add VP8/VP9 packetization fuzzers

and ensure consistent behavior on empty input.

BUG=webrtc:15755

Change-Id: Id70ab5d55251b4dd10eed8ab67ea8e75545a7a8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332740
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41502}

modules/rtp_rtcp/source/rtp_format_vp8.cc

@@ -63,7 +63,9 @@ RtpPacketizerVp8::RtpPacketizerVp8(rtc::ArrayView<const uint8_t> payload,
                                    const RTPVideoHeaderVP8& hdr_info)
     : hdr_(BuildHeader(hdr_info)), remaining_payload_(payload) {
   limits.max_payload_len -= hdr_.size();
-  payload_sizes_ = SplitAboutEqually(payload.size(), limits);
+  if (!payload.empty()) {
+    payload_sizes_ = SplitAboutEqually(payload.size(), limits);
+  }
   current_packet_ = payload_sizes_.begin();
 }
 

modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc

@@ -21,6 +21,18 @@ namespace {
 
 constexpr RtpPacketizer::PayloadSizeLimits kNoSizeLimits;
 
+TEST(RtpPacketizerVp8Test, EmptyPayload) {
+  RTPVideoHeaderVP8 hdr_info;
+  hdr_info.InitRTPVideoHeaderVP8();
+  hdr_info.pictureId = 200;
+  RtpFormatVp8TestHelper helper(&hdr_info, /*payload_len=*/30);
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 12;  // Small enough to produce 4 packets.
+  RtpPacketizerVp8 packetizer({}, limits, hdr_info);
+  EXPECT_EQ(packetizer.NumPackets(), 0u);
+}
+
 TEST(RtpPacketizerVp8Test, ResultPacketsAreAlmostEqualSize) {
   RTPVideoHeaderVP8 hdr_info;
   hdr_info.InitRTPVideoHeaderVP8();

modules/rtp_rtcp/source/rtp_format_vp9.cc

@@ -319,8 +319,9 @@ RtpPacketizerVp9::RtpPacketizerVp9(rtc::ArrayView<const uint8_t> payload,
   limits.max_payload_len -= header_size_;
   limits.first_packet_reduction_len += first_packet_extra_header_size_;
   limits.single_packet_reduction_len += first_packet_extra_header_size_;
-
-  payload_sizes_ = SplitAboutEqually(payload.size(), limits);
+  if (!payload.empty()) {
+    payload_sizes_ = SplitAboutEqually(payload.size(), limits);
+  }
   current_packet_ = payload_sizes_.begin();
 }
 

modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc

@@ -186,6 +186,11 @@ class RtpPacketizerVp9Test : public ::testing::Test {
   }
 };
 
+TEST_F(RtpPacketizerVp9Test, EmptyPayload) {
+  RTPVideoHeader video_header;
+  VideoRtpDepacketizerVp9::ParseRtpPayload({}, &video_header);
+}
+
 TEST_F(RtpPacketizerVp9Test, TestEqualSizedMode_OnePacket) {
   const size_t kFrameSize = 25;
   const size_t kPacketSize = 26;

modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc

@@ -99,6 +99,12 @@ Av1Frame ReassembleFrame(rtc::ArrayView<const RtpPayload> rtp_payloads) {
   return Av1Frame(VideoRtpDepacketizerAv1().AssembleFrame(payloads));
 }
 
+TEST(RtpPacketizerAv1Test, EmptyPayload) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  RtpPacketizerAv1 packetizer({}, limits, VideoFrameType::kVideoFrameKey, true);
+  EXPECT_EQ(packetizer.NumPackets(), 0u);
+}
+
 TEST(RtpPacketizerAv1Test, PacketizeOneObuWithoutSizeAndExtension) {
   auto kFrame = BuildAv1Frame({Av1Obu(kAv1ObuTypeFrame)
                                    .WithoutSize()

test/fuzzers/BUILD.gn

@@ -248,6 +248,26 @@ webrtc_fuzzer_test("rtp_format_h264_fuzzer") {
   ]
 }
 
+webrtc_fuzzer_test("rtp_format_vp8_fuzzer") {
+  sources = [ "rtp_format_vp8_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_format_vp9_fuzzer") {
+  sources = [ "rtp_format_vp9_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("receive_side_congestion_controller_fuzzer") {
   sources = [ "receive_side_congestion_controller_fuzzer.cc" ]
   deps = [

test/fuzzers/rtp_format_vp8_fuzzer.cc

@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2024 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_format_vp8.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.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);
+
+  RTPVideoHeaderVP8 hdr_info;
+  hdr_info.InitRTPVideoHeaderVP8();
+  uint16_t picture_id = fuzz_input.ReadOrDefaultValue<uint16_t>(0);
+  hdr_info.pictureId =
+      picture_id >= 0x8000 ? kNoPictureId : picture_id & 0x7fff;
+
+  // Main function under test: RtpPacketizerVp8's constructor.
+  RtpPacketizerVp8 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),
+                              limits, hdr_info);
+
+  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

test/fuzzers/rtp_format_vp9_fuzzer.cc

@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2024 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_format_vp9.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.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);
+
+  RTPVideoHeaderVP9 hdr_info;
+  hdr_info.InitRTPVideoHeaderVP9();
+  uint16_t picture_id = fuzz_input.ReadOrDefaultValue<uint16_t>(0);
+  hdr_info.picture_id =
+      picture_id >= 0x8000 ? kNoPictureId : picture_id & 0x7fff;
+
+  // Main function under test: RtpPacketizerVp9's constructor.
+  RtpPacketizerVp9 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),
+                              limits, hdr_info);
+
+  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