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test: fix fuzzers line-endings

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Bug: None
Change-Id: I95edb5482bfc9cfc7241963bbe43a3873aa814ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335143
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41633}
This commit is contained in:
Alfred E. Heggestad 2024-01-18 17:28:59 +01:00 committed by WebRTC LUCI CQ
parent 05a6f3b425
commit 765024e67b
3 changed files with 221 additions and 221 deletions
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150
test/fuzzers/rtp_format_h264_fuzzer.cc
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@ -1,75 +1,75 @@
/* /*
* Copyright (c) 2024 The WebRTC project authors. All Rights Reserved. * Copyright (c) 2024 The WebRTC project authors. All Rights Reserved.
* *
* Use of this source code is governed by a BSD-style license * 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 * that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found * tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may * in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include <stddef.h> #include <stddef.h>
#include <stdint.h> #include <stdint.h>
#include "api/video/video_frame_type.h" #include "api/video/video_frame_type.h"
#include "modules/rtp_rtcp/source/rtp_format.h" #include "modules/rtp_rtcp/source/rtp_format.h"
#include "modules/rtp_rtcp/source/rtp_format_h264.h" #include "modules/rtp_rtcp/source/rtp_format_h264.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h" #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "rtc_base/checks.h" #include "rtc_base/checks.h"
#include "test/fuzzers/fuzz_data_helper.h" #include "test/fuzzers/fuzz_data_helper.h"
namespace webrtc { namespace webrtc {
void FuzzOneInput(const uint8_t* data, size_t size) { void FuzzOneInput(const uint8_t* data, size_t size) {
test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size)); test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size));
RtpPacketizer::PayloadSizeLimits limits; RtpPacketizer::PayloadSizeLimits limits;
limits.max_payload_len = 1200; limits.max_payload_len = 1200;
// Read uint8_t to be sure reduction_lens are much smaller than // Read uint8_t to be sure reduction_lens are much smaller than
// max_payload_len and thus limits structure is valid. // max_payload_len and thus limits structure is valid.
limits.first_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0); limits.first_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
limits.last_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 = limits.single_packet_reduction_len =
fuzz_input.ReadOrDefaultValue<uint8_t>(0); fuzz_input.ReadOrDefaultValue<uint8_t>(0);
const H264PacketizationMode kPacketizationModes[] = { const H264PacketizationMode kPacketizationModes[] = {
H264PacketizationMode::NonInterleaved, H264PacketizationMode::NonInterleaved,
H264PacketizationMode::SingleNalUnit}; H264PacketizationMode::SingleNalUnit};
H264PacketizationMode packetization_mode = H264PacketizationMode packetization_mode =
fuzz_input.SelectOneOf(kPacketizationModes); fuzz_input.SelectOneOf(kPacketizationModes);
// Main function under test: RtpPacketizerH264's constructor. // Main function under test: RtpPacketizerH264's constructor.
RtpPacketizerH264 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()), RtpPacketizerH264 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),
limits, packetization_mode); limits, packetization_mode);
size_t num_packets = packetizer.NumPackets(); size_t num_packets = packetizer.NumPackets();
if (num_packets == 0) { if (num_packets == 0) {
return; return;
} }
// When packetization was successful, validate NextPacket function too. // When packetization was successful, validate NextPacket function too.
// While at it, check that packets respect the payload size limits. // While at it, check that packets respect the payload size limits.
RtpPacketToSend rtp_packet(nullptr); RtpPacketToSend rtp_packet(nullptr);
// Single packet. // Single packet.
if (num_packets == 1) { if (num_packets == 1) {
RTC_CHECK(packetizer.NextPacket(&rtp_packet)); RTC_CHECK(packetizer.NextPacket(&rtp_packet));
RTC_CHECK_LE(rtp_packet.payload_size(), RTC_CHECK_LE(rtp_packet.payload_size(),
limits.max_payload_len - limits.single_packet_reduction_len); limits.max_payload_len - limits.single_packet_reduction_len);
return; return;
} }
// First packet. // First packet.
