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/*
* Copyright (C) 2021 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at:
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*
*/
/**
* NOTE
* 1) The input to AudioFlinger binder calls are fuzzed in this fuzzer
* 2) AudioFlinger crashes due to the fuzzer are detected by the
Binder DeathRecipient, where the fuzzer aborts if AudioFlinger dies
*/
#include <android_audio_policy_configuration_V7_0-enums.h>
#include <android/content/AttributionSourceState.h>
#include <binder/IServiceManager.h>
#include <binder/MemoryDealer.h>
#include <media/AidlConversion.h>
#include <media/AudioEffect.h>
#include <media/AudioRecord.h>
#include <media/AudioSystem.h>
#include <media/AudioTrack.h>
#include <media/IAudioFlinger.h>
#include "fuzzer/FuzzedDataProvider.h"
#define MAX_STRING_LENGTH 256
#define MAX_ARRAY_LENGTH 256
constexpr int32_t kMinSampleRateHz = 4000;
constexpr int32_t kMaxSampleRateHz = 192000;
constexpr int32_t kSampleRateUnspecified = 0;
using namespace std;
using namespace android;
namespace xsd {
using namespace ::android::audio::policy::configuration::V7_0;
}
using android::content::AttributionSourceState;
constexpr audio_unique_id_use_t kUniqueIds[] = {
AUDIO_UNIQUE_ID_USE_UNSPECIFIED, AUDIO_UNIQUE_ID_USE_SESSION, AUDIO_UNIQUE_ID_USE_MODULE,
AUDIO_UNIQUE_ID_USE_EFFECT, AUDIO_UNIQUE_ID_USE_PATCH, AUDIO_UNIQUE_ID_USE_OUTPUT,
AUDIO_UNIQUE_ID_USE_INPUT, AUDIO_UNIQUE_ID_USE_CLIENT, AUDIO_UNIQUE_ID_USE_MAX,
};
constexpr audio_mode_t kModes[] = {
AUDIO_MODE_INVALID, AUDIO_MODE_CURRENT, AUDIO_MODE_NORMAL, AUDIO_MODE_RINGTONE,
AUDIO_MODE_IN_CALL, AUDIO_MODE_IN_COMMUNICATION, AUDIO_MODE_CALL_SCREEN,
AUDIO_MODE_CALL_REDIRECT, AUDIO_MODE_COMMUNICATION_REDIRECT};
constexpr audio_session_t kSessionId[] = {AUDIO_SESSION_NONE, AUDIO_SESSION_OUTPUT_STAGE,
AUDIO_SESSION_DEVICE};
constexpr audio_encapsulation_mode_t kEncapsulation[] = {
AUDIO_ENCAPSULATION_MODE_NONE,
AUDIO_ENCAPSULATION_MODE_ELEMENTARY_STREAM,
AUDIO_ENCAPSULATION_MODE_HANDLE,
};
constexpr audio_port_role_t kPortRoles[] = {
AUDIO_PORT_ROLE_NONE,
AUDIO_PORT_ROLE_SOURCE,
AUDIO_PORT_ROLE_SINK,
};
constexpr audio_port_type_t kPortTypes[] = {
AUDIO_PORT_TYPE_NONE,
AUDIO_PORT_TYPE_DEVICE,
AUDIO_PORT_TYPE_MIX,
AUDIO_PORT_TYPE_SESSION,
};
template <typename T, typename X, typename FUNC>
std::vector<T> getFlags(const xsdc_enum_range<X>& range, const FUNC& func,
const std::string& findString = {},
const std::set<X>& excludedValues = {}) {
std::vector<T> vec;
for (const auto &xsdEnumVal : range) {
T enumVal;
std::string enumString = toString(xsdEnumVal);
if (enumString.find(findString) != std::string::npos &&
(excludedValues.find(xsdEnumVal) == excludedValues.end()) &&
func(enumString.c_str(), &enumVal)) {
vec.push_back(enumVal);
}
}
return vec;
}
static const std::vector<audio_stream_type_t> kStreamtypes =
getFlags<audio_stream_type_t, xsd::AudioStreamType, decltype(audio_stream_type_from_string)>(
xsdc_enum_range<xsd::AudioStreamType>{}, audio_stream_type_from_string);
/**
* AudioFormat - AUDIO_FORMAT_HE_AAC_V1 and AUDIO_FORMAT_HE_AAC_V2
* are excluded from kFormats[] in order to avoid the abort triggered
* for these two types of AudioFormat in
* AidlConversion::legacy2aidl_audio_format_t_AudioFormatDescription()
*/
static const std::vector<audio_format_t> kFormats =
getFlags<audio_format_t, xsd::AudioFormat, decltype(audio_format_from_string)>(
