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AEC-m and AEC-2 fuzzing.

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Going through the coverage of audio_processing_fuzzer, it was noticed
that it didn't cover AEC-m and AEC-2 code. Therefore this CL adds 2
fuzzer targets that only fuzz the previous generation echo cancellers.

To avoid code duplication, the APM running code was broken out in a
new GN target. We have also changed all fuzzing code to use the
FuzzDataHelper class to avoid manual pointer arithmetic.

Bug: webrtc:7820
Change-Id: Ifea3266e396b487952a736945577fccea15d0e01
Reviewed-on: https://webrtc-review.googlesource.com/36500
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21638}
This commit is contained in:
Alex Loiko
2018-01-16 12:50:34 +01:00
committed by Commit Bot
parent 3ac67a736b
commit ab20a6016c
6 changed files with 237 additions and 270 deletions
Download Patch File Download Diff File Expand all files Collapse all files

20
test/fuzzers/BUILD.gn
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@ -421,14 +421,13 @@ webrtc_fuzzer_test("transport_feedback_packet_loss_tracker_fuzzer") {
"../../voice_engine",
]
}
webrtc_fuzzer_test("audio_processing_fuzzer") {
rtc_static_library("audio_processing_fuzzer_helper") {
sources = [
"audio_processing_fuzzer.cc",
"audio_processing_fuzzer.h",
"audio_processing_fuzzer_configs.cc",
"audio_processing_fuzzer_helper.cc",
"audio_processing_fuzzer_helper.h",
]
deps = [
":fuzz_data_helper",
"../../api:optional",
"../../modules:module_api",
"../../modules/audio_processing",
@ -437,6 +436,17 @@ webrtc_fuzzer_test("audio_processing_fuzzer") {
]
}
webrtc_fuzzer_test("audio_processing_fuzzer") {
sources = [
"audio_processing_configs_fuzzer.cc",
]
deps = [
":audio_processing_fuzzer_helper",
":fuzz_data_helper",
"../../modules/audio_processing",
]
}
webrtc_fuzzer_test("comfort_noise_decoder_fuzzer") {
sources = [
"comfort_noise_decoder_fuzzer.cc",