RTC_CHECK(packetizer.NextPacket(&rtp_packet)); RTC_CHECK(packetizer.NextPacket(&rtp_packet));
RTC_CHECK_LE(rtp_packet.payload_size(), RTC_CHECK_LE(rtp_packet.payload_size(),
limits.max_payload_len - limits.first_packet_reduction_len); limits.max_payload_len - limits.first_packet_reduction_len);
// Middle packets. // Middle packets.
for (size_t i = 1; i < num_packets - 1; ++i) { for (size_t i = 1; i < num_packets - 1; ++i) {
rtp_packet.Clear(); rtp_packet.Clear();
RTC_CHECK(packetizer.NextPacket(&rtp_packet)) RTC_CHECK(packetizer.NextPacket(&rtp_packet))
<< "Failed to get packet#" << i; << "Failed to get packet#" << i;
RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len) RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len)
<< "Packet #" << i << " exceeds it's limit"; << "Packet #" << i << " exceeds it's limit";
} }
// Last packet. // Last packet.
rtp_packet.Clear(); rtp_packet.Clear();
RTC_CHECK(packetizer.NextPacket(&rtp_packet)); RTC_CHECK(packetizer.NextPacket(&rtp_packet));
RTC_CHECK_LE(rtp_packet.payload_size(), RTC_CHECK_LE(rtp_packet.payload_size(),
limits.max_payload_len - limits.last_packet_reduction_len); limits.max_payload_len - limits.last_packet_reduction_len);
} }
} // namespace webrtc } // namespace webrtc

146
test/fuzzers/rtp_format_vp8_fuzzer.cc
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@ -1,73 +1,73 @@
/* /*
* Copyright (c) 2024 The WebRTC project authors. All Rights Reserved. * Copyright (c) 2024 The WebRTC project authors. All Rights Reserved.
* *
* Use of this source code is governed by a BSD-style license * 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 * that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found * tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may * in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include <stddef.h> #include <stddef.h>
#include <stdint.h> #include <stdint.h>
#include "api/video/video_frame_type.h" #include "api/video/video_frame_type.h"
#include "modules/rtp_rtcp/source/rtp_format.h" #include "modules/rtp_rtcp/source/rtp_format.h"
#include "modules/rtp_rtcp/source/rtp_format_vp8.h" #include "modules/rtp_rtcp/source/rtp_format_vp8.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h" #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "rtc_base/checks.h" #include "rtc_base/checks.h"
#include "test/fuzzers/fuzz_data_helper.h" #include "test/fuzzers/fuzz_data_helper.h"
namespace webrtc { namespace webrtc {
void FuzzOneInput(const uint8_t* data, size_t size) { void FuzzOneInput(const uint8_t* data, size_t size) {
test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size)); test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size));
RtpPacketizer::PayloadSizeLimits limits; RtpPacketizer::PayloadSizeLimits limits;
limits.max_payload_len = 1200; limits.max_payload_len = 1200;
// Read uint8_t to be sure reduction_lens are much smaller than // Read uint8_t to be sure reduction_lens are much smaller than
// max_payload_len and thus limits structure is valid. // max_payload_len and thus limits structure is valid.
limits.first_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0); limits.first_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
limits.last_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 = limits.single_packet_reduction_len =
fuzz_input.ReadOrDefaultValue<uint8_t>(0); fuzz_input.ReadOrDefaultValue<uint8_t>(0);
RTPVideoHeaderVP8 hdr_info; RTPVideoHeaderVP8 hdr_info;
hdr_info.InitRTPVideoHeaderVP8(); hdr_info.InitRTPVideoHeaderVP8();
uint16_t picture_id = fuzz_input.ReadOrDefaultValue<uint16_t>(0); uint16_t picture_id = fuzz_input.ReadOrDefaultValue<uint16_t>(0);
hdr_info.pictureId = hdr_info.pictureId =
picture_id >= 0x8000 ? kNoPictureId : picture_id & 0x7fff; picture_id >= 0x8000 ? kNoPictureId : picture_id & 0x7fff;
// Main function under test: RtpPacketizerVp8's constructor. // Main function under test: RtpPacketizerVp8's constructor.
