blob: ddf2ca9d3d6d118088a653c5229eac55b1e9540c [file] [log] [blame]
/*
* 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