Merging upstream version 126.0.

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

6 files changed, 267 insertions, 9 deletions

diff --git a/third_party/libwebrtc/test/fuzzers/BUILD.gn b/third_party/libwebrtc/test/fuzzers/BUILD.gn
index 43e9a5e922..083c20c6f4 100644
--- a/third_party/libwebrtc/test/fuzzers/BUILD.gn
+++ b/third_party/libwebrtc/test/fuzzers/BUILD.gn
@@ -238,6 +238,36 @@ webrtc_fuzzer_test("rtp_packetizer_av1_fuzzer") {
   ]
 }
 
+webrtc_fuzzer_test("rtp_format_h264_fuzzer") {
+  sources = [ "rtp_format_h264_fuzzer.cc" ]
+  deps = [
+    "../../api/video:video_frame_type",
+    "../../modules/rtp_rtcp:rtp_rtcp",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+    "../../rtc_base:checks",
+  ]
+}
+
+webrtc_fuzzer_test("rtp_format_vp8_fuzzer") {
+  sources = [ "rtp_format_vp8_fuzzer.cc" ]
+  deps = [
+    "../../api/video:video_frame_type",
+    "../../modules/rtp_rtcp:rtp_rtcp",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+    "../../rtc_base:checks",
+  ]
+}
+
+webrtc_fuzzer_test("rtp_format_vp9_fuzzer") {
+  sources = [ "rtp_format_vp9_fuzzer.cc" ]
+  deps = [
+    "../../api/video:video_frame_type",
+    "../../modules/rtp_rtcp:rtp_rtcp",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+    "../../rtc_base:checks",
+  ]
+}
+
 webrtc_fuzzer_test("receive_side_congestion_controller_fuzzer") {
   sources = [ "receive_side_congestion_controller_fuzzer.cc" ]
   deps = [
@@ -248,6 +278,7 @@ webrtc_fuzzer_test("receive_side_congestion_controller_fuzzer") {
     "../../modules/rtp_rtcp:rtp_rtcp_format",
     "../../system_wrappers",
   ]
+  seed_corpus = "corpora/receive-side-cc"
 }
 
 rtc_library("audio_decoder_fuzzer") {
@@ -469,6 +500,7 @@ webrtc_fuzzer_test("audio_processing_fuzzer") {
     "../../api:scoped_refptr",
     "../../api/audio:aec3_factory",
     "../../api/audio:echo_detector_creator",
+    "../../api/task_queue",
     "../../api/task_queue:default_task_queue_factory",
     "../../modules/audio_processing",
     "../../modules/audio_processing:api",
@@ -478,11 +510,13 @@ webrtc_fuzzer_test("audio_processing_fuzzer") {
     "../../modules/audio_processing/aec_dump",
     "../../modules/audio_processing/aec_dump:aec_dump_impl",
     "../../rtc_base:macromagic",
-    "../../rtc_base:rtc_task_queue",
     "../../rtc_base:safe_minmax",
     "../../system_wrappers:field_trial",
   ]
-  absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/base:nullability",
+    "//third_party/abseil-cpp/absl/memory",
+  ]
   seed_corpus = "corpora/audio_processing-corpus"
 }
 
diff --git a/third_party/libwebrtc/test/fuzzers/audio_processing_configs_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/audio_processing_configs_fuzzer.cc
index 331a373f4e..93bce2f2e7 100644
--- a/third_party/libwebrtc/test/fuzzers/audio_processing_configs_fuzzer.cc
+++ b/third_party/libwebrtc/test/fuzzers/audio_processing_configs_fuzzer.cc
@@ -11,16 +11,17 @@
 #include <bitset>
 #include <string>
 
+#include "absl/base/nullability.h"
 #include "absl/memory/memory.h"
 #include "api/audio/echo_canceller3_factory.h"
 #include "api/audio/echo_detector_creator.h"
 #include "api/task_queue/default_task_queue_factory.h"
+#include "api/task_queue/task_queue_base.h"
 #include "modules/audio_processing/aec_dump/aec_dump_factory.h"
 #include "modules/audio_processing/include/audio_processing.h"
 #include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/numerics/safe_minmax.h"
-#include "rtc_base/task_queue.h"
 #include "system_wrappers/include/field_trial.h"
 #include "test/fuzzers/audio_processing_fuzzer_helper.h"
 #include "test/fuzzers/fuzz_data_helper.h"
@@ -33,9 +34,10 @@ const std::string kFieldTrialNames[] = {
     "WebRTC-Aec3ShortHeadroomKillSwitch",
 };
 
-rtc::scoped_refptr<AudioProcessing> CreateApm(test::FuzzDataHelper* fuzz_data,
-                                              std::string* field_trial_string,
-                                              rtc::TaskQueue* worker_queue) {
+rtc::scoped_refptr<AudioProcessing> CreateApm(
+    test::FuzzDataHelper* fuzz_data,
+    std::string* field_trial_string,
+    absl::Nonnull<TaskQueueBase*> worker_queue) {
   // Parse boolean values for optionally enabling different
   // configurable public components of APM.
   bool use_ts = fuzz_data->ReadOrDefaultValue(true);
@@ -134,9 +136,10 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
   // for field_trial.h. Hence it's created here and not in CreateApm.
   std::string field_trial_string = "";
 
-  rtc::TaskQueue worker_queue(GetTaskQueueFactory()->CreateTaskQueue(
-      "rtc-low-prio", rtc::TaskQueue::Priority::LOW));
-  auto apm = CreateApm(&fuzz_data, &field_trial_string, &worker_queue);
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> worker_queue =
+      GetTaskQueueFactory()->CreateTaskQueue("rtc-low-prio",
+                                             TaskQueueFactory::Priority::LOW);
+  auto apm = CreateApm(&fuzz_data, &field_trial_string, worker_queue.get());
 
   if (apm) {
     FuzzAudioProcessing(&fuzz_data, std::move(apm));
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/receive-side-cc/testcase-5414098152390656 b/third_party/libwebrtc/test/fuzzers/corpora/receive-side-cc/testcase-5414098152390656
new file mode 100644
index 0000000000..98c423cdc2
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/receive-side-cc/testcase-5414098152390656
diff --git a/third_party/libwebrtc/test/fuzzers/rtp_format_h264_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_format_h264_fuzzer.cc
new file mode 100644
index 0000000000..ddf2ca9d3d
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/rtp_format_h264_fuzzer.cc
@@ -0,0 +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

diff --git a/third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc
new file mode 100644
index 0000000000..c3c055de0f
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc
@@ -0,0 +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

diff --git a/third_party/libwebrtc/test/fuzzers/rtp_format_vp9_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_format_vp9_fuzzer.cc
new file mode 100644
index 0000000000..3b5e67f697
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/rtp_format_vp9_fuzzer.cc
@@ -0,0 +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