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dbbf413085
Local media SSRC is mandatory, but let's give it a default value to make tests less brittle. Bug: chromium:1015256 Change-Id: If7f6505482d90651bc58d9b358290c4d43487f4e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157421 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29534}
52 lines
1.6 KiB
C++
52 lines
1.6 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 "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
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#include "modules/rtp_rtcp/source/rtcp_receiver.h"
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#include "rtc_base/checks.h"
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#include "system_wrappers/include/clock.h"
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namespace webrtc {
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namespace {
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constexpr int kRtcpIntervalMs = 1000;
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// RTCP is typically sent over UDP, which has a maximum payload length
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// of 65535 bytes. We err on the side of caution and check a bit above that.
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constexpr size_t kMaxInputLenBytes = 66000;
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class NullModuleRtpRtcp : public RTCPReceiver::ModuleRtpRtcp {
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public:
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void SetTmmbn(std::vector<rtcp::TmmbItem>) override {}
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void OnRequestSendReport() override {}
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void OnReceivedNack(const std::vector<uint16_t>&) override {}
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void OnReceivedRtcpReportBlocks(const ReportBlockList&) override {}
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};
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} // namespace
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void FuzzOneInput(const uint8_t* data, size_t size) {
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if (size > kMaxInputLenBytes) {
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return;
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}
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NullModuleRtpRtcp rtp_rtcp_module;
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SimulatedClock clock(1234);
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RtpRtcp::Configuration config;
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config.clock = &clock;
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config.rtcp_report_interval_ms = kRtcpIntervalMs;
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config.local_media_ssrc = 1;
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RTCPReceiver receiver(config, &rtp_rtcp_module);
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receiver.IncomingPacket(data, size);
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}
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} // namespace webrtc
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