RtpPacketizerVp8 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()), RtpPacketizerVp8 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),
limits, hdr_info); limits, hdr_info);
size_t num_packets = packetizer.NumPackets(); size_t num_packets = packetizer.NumPackets();
if (num_packets == 0) { if (num_packets == 0) {
return; return;
} }
// When packetization was successful, validate NextPacket function too. // When packetization was successful, validate NextPacket function too.
// While at it, check that packets respect the payload size limits. // While at it, check that packets respect the payload size limits.
RtpPacketToSend rtp_packet(nullptr); RtpPacketToSend rtp_packet(nullptr);
// Single packet. // Single packet.
if (num_packets == 1) { if (num_packets == 1) {
RTC_CHECK(packetizer.NextPacket(&rtp_packet)); RTC_CHECK(packetizer.NextPacket(&rtp_packet));
RTC_CHECK_LE(rtp_packet.payload_size(), RTC_CHECK_LE(rtp_packet.payload_size(),
limits.max_payload_len - limits.single_packet_reduction_len); limits.max_payload_len - limits.single_packet_reduction_len);
return; return;
} }
// First packet. // First packet.
RTC_CHECK(packetizer.NextPacket(&rtp_packet)); RTC_CHECK(packetizer.NextPacket(&rtp_packet));
RTC_CHECK_LE(rtp_packet.payload_size(), RTC_CHECK_LE(rtp_packet.payload_size(),
limits.max_payload_len - limits.first_packet_reduction_len); limits.max_payload_len - limits.first_packet_reduction_len);
// Middle packets. // Middle packets.
for (size_t i = 1; i < num_packets - 1; ++i) { for (size_t i = 1; i < num_packets - 1; ++i) {
RTC_CHECK(packetizer.NextPacket(&rtp_packet)) RTC_CHECK(packetizer.NextPacket(&rtp_packet))
<< "Failed to get packet#" << i; << "Failed to get packet#" << i;
RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len) RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len)
<< "Packet #" << i << " exceeds it's limit"; << "Packet #" << i << " exceeds it's limit";
} }
// Last packet. // Last packet.
RTC_CHECK(packetizer.NextPacket(&rtp_packet)); RTC_CHECK(packetizer.NextPacket(&rtp_packet));
RTC_CHECK_LE(rtp_packet.payload_size(), RTC_CHECK_LE(rtp_packet.payload_size(),
limits.max_payload_len - limits.last_packet_reduction_len); limits.max_payload_len - limits.last_packet_reduction_len);
} }
} // namespace webrtc } // namespace webrtc

146
test/fuzzers/rtp_format_vp9_fuzzer.cc
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@ -1,73 +1,73 @@
/* /*
* Copyright (c) 2024 The WebRTC project authors. All Rights Reserved. * Copyright (c) 2024 The WebRTC project authors. All Rights Reserved.
* *
* Use of this source code is governed by a BSD-style license * 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 * that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found * tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may * in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include <stddef.h> #include <stddef.h>
#include <stdint.h> #include <stdint.h>
#include "api/video/video_frame_type.h" #include "api/video/video_frame_type.h"
#include "modules/rtp_rtcp/source/rtp_format.h" #include "modules/rtp_rtcp/source/rtp_format.h"
#include "modules/rtp_rtcp/source/rtp_format_vp9.h" #include "modules/rtp_rtcp/source/rtp_format_vp9.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h" #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "rtc_base/checks.h" #include "rtc_base/checks.h"
#include "test/fuzzers/fuzz_data_helper.h" #include "test/fuzzers/fuzz_data_helper.h"
namespace webrtc { namespace webrtc {
void FuzzOneInput(const uint8_t* data, size_t size) { void FuzzOneInput(const uint8_t* data, size_t size) {
test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size)); test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size));
RtpPacketizer::PayloadSizeLimits limits; RtpPacketizer::PayloadSizeLimits limits;
limits.max_payload_len = 1200; limits.max_payload_len = 1200;
// Read uint8_t to be sure reduction_lens are much smaller than // Read uint8_t to be sure reduction_lens are much smaller than
// max_payload_len and thus limits structure is valid. // max_payload_len and thus limits structure is valid.