xsdc_enum_range<xsd::AudioFormat>{}, audio_format_from_string, {},
{xsd::AudioFormat::AUDIO_FORMAT_HE_AAC_V1,
xsd::AudioFormat::AUDIO_FORMAT_HE_AAC_V2});
/**
* AudioChannelMask - AUDIO_CHANNEL_IN_6
* is excluded from kChannelMasks[] in order to avoid the abort triggered
* for this type of AudioChannelMask in
* AidlConversion::legacy2aidl_audio_channel_mask_t_AudioChannelLayout()
*/
static const std::vector<audio_channel_mask_t> kChannelMasks =
getFlags<audio_channel_mask_t, xsd::AudioChannelMask,
decltype(audio_channel_mask_from_string)>(
xsdc_enum_range<xsd::AudioChannelMask>{}, audio_channel_mask_from_string, {},
{xsd::AudioChannelMask::AUDIO_CHANNEL_IN_6});
static const std::vector<audio_usage_t> kUsages =
getFlags<audio_usage_t, xsd::AudioUsage, decltype(audio_usage_from_string)>(
xsdc_enum_range<xsd::AudioUsage>{}, audio_usage_from_string);
static const std::vector<audio_content_type_t> kContentType =
getFlags<audio_content_type_t, xsd::AudioContentType, decltype(audio_content_type_from_string)>(
xsdc_enum_range<xsd::AudioContentType>{}, audio_content_type_from_string);
static const std::vector<audio_source_t> kInputSources =
getFlags<audio_source_t, xsd::AudioSource, decltype(audio_source_from_string)>(
xsdc_enum_range<xsd::AudioSource>{}, audio_source_from_string);
static const std::vector<audio_gain_mode_t> kGainModes =
getFlags<audio_gain_mode_t, xsd::AudioGainMode, decltype(audio_gain_mode_from_string)>(
xsdc_enum_range<xsd::AudioGainMode>{}, audio_gain_mode_from_string);
/**
* AudioDevice - AUDIO_DEVICE_IN_AMBIENT and AUDIO_DEVICE_IN_COMMUNICATION
* are excluded from kDevices[] in order to avoid the abort triggered
* for these two types of AudioDevice in
* AidlConversion::aidl2legacy_AudioDeviceDescription_audio_devices_t()
*/
static const std::vector<audio_devices_t> kDevices =
getFlags<audio_devices_t, xsd::AudioDevice, decltype(audio_device_from_string)>(
xsdc_enum_range<xsd::AudioDevice>{}, audio_device_from_string, {},
{xsd::AudioDevice::AUDIO_DEVICE_IN_AMBIENT,
xsd::AudioDevice::AUDIO_DEVICE_IN_COMMUNICATION});
static const std::vector<audio_input_flags_t> kInputFlags =
getFlags<audio_input_flags_t, xsd::AudioInOutFlag, decltype(audio_input_flag_from_string)>(
xsdc_enum_range<xsd::AudioInOutFlag>{}, audio_input_flag_from_string, "_INPUT_");
static const std::vector<audio_output_flags_t> kOutputFlags =
getFlags<audio_output_flags_t, xsd::AudioInOutFlag, decltype(audio_output_flag_from_string)>(
xsdc_enum_range<xsd::AudioInOutFlag>{}, audio_output_flag_from_string, "_OUTPUT_");
template <typename T, size_t size>
T getValue(FuzzedDataProvider *fdp, const T (&arr)[size]) {
return arr[fdp->ConsumeIntegralInRange<int32_t>(0, size - 1)];
}
template <typename T>
T getValue(FuzzedDataProvider *fdp, std::vector<T> vec) {
return vec[fdp->ConsumeIntegralInRange<int32_t>(0, vec.size() - 1)];
}
int32_t getSampleRate(FuzzedDataProvider *fdp) {
if (fdp->ConsumeBool()) {
return fdp->ConsumeIntegralInRange<int32_t>(kMinSampleRateHz, kMaxSampleRateHz);
}
return kSampleRateUnspecified;
}
class DeathNotifier : public IBinder::DeathRecipient {
public:
void binderDied(const wp<IBinder> &) { abort(); }
};
class AudioFlingerFuzzer {
public:
AudioFlingerFuzzer(const uint8_t *data, size_t size);
void process();
private:
FuzzedDataProvider mFdp;
void invokeAudioTrack();
void invokeAudioRecord();