96
test/fuzzers/audio_processing_configs_fuzzer.cc Normal file
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@ -0,0 +1,96 @@
/*
* Copyright (c) 2017 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 "modules/audio_processing/include/audio_processing.h"
#include "test/fuzzers/audio_processing_fuzzer_helper.h"
#include "test/fuzzers/fuzz_data_helper.h"
namespace webrtc {
namespace {
std::unique_ptr<AudioProcessing> CreateApm(test::FuzzDataHelper* fuzz_data) {
// Parse boolean values for optionally enabling different
// configurable public components of APM.
bool exp_agc = fuzz_data->ReadOrDefaultValue(true);
bool exp_ns = fuzz_data->ReadOrDefaultValue(true);
bool bf = fuzz_data->ReadOrDefaultValue(true);
bool ef = fuzz_data->ReadOrDefaultValue(true);
bool raf = fuzz_data->ReadOrDefaultValue(true);
bool da = fuzz_data->ReadOrDefaultValue(true);
bool ie = fuzz_data->ReadOrDefaultValue(true);
bool red = fuzz_data->ReadOrDefaultValue(true);
bool lc = fuzz_data->ReadOrDefaultValue(true);
bool hpf = fuzz_data->ReadOrDefaultValue(true);
bool aec3 = fuzz_data->ReadOrDefaultValue(true);
bool use_aec = fuzz_data->ReadOrDefaultValue(true);
bool use_aecm = fuzz_data->ReadOrDefaultValue(true);
bool use_agc = fuzz_data->ReadOrDefaultValue(true);
bool use_ns = fuzz_data->ReadOrDefaultValue(true);
bool use_le = fuzz_data->ReadOrDefaultValue(true);
bool use_vad = fuzz_data->ReadOrDefaultValue(true);
bool use_agc_limiter = fuzz_data->ReadOrDefaultValue(true);
// Filter out incompatible settings that lead to CHECK failures.
if (use_aecm && use_aec) {
return nullptr;
}
// Components can be enabled through webrtc::Config and
// webrtc::AudioProcessingConfig.
Config config;
std::unique_ptr<EchoControlFactory> echo_control_factory;
if (aec3) {
echo_control_factory.reset(new EchoCanceller3Factory());
}
config.Set<ExperimentalAgc>(new ExperimentalAgc(exp_agc));
config.Set<ExperimentalNs>(new ExperimentalNs(exp_ns));
if (bf) {
config.Set<Beamforming>(new Beamforming());
}
config.Set<ExtendedFilter>(new ExtendedFilter(ef));
config.Set<RefinedAdaptiveFilter>(new RefinedAdaptiveFilter(raf));
config.Set<DelayAgnostic>(new DelayAgnostic(da));
config.Set<Intelligibility>(new Intelligibility(ie));
std::unique_ptr<AudioProcessing> apm(
AudioProcessingBuilder()
.SetEchoControlFactory(std::move(echo_control_factory))
.Create(config));
webrtc::AudioProcessing::Config apm_config;
apm_config.residual_echo_detector.enabled = red;
apm_config.level_controller.enabled = lc;
apm_config.high_pass_filter.enabled = hpf;
apm->ApplyConfig(apm_config);
apm->echo_cancellation()->Enable(use_aec);
apm->echo_control_mobile()->Enable(use_aecm);
apm->gain_control()->Enable(use_agc);
apm->noise_suppression()->Enable(use_ns);
apm->level_estimator()->Enable(use_le);
apm->voice_detection()->Enable(use_vad);
apm->gain_control()->enable_limiter(use_agc_limiter);
return apm;
}
} // namespace
void FuzzOneInput(const uint8_t* data, size_t size) {
test::FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
auto apm = CreateApm(&fuzz_data);
if (apm) {
FuzzAudioProcessing(&fuzz_data, std::move(apm));
}
}
} // namespace webrtc

