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-rw-r--r--third_party/libwebrtc/test/fuzzers/flexfec_sender_fuzzer.cc67
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diff --git a/third_party/libwebrtc/test/fuzzers/flexfec_sender_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/flexfec_sender_fuzzer.cc
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+++ b/third_party/libwebrtc/test/fuzzers/flexfec_sender_fuzzer.cc
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+/*
+ * Copyright (c) 2016 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.
+ */
+
+#include <memory>
+
+#include "modules/rtp_rtcp/include/flexfec_sender.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kFlexfecPayloadType = 123;
+constexpr uint32_t kMediaSsrc = 1234;
+constexpr uint32_t kFlexfecSsrc = 5678;
+const char kNoMid[] = "";
+const std::vector<RtpExtension> kNoRtpHeaderExtensions;
+const std::vector<RtpExtensionSize> kNoRtpHeaderExtensionSizes;
+
+} // namespace
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+ size_t i = 0;
+ if (size < 5 || size > 200) {
+ return;
+ }
+ SimulatedClock clock(1 + data[i++]);
+ FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
+ kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
+ nullptr /* rtp_state */, &clock);
+ FecProtectionParams params = {
+ data[i++], static_cast<int>(data[i++] % 100),
+ data[i++] <= 127 ? kFecMaskRandom : kFecMaskBursty};
+ sender.SetProtectionParameters(params, params);
+ uint16_t seq_num = data[i++];
+
+ while (i + 1 < size) {
+ // Everything past the base RTP header (12 bytes) is payload,
+ // from the perspective of FlexFEC.
+ size_t payload_size = data[i++];
+ if (i + kRtpHeaderSize + payload_size >= size)
+ break;
+ std::unique_ptr<uint8_t[]> packet(
+ new uint8_t[kRtpHeaderSize + payload_size]);
+ memcpy(packet.get(), &data[i], kRtpHeaderSize + payload_size);
+ i += kRtpHeaderSize + payload_size;
+ ByteWriter<uint16_t>::WriteBigEndian(&packet[2], seq_num++);
+ ByteWriter<uint32_t>::WriteBigEndian(&packet[8], kMediaSsrc);
+ RtpPacketToSend rtp_packet(nullptr);
+ if (!rtp_packet.Parse(packet.get(), kRtpHeaderSize + payload_size))
+ break;
+ sender.AddPacketAndGenerateFec(rtp_packet);
+ sender.GetFecPackets();
+ }
+}
+
+} // namespace webrtc