status_t invokeAudioEffect();
void invokeAudioSystem();
status_t invokeAudioInputDevice();
status_t invokeAudioOutputDevice();
void invokeAudioPatch();
sp<DeathNotifier> mDeathNotifier;
};
AudioFlingerFuzzer::AudioFlingerFuzzer(const uint8_t *data, size_t size) : mFdp(data, size) {
sp<IServiceManager> sm = defaultServiceManager();
sp<IBinder> binder = sm->getService(String16("media.audio_flinger"));
if (binder == nullptr) {
return;
}
mDeathNotifier = new DeathNotifier();
binder->linkToDeath(mDeathNotifier);
}
void AudioFlingerFuzzer::invokeAudioTrack() {
uint32_t sampleRate = getSampleRate(&mFdp);
audio_format_t format = getValue(&mFdp, kFormats);
audio_channel_mask_t channelMask = getValue(&mFdp, kChannelMasks);
size_t frameCount = static_cast<size_t>(mFdp.ConsumeIntegral<uint32_t>());
int32_t notificationFrames = mFdp.ConsumeIntegral<int32_t>();
uint32_t useSharedBuffer = mFdp.ConsumeBool();
audio_output_flags_t flags = getValue(&mFdp, kOutputFlags);
audio_session_t sessionId = getValue(&mFdp, kSessionId);
audio_usage_t usage = getValue(&mFdp, kUsages);
audio_content_type_t contentType = getValue(&mFdp, kContentType);
audio_attributes_t attributes = {};
sp<IMemory> sharedBuffer;
sp<MemoryDealer> heap = nullptr;
audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER;
bool offload = false;
bool fast = ((flags & AUDIO_OUTPUT_FLAG_FAST) != 0);
if (useSharedBuffer != 0) {
size_t heapSize = audio_channel_count_from_out_mask(channelMask) *
audio_bytes_per_sample(format) * frameCount;
heap = new MemoryDealer(heapSize, "AudioTrack Heap Base");
sharedBuffer = heap->allocate(heapSize);
frameCount = 0;
notificationFrames = 0;
}
if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
offloadInfo.sample_rate = sampleRate;
offloadInfo.channel_mask = channelMask;
offloadInfo.format = format;
offload = true;
}
attributes.content_type = contentType;
attributes.usage = usage;
sp<AudioTrack> track = new AudioTrack();
// TODO b/182392769: use attribution source util
AttributionSourceState attributionSource;
attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
attributionSource.token = sp<BBinder>::make();
track->set(AUDIO_STREAM_DEFAULT, sampleRate, format, channelMask, frameCount, flags, nullptr,
notificationFrames, sharedBuffer, false, sessionId,
((fast && sharedBuffer == 0) || offload) ? AudioTrack::TRANSFER_CALLBACK
: AudioTrack::TRANSFER_DEFAULT,
offload ? &offloadInfo : nullptr, attributionSource, &attributes, false, 1.0f,
AUDIO_PORT_HANDLE_NONE);
status_t status = track->initCheck();
if (status != NO_ERROR) {
track.clear();
return;
}
track->getSampleRate();
track->latency();
track->getUnderrunCount();
track->streamType();
track->channelCount();
track->getNotificationPeriodInFrames();
uint32_t bufferSizeInFrames = mFdp.ConsumeIntegral<uint32_t>();
track->setBufferSizeInFrames(bufferSizeInFrames);
track->getBufferSizeInFrames();
int64_t duration = mFdp.ConsumeIntegral<int64_t>();
track->getBufferDurationInUs(&duration);
sp<IMemory> sharedBuffer2 = track->sharedBuffer();
track->setCallerName(mFdp.ConsumeRandomLengthString(MAX_STRING_LENGTH));
track->setVolume(mFdp.ConsumeFloatingPoint<float>(), mFdp.ConsumeFloatingPoint<float>());
track->setVolume(mFdp.ConsumeFloatingPoint<float>());
track->setAuxEffectSendLevel(mFdp.ConsumeFloatingPoint<float>());
float auxEffectSendLevel;
track->getAuxEffectSendLevel(&auxEffectSendLevel);