156
test/fuzzers/audio_processing_fuzzer.cc
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@ -1,156 +0,0 @@
/*
* Copyright (c) 2017 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 "test/fuzzers/audio_processing_fuzzer.h"
#include <algorithm>
#include <array>
#include <cmath>
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace {
size_t ByteToNativeRate(uint8_t data) {
using Rate = AudioProcessing::NativeRate;
switch (data % 4) {
case 0:
return static_cast<size_t>(Rate::kSampleRate8kHz);
case 1:
return static_cast<size_t>(Rate::kSampleRate16kHz);
case 2:
return static_cast<size_t>(Rate::kSampleRate32kHz);
default:
return static_cast<size_t>(Rate::kSampleRate48kHz);
}
}
template <class T>
bool ParseSequence(size_t size,
const uint8_t** data,
size_t* remaining_size,
T* result_data) {
const size_t data_size_bytes = sizeof(T) * size;
if (data_size_bytes > *remaining_size) {
return false;
}
std::copy(*data, *data + data_size_bytes,
reinterpret_cast<uint8_t*>(result_data));
*data += data_size_bytes;
*remaining_size -= data_size_bytes;
return true;
}
void FuzzAudioProcessing(const uint8_t* data,
size_t size,
bool is_float,
AudioProcessing* apm) {
AudioFrame fixed_frame;
std::array<float, 480> float_frame{};
float* const first_channel = &float_frame[0];
while (size > 0) {
// Decide input/output rate for this iteration.
const auto input_rate_byte = ParseByte(&data, &size);
const auto output_rate_byte = ParseByte(&data, &size);
if (!input_rate_byte || !output_rate_byte) {
return;
}
const auto input_rate_hz = ByteToNativeRate(*input_rate_byte);
const auto output_rate_hz = ByteToNativeRate(*output_rate_byte);
const size_t samples_per_input_channel =
rtc::CheckedDivExact(input_rate_hz, 100ul);
fixed_frame.samples_per_channel_ = samples_per_input_channel;
fixed_frame.sample_rate_hz_ = input_rate_hz;
// Two channels breaks AEC3.
fixed_frame.num_channels_ = 1;
// Fill the arrays with audio samples from the data.
if (is_float) {
if (!ParseSequence(samples_per_input_channel, &data, &size,
&float_frame[0])) {
return;
}
} else if (!ParseSequence(samples_per_input_channel, &data, &size,
fixed_frame.mutable_data())) {
return;
}
// Filter obviously wrong values like inf/nan and values that will
// lead to inf/nan in calculations. 1e6 leads to DCHECKS failing.
for (auto& x : float_frame) {
if (!std::isnormal(x) || std::abs(x) > 1e5) {
x = 0;
}
}
// Make the APM call depending on capture/render mode and float /
// fix interface.
const auto is_capture = ParseBool(&data, &size);
if (!is_capture) {
return;
}
if (*is_capture) {
auto apm_return_code =
is_float ? (apm->ProcessStream(
&first_channel, StreamConfig(input_rate_hz, 1),
StreamConfig(output_rate_hz, 1), &first_channel))
: (apm->ProcessStream(&fixed_frame));
RTC_DCHECK_NE(apm_return_code, AudioProcessing::kBadDataLengthError);
} else {
auto apm_return_code =
is_float ? (apm->ProcessReverseStream(
&first_channel, StreamConfig(input_rate_hz, 1),
StreamConfig(output_rate_hz, 1), &first_channel))
: (apm->ProcessReverseStream(&fixed_frame));
RTC_DCHECK_NE(apm_return_code, AudioProcessing::kBadDataLengthError);
}
}
}
} // namespace
rtc::Optional<bool> ParseBool(const uint8_t** data, size_t* remaining_size) {
if (1 > *remaining_size) {
return rtc::nullopt;
}
auto res = (**data) % 2;
*data += 1;
*remaining_size -= 1;
return res;
}
rtc::Optional<uint8_t> ParseByte(const uint8_t** data, size_t* remaining_size) {
if (1 > *remaining_size) {
return rtc::nullopt;
}
auto res = **data;
*data += 1;
*remaining_size -= 1;
return res;
}
void FuzzAudioProcessing(const uint8_t* data,
size_t size,
std::unique_ptr<AudioProcessing> apm) {
const auto is_float = ParseBool(&data, &size);
if (!is_float) {
return;
}
FuzzAudioProcessing(data, size, *is_float, apm.get());
}
} // namespace webrtc

