From 8dd16259287f58f9273002717ec4d27e97127719 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Wed, 12 Jun 2024 07:43:14 +0200 Subject: Merging upstream version 127.0. Signed-off-by: Daniel Baumann --- third_party/libwebrtc/test/fuzzers/BUILD.gn | 6 + .../test/fuzzers/h265_depacketizer_fuzzer.cc | 19 +++ .../libwebrtc/test/fuzzers/neteq_signal_fuzzer.cc | 1 - .../test/fuzzers/rtp_format_h264_fuzzer.cc | 150 ++++++++++----------- .../test/fuzzers/rtp_format_vp8_fuzzer.cc | 146 ++++++++++---------- .../test/fuzzers/rtp_format_vp9_fuzzer.cc | 146 ++++++++++---------- 6 files changed, 246 insertions(+), 222 deletions(-) create mode 100644 third_party/libwebrtc/test/fuzzers/h265_depacketizer_fuzzer.cc (limited to 'third_party/libwebrtc/test/fuzzers') diff --git a/third_party/libwebrtc/test/fuzzers/BUILD.gn b/third_party/libwebrtc/test/fuzzers/BUILD.gn index 083c20c6f4..642b0c8e08 100644 --- a/third_party/libwebrtc/test/fuzzers/BUILD.gn +++ b/third_party/libwebrtc/test/fuzzers/BUILD.gn @@ -132,6 +132,11 @@ if (rtc_use_h265) { "../../modules/video_coding/", ] } + + webrtc_fuzzer_test("h265_depacketizer_fuzzer") { + sources = [ "h265_depacketizer_fuzzer.cc" ] + deps = [ "../../modules/rtp_rtcp" ] + } } webrtc_fuzzer_test("forward_error_correction_fuzzer") { @@ -471,6 +476,7 @@ webrtc_fuzzer_test("stun_validator_fuzzer") { webrtc_fuzzer_test("pseudotcp_parser_fuzzer") { sources = [ "pseudotcp_parser_fuzzer.cc" ] deps = [ + "../../p2p:pseudo_tcp", "../../p2p:rtc_p2p", "../../rtc_base:threading", ] diff --git a/third_party/libwebrtc/test/fuzzers/h265_depacketizer_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/h265_depacketizer_fuzzer.cc new file mode 100644 index 0000000000..00025ef887 --- /dev/null +++ b/third_party/libwebrtc/test/fuzzers/h265_depacketizer_fuzzer.cc @@ -0,0 +1,19 @@ +/* + * 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 "modules/rtp_rtcp/source/video_rtp_depacketizer_h265.h" + +namespace webrtc { +void FuzzOneInput(const uint8_t* data, size_t size) { + if (size > 200000) + return; + VideoRtpDepacketizerH265 depacketizer; + depacketizer.Parse(rtc::CopyOnWriteBuffer(data, size)); +} +} // namespace webrtc diff --git a/third_party/libwebrtc/test/fuzzers/neteq_signal_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/neteq_signal_fuzzer.cc index 485c38085e..3b1f70cdb4 100644 --- a/third_party/libwebrtc/test/fuzzers/neteq_signal_fuzzer.cc +++ b/third_party/libwebrtc/test/fuzzers/neteq_signal_fuzzer.cc @@ -179,7 +179,6 @@ void FuzzOneInputTest(const uint8_t* data, size_t size) { // Configure NetEq and the NetEqTest object. NetEqTest::Callbacks callbacks; NetEq::Config config; - config.enable_post_decode_vad = true; config.enable_fast_accelerate = true; auto codecs = NetEqTest::StandardDecoderMap(); // rate_types contains the payload types that will be used for encoding. diff --git a/third_party/libwebrtc/test/fuzzers/rtp_format_h264_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_format_h264_fuzzer.cc index ddf2ca9d3d..97b0ce2c03 100644 --- a/third_party/libwebrtc/test/fuzzers/rtp_format_h264_fuzzer.cc +++ b/third_party/libwebrtc/test/fuzzers/rtp_format_h264_fuzzer.cc @@ -1,75 +1,75 @@ -/* - * 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 -#include - -#include "api/video/video_frame_type.h" -#include "modules/rtp_rtcp/source/rtp_format.h" -#include "modules/rtp_rtcp/source/rtp_format_h264.