track->setSampleRate(getSampleRate(&mFdp));
track->getSampleRate();
track->getOriginalSampleRate();
AudioPlaybackRate playbackRate = {};
playbackRate.mSpeed = mFdp.ConsumeFloatingPoint<float>();
playbackRate.mPitch = mFdp.ConsumeFloatingPoint<float>();
track->setPlaybackRate(playbackRate);
track->getPlaybackRate();
track->setLoop(mFdp.ConsumeIntegral<uint32_t>(), mFdp.ConsumeIntegral<uint32_t>(),
mFdp.ConsumeIntegral<uint32_t>());
track->setMarkerPosition(mFdp.ConsumeIntegral<uint32_t>());
uint32_t marker = {};
track->getMarkerPosition(&marker);
track->setPositionUpdatePeriod(mFdp.ConsumeIntegral<uint32_t>());
uint32_t updatePeriod = {};
track->getPositionUpdatePeriod(&updatePeriod);
track->setPosition(mFdp.ConsumeIntegral<uint32_t>());
uint32_t position = {};
track->getPosition(&position);
track->getBufferPosition(&position);
track->reload();
track->start();
track->pause();
track->flush();
track->stop();
track->stopped();
}
void AudioFlingerFuzzer::invokeAudioRecord() {
int32_t notificationFrames = mFdp.ConsumeIntegral<int32_t>();
uint32_t sampleRate = getSampleRate(&mFdp);
size_t frameCount = static_cast<size_t>(mFdp.ConsumeIntegral<uint32_t>());
audio_format_t format = getValue(&mFdp, kFormats);
audio_channel_mask_t channelMask = getValue(&mFdp, kChannelMasks);
audio_input_flags_t flags = getValue(&mFdp, kInputFlags);
audio_session_t sessionId = getValue(&mFdp, kSessionId);
audio_source_t inputSource = getValue(&mFdp, kInputSources);
audio_attributes_t attributes = {};
bool fast = ((flags & AUDIO_OUTPUT_FLAG_FAST) != 0);
attributes.source = inputSource;
// TODO b/182392769: use attribution source util
AttributionSourceState attributionSource;
attributionSource.packageName = std::string(mFdp.ConsumeRandomLengthString().c_str());
attributionSource.token = sp<BBinder>::make();
sp<AudioRecord> record = new AudioRecord(attributionSource);
record->set(AUDIO_SOURCE_DEFAULT, sampleRate, format, channelMask, frameCount, nullptr,
notificationFrames, false, sessionId,
fast ? AudioRecord::TRANSFER_CALLBACK : AudioRecord::TRANSFER_DEFAULT, flags,
getuid(), getpid(), &attributes, AUDIO_PORT_HANDLE_NONE);
status_t status = record->initCheck();
if (status != NO_ERROR) {
return;
}
record->latency();
record->format();
record->channelCount();
record->frameCount();
record->frameSize();
record->inputSource();
record->getNotificationPeriodInFrames();
record->start();
record->stop();
record->stopped();
uint32_t marker = mFdp.ConsumeIntegral<uint32_t>();
record->setMarkerPosition(marker);
record->getMarkerPosition(&marker);
uint32_t updatePeriod = mFdp.ConsumeIntegral<uint32_t>();
record->setPositionUpdatePeriod(updatePeriod);
record->getPositionUpdatePeriod(&updatePeriod);
uint32_t position;
record->getPosition(&position);
ExtendedTimestamp timestamp;
record->getTimestamp(&timestamp);
record->getSessionId();
record->getCallerName();
android::AudioRecord::Buffer audioBuffer;
int32_t waitCount = mFdp.ConsumeIntegral<int32_t>();
size_t nonContig = static_cast<size_t>(mFdp.ConsumeIntegral<uint32_t>());
audioBuffer.frameCount = static_cast<size_t>(mFdp.ConsumeIntegral<uint32_t>());
record->obtainBuffer(&audioBuffer, waitCount, &nonContig);
bool blocking = false;
record->read(audioBuffer.data(), audioBuffer.size(), blocking);
record->getInputFramesLost();
record->getFlags();
std::vector<media::MicrophoneInfoFw> activeMicrophones;
