Merging upstream version 127.0.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

6 files changed, 246 insertions, 222 deletions

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 <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/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 <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/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 <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