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(0); - limits.last_packet_reduction_len = fuzz_input.ReadOrDefaultValue(0); - limits.single_packet_reduction_len = - fuzz_input.ReadOrDefaultValue(0); - const H264PacketizationMode kPacketizationModes[] = { - H264PacketizationMode::NonInterleaved, - H264PacketizationMode::SingleNalUnit}; - - H264PacketizationMode packetization_mode = - fuzz_input.SelectOneOf(kPacketizationModes); - - // Main function under test: RtpPacketizerH264's constructor. - RtpPacketizerH264 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()), - limits, packetization_mode); - - 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) { - rtp_packet.Clear(); - 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. - rtp_packet.Clear(); - RTC_CHECK(packetizer.NextPacket(&rtp_packet)); - RTC_CHECK_LE(rtp_packet.payload_size(), - limits.max_payload_len - limits.last_packet_reduction_len); -} -} // namespace webrtc +/* + * 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 +#include + +#include "api/video/video_frame_type.h" +#include "modules/rtp_rtcp/source/rtp_format.h" +#include "modules/rtp_rtcp/source/rtp_format_h264.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(0); + limits.last_packet_reduction_len = fuzz_input.ReadOrDefaultValue(0); + limits.single_packet_reduction_len = + fuzz_input.ReadOrDefaultValue(0); + const H264PacketizationMode kPacketizationModes[] = { + H264PacketizationMode::NonInterleaved, + H264PacketizationMode::SingleNalUnit}; + + H264PacketizationMode packetization_mode = + fuzz_input.SelectOneOf(kPacketizationModes); + + // Main function under test: RtpPacketizerH264's constructor. + RtpPacketizerH264 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()), + limits, packetization_mode); + + 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) { + rtp_packet.Clear(); + 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. + rtp_packet.Clear(); + RTC_CHECK(packetizer.NextPacket(&rtp_packet)); + RTC_CHECK_LE(rtp_packet.payload_size(), + limits.max_payload_len - limits.last_packet_reduction_len); +} +} // namespace webrtc diff --git a/third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc index c3c055de0f..93706e9253 100644 --- a/third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc +++ b/third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc @@ -1,73 +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 -#include - -#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(0); - limits.last_packet_reduction_len = fuzz_input.ReadOrDefaultValue(0); - limits.single_packet_reduction_len = - fuzz_input.ReadOrDefaultValue(0); - - RTPVideoHeaderVP8 hdr_info; - hdr_info.InitRTPVideoHeaderVP8(); - uint16_t picture_id = fuzz_input.ReadOrDefaultValue(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 +/* + * 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 +#include + +#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(0); + limits.last_packet_reduction_len = fuzz_input.ReadOrDefaultValue(0); + limits.single_packet_reduction_len = + fuzz_input.ReadOrDefaultValue(0); + + RTPVideoHeaderVP8 hdr_info; + hdr_info.InitRTPVideoHeaderVP8(); + uint16_t picture_id = fuzz_input.ReadOrDefaultValue(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 diff --git a/third_party/libwebrtc/test/fuzzers/rtp_format_vp9_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_format_vp9_fuzzer.cc index 3b5e67f697..d95114eaef 100644 --- a/third_party/libwebrtc/test/fuzzers/rtp_format_vp9_fuzzer.cc +++ b/third_party/libwebrtc/test/fuzzers/rtp_format_vp9_fuzzer.cc @@ -1,73 +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 -#include - -#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(0); - limits.last_packet_reduction_len = fuzz_input.ReadOrDefaultValue(0); - limits.single_packet_reduction_len = - fuzz_input.ReadOrDefaultValue(0); - - RTPVideoHeaderVP9 hdr_info; - hdr_info.InitRTPVideoHeaderVP9(); - uint16_t picture_id = fuzz_input.ReadOrDefaultValue(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 +/* + * 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 +#include + +#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(0); + limits.last_packet_reduction_len = fuzz_input.ReadOrDefaultValue(0); + limits.single_packet_reduction_len = + fuzz_input.ReadOrDefaultValue(0); + + RTPVideoHeaderVP9 hdr_info; + hdr_info.InitRTPVideoHeaderVP9(); + uint16_t picture_id = fuzz_input.ReadOrDefaultValue(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 -- cgit v1.2.3