record->getActiveMicrophones(&activeMicrophones);
record->releaseBuffer(&audioBuffer);
audio_port_handle_t deviceId =
static_cast<audio_port_handle_t>(mFdp.ConsumeIntegral<int32_t>());
record->setInputDevice(deviceId);
record->getInputDevice();
record->getRoutedDeviceId();
record->getPortId();
}
struct EffectClient : public android::media::BnEffectClient {
EffectClient() {}
binder::Status controlStatusChanged(bool controlGranted __unused) override {
return binder::Status::ok();
}
binder::Status enableStatusChanged(bool enabled __unused) override {
return binder::Status::ok();
}
binder::Status commandExecuted(int32_t cmdCode __unused,
const std::vector<uint8_t> &cmdData __unused,
const std::vector<uint8_t> &replyData __unused) override {
return binder::Status::ok();
}
binder::Status framesProcessed(int32_t frames __unused) override {
return binder::Status::ok();
}
};
status_t AudioFlingerFuzzer::invokeAudioEffect() {
effect_uuid_t type;
type.timeLow = mFdp.ConsumeIntegral<uint32_t>();
type.timeMid = mFdp.ConsumeIntegral<uint16_t>();
type.timeHiAndVersion = mFdp.ConsumeIntegral<uint16_t>();
type.clockSeq = mFdp.ConsumeIntegral<uint16_t>();
for (int i = 0; i < 6; ++i) {
type.node[i] = mFdp.ConsumeIntegral<uint8_t>();
}
effect_descriptor_t descriptor = {};
descriptor.type = type;
descriptor.uuid = *EFFECT_UUID_NULL;
sp<EffectClient> effectClient(new EffectClient());
const int32_t priority = mFdp.ConsumeIntegral<int32_t>();
audio_session_t sessionId = static_cast<audio_session_t>(mFdp.ConsumeIntegral<int32_t>());
const audio_io_handle_t io = mFdp.ConsumeIntegral<int32_t>();
std::string opPackageName = static_cast<std::string>(mFdp.ConsumeRandomLengthString().c_str());
AudioDeviceTypeAddr device;
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (!af) {
return NO_ERROR;
}
media::CreateEffectRequest request{};
request.desc =
VALUE_OR_RETURN_STATUS(legacy2aidl_effect_descriptor_t_EffectDescriptor(descriptor));
request.client = effectClient;
request.priority = priority;
request.output = io;
request.sessionId = sessionId;
request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioDeviceTypeAddress(device));
// TODO b/182392769: use attribution source util
request.attributionSource.packageName = opPackageName;
request.attributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(getpid()));
request.probe = false;
request.notifyFramesProcessed = false;
media::CreateEffectResponse response{};
status_t status = af->createEffect(request, &response);
if (status != OK) {
return NO_ERROR;
}
descriptor =
VALUE_OR_RETURN_STATUS(aidl2legacy_EffectDescriptor_effect_descriptor_t(response.desc));
uint32_t numEffects;
af->queryNumberEffects(&numEffects);
uint32_t queryIndex = mFdp.ConsumeIntegral<uint32_t>();
af->queryEffect(queryIndex, &descriptor);
effect_descriptor_t getDescriptor;
uint32_t preferredTypeFlag = mFdp.ConsumeIntegral<int32_t>();
af->getEffectDescriptor(&descriptor.uuid, &descriptor.type, preferredTypeFlag, &getDescriptor);
sessionId = static_cast<audio_session_t>(mFdp.ConsumeIntegral<int32_t>());
audio_io_handle_t srcOutput = mFdp.ConsumeIntegral<int32_t>();
audio_io_handle_t dstOutput = mFdp.ConsumeIntegral<int32_t>();
af->moveEffects(sessionId, srcOutput, dstOutput);
int effectId = mFdp.ConsumeIntegral<int32_t>();
sessionId = static_cast<audio_session_t>(mFdp.ConsumeIntegral<int32_t>());
