include "api/environment/environment_factory.h"
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#include "api/units/time_delta.h"
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#include "api/units/timestamp.h"
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#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
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#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
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#include "modules/rtp_rtcp/source/byte_io.h"
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#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "system_wrappers/include/clock.h"
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namespace webrtc {
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void FuzzOneInput(const uint8_t* data, size_t size) {
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Timestamp arrival_time = Timestamp::Micros(123'456'789);
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SimulatedClock clock(arrival_time);
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ReceiveSideCongestionController cc(
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CreateEnvironment(&clock),
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/*feedback_sender=*/[](auto...) {},
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/*remb_sender=*/[](auto...) {},
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/*network_state_estimator=*/nullptr);
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RtpHeaderExtensionMap extensions;
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extensions.Register<TransmissionOffset>(1);
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extensions.Register<AbsoluteSendTime>(2);
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extensions.Register<TransportSequenceNumber>(3);
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extensions.Register<TransportSequenceNumberV2>(4);
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RtpPacketReceived rtp_packet(&extensions);
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constexpr int kMinPacketSize = sizeof(uint16_t) + sizeof(uint8_t) + 12;
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const uint8_t* const end_data = data + size;
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while (end_data - data >= kMinPacketSize) {
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size_t packet_size = ByteReader<uint16_t>::ReadBigEndian(data) % 1500;
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data += sizeof(uint16_t);
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arrival_time += TimeDelta::Millis(ByteReader<uint8_t>::ReadBigEndian(data));
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data += sizeof(uint8_t);
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packet_size = std::min<size_t>(end_data - data, packet_size);
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auto raw_packet = rtc::MakeArrayView(data, packet_size);
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data += packet_size;
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if (!rtp_packet.Parse(raw_packet)) {
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continue;
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}
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rtp_packet.set_arrival_time(arrival_time);
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cc.OnReceivedPacket(rtp_packet, MediaType::VIDEO);
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clock.AdvanceTimeMilliseconds(5);
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cc.MaybeProcess();
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}
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}
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} // namespace webrtc
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