Ensure RtpVideoLayersAllocationExtension::Parse validate sanity of the output

This is tested by a simple unit test and a new fuzzer that verify that all that can be parsed also can be written.

Bug: webrtc:12000
Change-Id: I461aedf97d3dec6e8916e72110fa097c3b31c27f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231642
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34986}
This commit is contained in:
Per Kjellander 2021-09-13 12:59:38 +02:00 committed by WebRTC LUCI CQ
parent f1384afda0
commit 52b9e1ecfb
5 changed files with 100 additions and 3 deletions

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@ -289,7 +289,7 @@ bool RtpVideoLayersAllocationExtension::Parse(
if (data.size() == 1 && *read_at == 0) {
allocation->rtp_stream_index = 0;
allocation->resolution_and_frame_rate_is_valid = true;
return true;
return AllocationIsValid(*allocation);
}
// Header byte.
@ -366,7 +366,7 @@ bool RtpVideoLayersAllocationExtension::Parse(
if (read_at == end) {
allocation->resolution_and_frame_rate_is_valid = false;
return true;
return AllocationIsValid(*allocation);
}
if (read_at + 5 * allocation->active_spatial_layers.size() != end) {
@ -383,7 +383,8 @@ bool RtpVideoLayersAllocationExtension::Parse(
layer.frame_rate_fps = *read_at;
++read_at;
}
return true;
return AllocationIsValid(*allocation);
}
size_t RtpVideoLayersAllocationExtension::ValueSize(

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@ -256,5 +256,32 @@ TEST(RtpVideoLayersAllocationExtension, DiscardsOverLargeDataRate) {
EXPECT_FALSE(RtpVideoLayersAllocationExtension::Parse(buffer, &allocation));
}
TEST(RtpVideoLayersAllocationExtension, DiscardsInvalidHeight) {
VideoLayersAllocation written_allocation;
written_allocation.rtp_stream_index = 0;
written_allocation.resolution_and_frame_rate_is_valid = true;
written_allocation.active_spatial_layers = {
{
/*rtp_stream_index*/ 0,
/*spatial_id*/ 0,
/*target_bitrate_per_temporal_layer*/
{DataRate::KilobitsPerSec(25), DataRate::KilobitsPerSec(50)},
/*width*/ 320,
/*height*/ 240,
/*frame_rate_fps*/ 8,
},
};
rtc::Buffer buffer(
RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
ASSERT_TRUE(
RtpVideoLayersAllocationExtension::Write(buffer, written_allocation));
// Modify the height to be invalid.
buffer[buffer.size() - 3] = 0xff;
buffer[buffer.size() - 2] = 0xff;
VideoLayersAllocation allocation;
EXPECT_FALSE(RtpVideoLayersAllocationExtension::Parse(buffer, &allocation));
}
} // namespace
} // namespace webrtc

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@ -559,6 +559,17 @@ webrtc_fuzzer_test("rtp_dependency_descriptor_fuzzer") {
]
}
webrtc_fuzzer_test("rtp_video_layers_allocation_fuzzer") {
sources = [ "rtp_video_layers_allocation_fuzzer.cc" ]
seed_corpus = "corpora/video_layers_allocation-corpus"
deps = [
"../../api:array_view",
"../../api/video:video_layers_allocation",
"../../modules/rtp_rtcp:rtp_rtcp_format",
"../../rtc_base:checks",
]
}
webrtc_fuzzer_test("rtp_frame_reference_finder_fuzzer") {
sources = [ "rtp_frame_reference_finder_fuzzer.cc" ]
deps = [

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@ -0,0 +1 @@
@2?<3F><>

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@ -0,0 +1,57 @@
/*
* Copyright (c) 2021 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 <cstddef>
#include <cstdint>
#include "api/array_view.h"
#include "api/video/video_layers_allocation.h"
#include "modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h"
#include "rtc_base/checks.h"
namespace webrtc {
void FuzzOneInput(const uint8_t* data, size_t size) {
auto raw = rtc::MakeArrayView(data, size);
VideoLayersAllocation allocation1;
if (!RtpVideoLayersAllocationExtension::Parse(raw, &allocation1)) {
// Ignore invalid buffer and move on.
return;
}
// Write parsed allocation back into raw buffer.
size_t value_size = RtpVideoLayersAllocationExtension::ValueSize(allocation1);
// Check `writer` use minimal number of bytes to pack the extension by
// checking it doesn't use more than reader consumed.
RTC_CHECK_LE(value_size, raw.size());
uint8_t some_memory[256];
// An extension may not be larger than 255 bytes since the extension lenght
// field is only one byte.
RTC_CHECK_LT(value_size, 256);
rtc::ArrayView<uint8_t> write_buffer(some_memory, value_size);
RTC_CHECK(
RtpVideoLayersAllocationExtension::Write(write_buffer, allocation1));
// Parse what Write assembled.
// Unlike random input that should always succeed.
VideoLayersAllocation allocation2;
RTC_CHECK(
RtpVideoLayersAllocationExtension::Parse(write_buffer, &allocation2));
RTC_CHECK_EQ(allocation1.rtp_stream_index, allocation2.rtp_stream_index);
RTC_CHECK_EQ(allocation1.resolution_and_frame_rate_is_valid,
allocation2.resolution_and_frame_rate_is_valid);
RTC_CHECK_EQ(allocation1.active_spatial_layers.size(),
allocation2.active_spatial_layers.size());
RTC_CHECK(allocation1 == allocation2);
}
} // namespace webrtc