af->setEffectSuspended(effectId, sessionId, mFdp.ConsumeBool());
return NO_ERROR;
}
void AudioFlingerFuzzer::invokeAudioSystem() {
AudioSystem::muteMicrophone(mFdp.ConsumeBool());
AudioSystem::setMasterMute(mFdp.ConsumeBool());
AudioSystem::setMasterVolume(mFdp.ConsumeFloatingPoint<float>());
AudioSystem::setMasterBalance(mFdp.ConsumeFloatingPoint<float>());
AudioSystem::setVoiceVolume(mFdp.ConsumeFloatingPoint<float>());
float volume;
AudioSystem::getMasterVolume(&volume);
bool state;
AudioSystem::getMasterMute(&state);
AudioSystem::isMicrophoneMuted(&state);
audio_stream_type_t stream = getValue(&mFdp, kStreamtypes);
AudioSystem::setStreamMute(getValue(&mFdp, kStreamtypes), mFdp.ConsumeBool());
stream = getValue(&mFdp, kStreamtypes);
AudioSystem::setStreamVolume(stream, mFdp.ConsumeFloatingPoint<float>(),
mFdp.ConsumeIntegral<int32_t>());
audio_mode_t mode = getValue(&mFdp, kModes);
AudioSystem::setMode(mode);
size_t frameCount;
stream = getValue(&mFdp, kStreamtypes);
AudioSystem::getOutputFrameCount(&frameCount, stream);
uint32_t latency;
stream = getValue(&mFdp, kStreamtypes);
AudioSystem::getOutputLatency(&latency, stream);
stream = getValue(&mFdp, kStreamtypes);
AudioSystem::getStreamVolume(stream, &volume, mFdp.ConsumeIntegral<int32_t>());
stream = getValue(&mFdp, kStreamtypes);
AudioSystem::getStreamMute(stream, &state);
uint32_t samplingRate;
AudioSystem::getSamplingRate(mFdp.ConsumeIntegral<int32_t>(), &samplingRate);
AudioSystem::getFrameCount(mFdp.ConsumeIntegral<int32_t>(), &frameCount);
AudioSystem::getLatency(mFdp.ConsumeIntegral<int32_t>(), &latency);
AudioSystem::setVoiceVolume(mFdp.ConsumeFloatingPoint<float>());
uint32_t halFrames;
uint32_t dspFrames;
AudioSystem::getRenderPosition(mFdp.ConsumeIntegral<int32_t>(), &halFrames, &dspFrames);
AudioSystem::getInputFramesLost(mFdp.ConsumeIntegral<int32_t>());
AudioSystem::getInputFramesLost(mFdp.ConsumeIntegral<int32_t>());
audio_unique_id_use_t uniqueIdUse = getValue(&mFdp, kUniqueIds);
AudioSystem::newAudioUniqueId(uniqueIdUse);
audio_session_t sessionId = getValue(&mFdp, kSessionId);
pid_t pid = mFdp.ConsumeBool() ? getpid() : mFdp.ConsumeIntegral<int32_t>();
uid_t uid = mFdp.ConsumeBool() ? getuid() : mFdp.ConsumeIntegral<int32_t>();
AudioSystem::acquireAudioSessionId(sessionId, pid, uid);
pid = mFdp.ConsumeBool() ? getpid() : mFdp.ConsumeIntegral<int32_t>();
sessionId = getValue(&mFdp, kSessionId);
AudioSystem::releaseAudioSessionId(sessionId, pid);
sessionId = getValue(&mFdp, kSessionId);
AudioSystem::getAudioHwSyncForSession(sessionId);
AudioSystem::systemReady();
AudioSystem::getFrameCountHAL(mFdp.ConsumeIntegral<int32_t>(), &frameCount);
size_t buffSize;
uint32_t sampleRate = getSampleRate(&mFdp);
audio_format_t format = getValue(&mFdp, kFormats);
audio_channel_mask_t channelMask = getValue(&mFdp, kChannelMasks);
AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &buffSize);
AudioSystem::getPrimaryOutputSamplingRate();
AudioSystem::getPrimaryOutputFrameCount();
AudioSystem::setLowRamDevice(mFdp.ConsumeBool(), mFdp.ConsumeIntegral<int64_t>());
std::vector<media::MicrophoneInfoFw> microphones;
AudioSystem::getMicrophones(&microphones);
std::vector<pid_t> pids;
pids.insert(pids.begin(), getpid());
for (int i = 1; i < mFdp.ConsumeIntegralInRange<int32_t>(2, MAX_ARRAY_LENGTH); ++i) {
pids.insert(pids.begin() + i, static_cast<pid_t>(mFdp.ConsumeIntegral<int32_t>()));
