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