101
test/fuzzers/audio_processing_fuzzer_configs.cc
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@ -1,101 +0,0 @@
/*
* Copyright (c) 2017 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 "modules/audio_processing/include/audio_processing.h"
#include "test/fuzzers/audio_processing_fuzzer.h"
#include "api/optional.h"
namespace webrtc {
std::unique_ptr<AudioProcessing> CreateAPM(const uint8_t** data,
size_t* remaining_size) {
// Parse boolean values for optionally enabling different
// configurable public components of APM.
auto exp_agc = ParseBool(data, remaining_size);
auto exp_ns = ParseBool(data, remaining_size);
auto bf = ParseBool(data, remaining_size);
auto ef = ParseBool(data, remaining_size);
auto raf = ParseBool(data, remaining_size);
auto da = ParseBool(data, remaining_size);
auto ie = ParseBool(data, remaining_size);
auto red = ParseBool(data, remaining_size);
auto lc = ParseBool(data, remaining_size);
auto hpf = ParseBool(data, remaining_size);
auto aec3 = ParseBool(data, remaining_size);
auto use_aec = ParseBool(data, remaining_size);
auto use_aecm = ParseBool(data, remaining_size);
auto use_agc = ParseBool(data, remaining_size);
auto use_ns = ParseBool(data, remaining_size);
auto use_le = ParseBool(data, remaining_size);
auto use_vad = ParseBool(data, remaining_size);
auto use_agc_limiter = ParseBool(data, remaining_size);
if (!(exp_agc && exp_ns && bf && ef && raf && da && ie && red && lc && hpf &&
aec3 && use_aec && use_aecm && use_agc && use_ns && use_le && use_vad &&
use_agc_limiter)) {
return nullptr;
}
// Filter out incompatible settings that lead to CHECK failures.
if (*use_aecm && *use_aec) {
return nullptr;
}
// Components can be enabled through webrtc::Config and
// webrtc::AudioProcessingConfig.
Config config;
std::unique_ptr<EchoControlFactory> echo_control_factory;
if (*aec3) {
echo_control_factory.reset(new EchoCanceller3Factory());
}
config.Set<ExperimentalAgc>(new ExperimentalAgc(*exp_agc));
config.Set<ExperimentalNs>(new ExperimentalNs(*exp_ns));
if (*bf) {
config.Set<Beamforming>(new Beamforming());
}
config.Set<ExtendedFilter>(new ExtendedFilter(*ef));
config.Set<RefinedAdaptiveFilter>(new RefinedAdaptiveFilter(*raf));
config.Set<DelayAgnostic>(new DelayAgnostic(*da));
config.Set<Intelligibility>(new Intelligibility(*ie));
std::unique_ptr<AudioProcessing> apm(
AudioProcessingBuilder()
.SetEchoControlFactory(std::move(echo_control_factory))
.Create(config));
webrtc::AudioProcessing::Config apm_config;
apm_config.residual_echo_detector.enabled = *red;
apm_config.level_controller.enabled = *lc;
apm_config.high_pass_filter.enabled = *hpf;
apm->ApplyConfig(apm_config);
apm->echo_cancellation()->Enable(*use_aec);
apm->echo_control_mobile()->Enable(*use_aecm);
apm->gain_control()->Enable(*use_agc);
apm->noise_suppression()->Enable(*use_ns);
apm->level_estimator()->Enable(*use_le);
apm->voice_detection()->Enable(*use_vad);
apm->gain_control()->enable_limiter(*use_agc_limiter);
return apm;
}
void FuzzOneInput(const uint8_t* data, size_t size) {
auto apm = CreateAPM(&data, &size);
if (apm) {
FuzzAudioProcessing(data, size, std::move(apm));
}
}
} // namespace webrtc

