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-rw-r--r--third_party/libwebrtc/test/fuzzers/BUILD.gn38
-rw-r--r--third_party/libwebrtc/test/fuzzers/audio_processing_configs_fuzzer.cc17
-rw-r--r--third_party/libwebrtc/test/fuzzers/corpora/receive-side-cc/testcase-5414098152390656bin0 -> 87 bytes
-rw-r--r--third_party/libwebrtc/test/fuzzers/rtp_format_h264_fuzzer.cc75
-rw-r--r--third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc73
-rw-r--r--third_party/libwebrtc/test/fuzzers/rtp_format_vp9_fuzzer.cc73
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
Binary files differ
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
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-22 23:27:21 +0000