Fuzz RtpPacketizerAv1
Bug: webrtc:11042
Change-Id: Id44699395f6dee9cb3bde84c936573b65ad0d848
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161009
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30007}
diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
index 068254d..b95773d 100644
--- a/test/fuzzers/BUILD.gn
+++ b/test/fuzzers/BUILD.gn
@@ -240,6 +240,18 @@
seed_corpus = "corpora/rtp-corpus"
}
+webrtc_fuzzer_test("rtp_packetizer_av1_fuzzer") {
+ sources = [
+ "rtp_packetizer_av1_fuzzer.cc",
+ ]
+ deps = [
+ "../../api/video:video_frame_type",
+ "../../modules/rtp_rtcp:rtp_rtcp",
+ "../../modules/rtp_rtcp:rtp_rtcp_format",
+ "../../rtc_base:checks",
+ ]
+}
+
webrtc_fuzzer_test("rtp_header_fuzzer") {
sources = [
"rtp_header_fuzzer.cc",
diff --git a/test/fuzzers/rtp_packetizer_av1_fuzzer.cc b/test/fuzzers/rtp_packetizer_av1_fuzzer.cc
new file mode 100644
index 0000000..5277c10f
--- /dev/null
+++ b/test/fuzzers/rtp_packetizer_av1_fuzzer.cc
@@ -0,0 +1,70 @@
+/*
+ * Copyright (c) 2019 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 <stddef.h>
+#include <stdint.h>
+
+#include "api/video/video_frame_type.h"
+#include "modules/rtp_rtcp/source/rtp_format.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/rtp_packetizer_av1.h"
+#include "rtc_base/checks.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+ test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size));
+
+ RtpPacketizer::PayloadSizeLimits limits;
+ limits.max_payload_len = 1200;
+ // Read uint8_t to be sure reduction_lens are much smaller than
+ // max_payload_len and thus limits structure is valid.
+ limits.first_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
+ limits.last_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
+ limits.single_packet_reduction_len =
+ fuzz_input.ReadOrDefaultValue<uint8_t>(0);
+ const VideoFrameType kFrameTypes[] = {VideoFrameType::kVideoFrameKey,
+ VideoFrameType::kVideoFrameDelta};
+ VideoFrameType frame_type = fuzz_input.SelectOneOf(kFrameTypes);
+
+ // Main function under test: RtpPacketizerAv1's constructor.
+ RtpPacketizerAv1 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),
+ limits, frame_type);
+
+ size_t num_packets = packetizer.NumPackets();
+ if (num_packets == 0) {
+ return;
+ }
+ // When packetization was successful, validate NextPacket function too.
+ // While at it, check that packets respect the payload size limits.
+ RtpPacketToSend rtp_packet(nullptr);
+ // Single packet.
+ if (num_packets == 1) {
+ RTC_CHECK(packetizer.NextPacket(&rtp_packet));
+ RTC_CHECK_LE(rtp_packet.payload_size(),
+ limits.max_payload_len - limits.single_packet_reduction_len);
+ return;
+ }
+ // First packet.
+ RTC_CHECK(packetizer.NextPacket(&rtp_packet));
+ RTC_CHECK_LE(rtp_packet.payload_size(),
+ limits.max_payload_len - limits.first_packet_reduction_len);
+ // Middle packets.
+ for (size_t i = 1; i < num_packets - 1; ++i) {
+ RTC_CHECK(packetizer.NextPacket(&rtp_packet))
+ << "Failed to get packet#" << i;
+ RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len)
+ << "Packet #" << i << " exceeds it's limit";
+ }
+ // Last packet.
+ RTC_CHECK(packetizer.NextPacket(&rtp_packet));
+ RTC_CHECK_LE(rtp_packet.payload_size(),
+ limits.max_payload_len - limits.last_packet_reduction_len);
+}
+} // namespace webrtc