limits.first_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0); limits.first_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
limits.last_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 = limits.single_packet_reduction_len =
fuzz_input.ReadOrDefaultValue<uint8_t>(0); fuzz_input.ReadOrDefaultValue<uint8_t>(0);
RTPVideoHeaderVP9 hdr_info; RTPVideoHeaderVP9 hdr_info;
hdr_info.InitRTPVideoHeaderVP9(); hdr_info.InitRTPVideoHeaderVP9();
uint16_t picture_id = fuzz_input.ReadOrDefaultValue<uint16_t>(0); uint16_t picture_id = fuzz_input.ReadOrDefaultValue<uint16_t>(0);
hdr_info.picture_id = hdr_info.picture_id =
picture_id >= 0x8000 ? kNoPictureId : picture_id & 0x7fff; picture_id >= 0x8000 ? kNoPictureId : picture_id & 0x7fff;
// Main function under test: RtpPacketizerVp9's constructor. // Main function under test: RtpPacketizerVp9's constructor.
RtpPacketizerVp9 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()), RtpPacketizerVp9 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),
limits, hdr_info); limits, hdr_info);
size_t num_packets = packetizer.NumPackets(); size_t num_packets = packetizer.NumPackets();
if (num_packets == 0) { if (num_packets == 0) {
return; return;
} }
// When packetization was successful, validate NextPacket function too. // When packetization was successful, validate NextPacket function too.
// While at it, check that packets respect the payload size limits. // While at it, check that packets respect the payload size limits.
RtpPacketToSend rtp_packet(nullptr); RtpPacketToSend rtp_packet(nullptr);
// Single packet. // Single packet.
if (num_packets == 1) { if (num_packets == 1) {
RTC_CHECK(packetizer.NextPacket(&rtp_packet)); RTC_CHECK(packetizer.NextPacket(&rtp_packet));
RTC_CHECK_LE(rtp_packet.payload_size(), RTC_CHECK_LE(rtp_packet.payload_size(),
limits.max_payload_len - limits.single_packet_reduction_len); limits.max_payload_len - limits.single_packet_reduction_len);
return; return;
} }
// First packet. // First packet.
RTC_CHECK(packetizer.NextPacket(&rtp_packet)); RTC_CHECK(packetizer.NextPacket(&rtp_packet));
RTC_CHECK_LE(rtp_packet.payload_size(), RTC_CHECK_LE(rtp_packet.payload_size(),
limits.max_payload_len - limits.first_packet_reduction_len); limits.max_payload_len - limits.first_packet_reduction_len);
// Middle packets. // Middle packets.
for (size_t i = 1; i < num_packets - 1; ++i) { for (size_t i = 1; i < num_packets - 1; ++i) {
RTC_CHECK(packetizer.NextPacket(&rtp_packet)) RTC_CHECK(packetizer.NextPacket(&rtp_packet))
<< "Failed to get packet#" << i; << "Failed to get packet#" << i;
RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len) RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len)
<< "Packet #" << i << " exceeds it's limit"; << "Packet #" << i << " exceeds it's limit";
} }
// Last packet. // Last packet.
RTC_CHECK(packetizer.NextPacket(&rtp_packet)); RTC_CHECK(packetizer.NextPacket(&rtp_packet));
RTC_CHECK_LE(rtp_packet.payload_size(), RTC_CHECK_LE(rtp_packet.payload_size(),
limits.max_payload_len - limits.last_packet_reduction_len); limits.max_payload_len - limits.last_packet_reduction_len);
} }
} // namespace webrtc } // namespace webrtc