From d4bcdad26349ab9e03c065442ab7d5d13e166800 Mon Sep 17 00:00:00 2001
From: katrielc <katrielc@webrtc.org>
Date: Thu, 23 Jun 2016 03:50:39 -0700
Subject: [PATCH] Add a libfuzzer for RtpHeaderParser.

NOTRY=true

Review-Url: https://codereview.webrtc.org/2062103002
Cr-Commit-Position: refs/heads/master@{#13271}
---
 .../rtp_rtcp/include/rtp_rtcp_defines.h       |  1 +
 webrtc/modules/rtp_rtcp/source/rtp_sender.cc  |  1 +
 webrtc/test/fuzzers/BUILD.gn                  |  9 ++++
 webrtc/test/fuzzers/rtp_header_fuzzer.cc      | 43 +++++++++++++++++++
 4 files changed, 54 insertions(+)
 create mode 100644 webrtc/test/fuzzers/rtp_header_fuzzer.cc

diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 05320f72e1..90f7784bfb 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -68,6 +68,7 @@ enum RTPExtensionType {
   kRtpExtensionVideoRotation,
   kRtpExtensionTransportSequenceNumber,
   kRtpExtensionPlayoutDelay,
+  kRtpExtensionNumberOfExtensions,
 };
 
 enum RTCPAppSubTypes { kAppSubtypeBwe = 0x00 };
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 9b6b72ae1f..4ee2524abc 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -286,6 +286,7 @@ int32_t RTPSender::RegisterRtpHeaderExtension(RTPExtensionType type,
     case kRtpExtensionTransportSequenceNumber:
       return rtp_header_extension_map_.Register(type, id);
     case kRtpExtensionNone:
+    case kRtpExtensionNumberOfExtensions:
       LOG(LS_ERROR) << "Invalid RTP extension type for registration";
       return -1;
   }
diff --git a/webrtc/test/fuzzers/BUILD.gn b/webrtc/test/fuzzers/BUILD.gn
index 7d318f1796..c1b9ed9985 100644
--- a/webrtc/test/fuzzers/BUILD.gn
+++ b/webrtc/test/fuzzers/BUILD.gn
@@ -99,6 +99,15 @@ webrtc_fuzzer_test("rtp_packet_fuzzer") {
   seed_corpus = "corpora/rtp-corpus"
 }
 
+webrtc_fuzzer_test("rtp_header_fuzzer") {
+  sources = [
+    "rtp_header_fuzzer.cc",
+  ]
+  deps = [
+    "../../modules/rtp_rtcp/",
+  ]
+}
+
 source_set("audio_decoder_fuzzer") {
   public_configs = [ "../..:common_inherited_config" ]
   sources = [
diff --git a/webrtc/test/fuzzers/rtp_header_fuzzer.cc b/webrtc/test/fuzzers/rtp_header_fuzzer.cc
new file mode 100644
index 0000000000..fcc737728c
--- /dev/null
+++ b/webrtc/test/fuzzers/rtp_header_fuzzer.cc
@@ -0,0 +1,43 @@
+/*
+ *  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 <bitset>
+
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size <= 1)
+    return;
+
+  // We decide which header extensions to register by reading one byte
+  // from the beginning of |data| and interpreting it as a bitmask
+  // over the RTPExtensionType enum. That byte shouldn't also be part
+  // of the packet, so we adjust |data| and |size| to remove it.
+  std::bitset<8> extensionMask(data[0]);
+  data++;
+  size--;
+
+  RtpPacketReceived::ExtensionManager extensions;
+  for (int i = 1; i <= kRtpExtensionNumberOfExtensions; i++) {
+    // Skip i=0 which is kRtpExtensionNone i.e. not an actual extension.
+    if (extensionMask[i]) {
+      // We use i as the ID; it's used in negotiation so not relevant.
+      extensions.Register(static_cast<RTPExtensionType>(i), i);
+    }
+  }
+
+  RTPHeader rtp_header;
+  RtpUtility::RtpHeaderParser rtp_parser(data, size);
+  rtp_parser.Parse(&rtp_header, &extensions);
+}
+}  // namespace webrtc