test: fix fuzzers line-endings
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}
diff --git a/test/fuzzers/rtp_format_h264_fuzzer.cc b/test/fuzzers/rtp_format_h264_fuzzer.cc
index ddf2ca9..97b0ce2 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 <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_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<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 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 <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_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<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 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 c3c055d..93706e9 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 <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
+/*
+ * 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
diff --git a/test/fuzzers/rtp_format_vp9_fuzzer.cc b/test/fuzzers/rtp_format_vp9_fuzzer.cc
index 3b5e67f..d95114e 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 <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
+/*
+ * 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
