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https://github.com/mollyim/webrtc.git
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765024e67b
Bug: None Change-Id: I95edb5482bfc9cfc7241963bbe43a3873aa814ad Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335143 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41633}
75 lines
2.9 KiB
C++
75 lines
2.9 KiB
C++
/*
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* Copyright (c) 2024 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <stddef.h>
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#include <stdint.h>
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#include "api/video/video_frame_type.h"
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#include "modules/rtp_rtcp/source/rtp_format.h"
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#include "modules/rtp_rtcp/source/rtp_format_h264.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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#include "rtc_base/checks.h"
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#include "test/fuzzers/fuzz_data_helper.h"
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namespace webrtc {
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void FuzzOneInput(const uint8_t* data, size_t size) {
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test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size));
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RtpPacketizer::PayloadSizeLimits limits;
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limits.max_payload_len = 1200;
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// Read uint8_t to be sure reduction_lens are much smaller than
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// max_payload_len and thus limits structure is valid.
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limits.first_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
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limits.last_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
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limits.single_packet_reduction_len =
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fuzz_input.ReadOrDefaultValue<uint8_t>(0);
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const H264PacketizationMode kPacketizationModes[] = {
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H264PacketizationMode::NonInterleaved,
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H264PacketizationMode::SingleNalUnit};
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H264PacketizationMode packetization_mode =
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fuzz_input.SelectOneOf(kPacketizationModes);
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// Main function under test: RtpPacketizerH264's constructor.
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RtpPacketizerH264 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),
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limits, packetization_mode);
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size_t num_packets = packetizer.NumPackets();
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if (num_packets == 0) {
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return;
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}
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// When packetization was successful, validate NextPacket function too.
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// While at it, check that packets respect the payload size limits.
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RtpPacketToSend rtp_packet(nullptr);
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// Single packet.
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if (num_packets == 1) {
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RTC_CHECK(packetizer.NextPacket(&rtp_packet));
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RTC_CHECK_LE(rtp_packet.payload_size(),
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limits.max_payload_len - limits.single_packet_reduction_len);
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return;
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}
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// First packet.
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RTC_CHECK(packetizer.NextPacket(&rtp_packet));
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RTC_CHECK_LE(rtp_packet.payload_size(),
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limits.max_payload_len - limits.first_packet_reduction_len);
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// Middle packets.
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for (size_t i = 1; i < num_packets - 1; ++i) {
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rtp_packet.Clear();
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RTC_CHECK(packetizer.NextPacket(&rtp_packet))
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<< "Failed to get packet#" << i;
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RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len)
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<< "Packet #" << i << " exceeds it's limit";
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}
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// Last packet.
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rtp_packet.Clear();
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RTC_CHECK(packetizer.NextPacket(&rtp_packet));
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RTC_CHECK_LE(rtp_packet.payload_size(),
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limits.max_payload_len - limits.last_packet_reduction_len);
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}
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} // namespace webrtc
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