120
test/fuzzers/audio_processing_fuzzer_helper.cc Normal file
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@ -0,0 +1,120 @@
/*
* Copyright (c) 2017 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 "test/fuzzers/audio_processing_fuzzer_helper.h"
#include <algorithm>
#include <array>
#include <cmath>
#include <limits>
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace {
void GenerateFloatFrame(test::FuzzDataHelper* fuzz_data,
size_t input_rate,
size_t num_channels,
float* const* float_frames) {
const size_t samples_per_input_channel =
rtc::CheckedDivExact(input_rate, 100ul);
RTC_DCHECK_LE(samples_per_input_channel, 480);
for (size_t i = 0; i < num_channels; ++i) {
for (size_t j = 0; j < samples_per_input_channel; ++j) {
float_frames[i][j] =
static_cast<float>(fuzz_data->ReadOrDefaultValue<int16_t>(0)) /
static_cast<float>(std::numeric_limits<int16_t>::max());
}
}
}
void GenerateFixedFrame(test::FuzzDataHelper* fuzz_data,
size_t input_rate,
size_t num_channels,
AudioFrame* fixed_frame) {
const size_t samples_per_input_channel =
rtc::CheckedDivExact(input_rate, 100ul);
fixed_frame->samples_per_channel_ = samples_per_input_channel;
fixed_frame->sample_rate_hz_ = input_rate;
fixed_frame->num_channels_ = num_channels;
RTC_DCHECK_LE(samples_per_input_channel * num_channels,
AudioFrame::kMaxDataSizeSamples);
for (size_t i = 0; i < samples_per_input_channel * num_channels; ++i) {
fixed_frame->mutable_data()[i] = fuzz_data->ReadOrDefaultValue(0);
}
}
} // namespace
void FuzzAudioProcessing(test::FuzzDataHelper* fuzz_data,
std::unique_ptr<AudioProcessing> apm) {
AudioFrame fixed_frame;
std::array<float, 480> float_frame1;
std::array<float, 480> float_frame2;
std::array<float* const, 2> float_frame_ptrs = {
&float_frame1[0], &float_frame2[0],
};
float* const* ptr_to_float_frames = &float_frame_ptrs[0];
using Rate = AudioProcessing::NativeRate;
const Rate rate_kinds[] = {Rate::kSampleRate8kHz, Rate::kSampleRate16kHz,
Rate::kSampleRate32kHz, Rate::kSampleRate48kHz};
// We may run out of fuzz data in the middle of a loop iteration. In
// that case, default values will be used for the rest of that
// iteration.
while (fuzz_data->CanReadBytes(1)) {
const bool is_float = fuzz_data->ReadOrDefaultValue(true);
// Decide input/output rate for this iteration.
const auto input_rate =
static_cast<size_t>(fuzz_data->SelectOneOf(rate_kinds));
const auto output_rate =
static_cast<size_t>(fuzz_data->SelectOneOf(rate_kinds));
const bool num_channels = fuzz_data->ReadOrDefaultValue(true) ? 2 : 1;
const uint8_t stream_delay = fuzz_data->ReadOrDefaultValue(0);
// API call needed for AEC-2 and AEC-m to run.
apm->set_stream_delay_ms(stream_delay);
// Make the APM call depending on capture/render mode and float /
// fix interface.
const bool is_capture = fuzz_data->ReadOrDefaultValue(true);
// Fill the arrays with audio samples from the data.
int apm_return_code = AudioProcessing::Error::kNoError;
if (is_float) {
GenerateFloatFrame(fuzz_data, input_rate, num_channels,
ptr_to_float_frames);
if (is_capture) {
apm_return_code = apm->ProcessStream(
ptr_to_float_frames, StreamConfig(input_rate, num_channels),
StreamConfig(output_rate, num_channels), ptr_to_float_frames);
} else {
apm_return_code = apm->ProcessReverseStream(
ptr_to_float_frames, StreamConfig(input_rate, 1),
StreamConfig(output_rate, 1), ptr_to_float_frames);
}
} else {
GenerateFixedFrame(fuzz_data, input_rate, num_channels, &fixed_frame);
if (is_capture) {
apm_return_code = apm->ProcessStream(&fixed_frame);
} else {
apm_return_code = apm->ProcessReverseStream(&fixed_frame);
}
}
RTC_DCHECK_NE(apm_return_code, AudioProcessing::kBadDataLengthError);
}
}
} // namespace webrtc

14
test/fuzzers/audio_processing_fuzzer.h → test/fuzzers/audio_processing_fuzzer_helper.h
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@ -8,20 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef TEST_FUZZERS_AUDIO_PROCESSING_FUZZER_H_
#define TEST_FUZZERS_AUDIO_PROCESSING_FUZZER_H_
#ifndef TEST_FUZZERS_AUDIO_PROCESSING_FUZZER_HELPER_H_
#define TEST_FUZZERS_AUDIO_PROCESSING_FUZZER_HELPER_H_
#include <memory>
#include "modules/audio_processing/include/audio_processing.h"
#include "test/fuzzers/fuzz_data_helper.h"
namespace webrtc {
rtc::Optional<bool> ParseBool(const uint8_t** data, size_t* remaining_size);
rtc::Optional<uint8_t> ParseByte(const uint8_t** data, size_t* remaining_size);
void FuzzAudioProcessing(const uint8_t* data,
size_t size,
void FuzzAudioProcessing(test::FuzzDataHelper* fuzz_data,
std::unique_ptr<AudioProcessing> apm);
} // namespace webrtc
#endif // TEST_FUZZERS_AUDIO_PROCESSING_FUZZER_H_
#endif // TEST_FUZZERS_AUDIO_PROCESSING_FUZZER_HELPER_H_