Add fuzzer for FlexfecSender.

webrtc/test/fuzzers/flexfec_sender_fuzzer.cc

Index: webrtc/test/fuzzers/flexfec_sender_fuzzer.cc
diff --git a/webrtc/test/fuzzers/flexfec_sender_fuzzer.cc b/webrtc/test/fuzzers/flexfec_sender_fuzzer.cc
new file mode 100644
index 0000000000000000000000000000000000000000..bc486dda07d56b77f6b10f7ee25f96de7bce2294
--- /dev/null
+++ b/webrtc/test/fuzzers/flexfec_sender_fuzzer.cc
@@ -0,0 +1,68 @@
+/*
+ *  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 "webrtc/modules/rtp_rtcp/include/flexfec_sender.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kFlexfecPayloadType = 123;
+constexpr uint32_t kMediaSsrc = 1234;
+constexpr uint32_t kFlexfecSsrc = 5678;
+const std::vector<RtpExtension> kNoRtpHeaderExtensions;
+
+}  // namespace
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  size_t i = 0;
+  if (size < 5) {
+    return;
+  }
+
+  SimulatedClock clock(1 + data[i++]);
+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
+                       kNoRtpHeaderExtensions, &clock);
+  FecProtectionParams params = {
+      data[i++], static_cast<int>(data[i++] % 100),
+      data[i++] <= 127 ? kFecMaskRandom : kFecMaskBursty};
+  sender.SetFecParameters(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.AddRtpPacketAndGenerateFec(rtp_packet);
+    if (sender.FecAvailable()) {
+      std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets =
+          sender.GetFecPackets();
+    }
+  }
+}
+
+}  // namespace webrtc