From 765024e67b6cc1aed6adbc69488e691a44c9684e Mon Sep 17 00:00:00 2001 From: "Alfred E. Heggestad" Date: Thu, 18 Jan 2024 17:28:59 +0100 Subject: [PATCH] test: fix fuzzers line-endings MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Bug: None Change-Id: I95edb5482bfc9cfc7241963bbe43a3873aa814ad Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335143 Reviewed-by: Henrik Boström Commit-Queue: Philipp Hancke Reviewed-by: Mirko Bonadei Cr-Commit-Position: refs/heads/main@{#41633} --- test/fuzzers/rtp_format_h264_fuzzer.cc | 150 ++++++++++++------------- test/fuzzers/rtp_format_vp8_fuzzer.cc | 146 ++++++++++++------------ test/fuzzers/rtp_format_vp9_fuzzer.cc | 146 ++++++++++++------------ 3 files changed, 221 insertions(+), 221 deletions(-) diff --git a/test/fuzzers/rtp_format_h264_fuzzer.cc b/test/fuzzers/rtp_format_h264_fuzzer.cc index ddf2ca9d3d..97b0ce2c03 100644 --- a/test/fuzzers/rtp_format_h264_fuzzer.cc +++ b/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/test/fuzzers/rtp_format_vp8_fuzzer.cc b/test/fuzzers/rtp_format_vp8_fuzzer.cc index c3c055de0f..93706e9253 100644 --- a/test/fuzzers/rtp_format_vp8_fuzzer.cc +++ b/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/test/fuzzers/rtp_format_vp9_fuzzer.cc b/test/fuzzers/rtp_format_vp9_fuzzer.cc index 3b5e67f697..d95114eaef 100644 --- a/test/fuzzers/rtp_format_vp9_fuzzer.cc +++ b/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