}
AudioSystem::setAudioHalPids(pids);
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (!af) {
return;
}
af->setRecordSilenced(mFdp.ConsumeIntegral<uint32_t>(), mFdp.ConsumeBool());
float balance = mFdp.ConsumeFloatingPoint<float>();
af->getMasterBalance(&balance);
std::vector<audio_port_handle_t> tracks;
for (int i = 0; i < mFdp.ConsumeIntegralInRange<int32_t>(0, MAX_ARRAY_LENGTH); ++i) {
tracks.push_back(static_cast<audio_port_handle_t>(mFdp.ConsumeIntegral<int32_t>()));
}
af->invalidateTracks(tracks);
}
status_t AudioFlingerFuzzer::invokeAudioInputDevice() {
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (!af) {
return NO_ERROR;
}
audio_config_t config = {};
audio_module_handle_t module = mFdp.ConsumeIntegral<int32_t>();
audio_io_handle_t input = mFdp.ConsumeIntegral<int32_t>();
config.frame_count = mFdp.ConsumeIntegral<uint32_t>();
String8 address = static_cast<String8>(mFdp.ConsumeRandomLengthString().c_str());
config.channel_mask = getValue(&mFdp, kChannelMasks);
config.format = getValue(&mFdp, kFormats);
config.offload_info = AUDIO_INFO_INITIALIZER;
config.offload_info.bit_rate = mFdp.ConsumeIntegral<uint32_t>();
config.offload_info.bit_width = mFdp.ConsumeIntegral<uint32_t>();
config.offload_info.content_id = mFdp.ConsumeIntegral<uint32_t>();
config.offload_info.channel_mask = getValue(&mFdp, kChannelMasks);
config.offload_info.duration_us = mFdp.ConsumeIntegral<int64_t>();
config.offload_info.encapsulation_mode = getValue(&mFdp, kEncapsulation);
config.offload_info.format = getValue(&mFdp, kFormats);
config.offload_info.has_video = mFdp.ConsumeBool();
config.offload_info.is_streaming = mFdp.ConsumeBool();
config.offload_info.sample_rate = getSampleRate(&mFdp);
config.offload_info.sync_id = mFdp.ConsumeIntegral<uint32_t>();
config.offload_info.stream_type = getValue(&mFdp, kStreamtypes);
config.offload_info.usage = getValue(&mFdp, kUsages);
config.sample_rate = getSampleRate(&mFdp);
audio_devices_t device = getValue(&mFdp, kDevices);
audio_source_t source = getValue(&mFdp, kInputSources);
audio_input_flags_t flags = getValue(&mFdp, kInputFlags);
AudioDeviceTypeAddr deviceTypeAddr(device, address.c_str());
media::OpenInputRequest request{};
request.module = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_module_handle_t_int32_t(module));
request.input = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(input));
request.config = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_config_t_AudioConfig(config, true /*isInput*/));
request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioDeviceTypeAddress(deviceTypeAddr));
request.source = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_source_t_AudioSource(source));
request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_input_flags_t_int32_t_mask(flags));
media::OpenInputResponse response{};
status_t status = af->openInput(request, &response);
if (status != NO_ERROR) {
return NO_ERROR;
}
input = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_module_handle_t(response.input));
af->closeInput(input);
return NO_ERROR;
}
status_t AudioFlingerFuzzer::invokeAudioOutputDevice() {
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (!af) {
return NO_ERROR;
}
audio_config_t config = {};
audio_module_handle_t module = mFdp.ConsumeIntegral<int32_t>();
audio_io_handle_t output = mFdp.ConsumeIntegral<int32_t>();
config.frame_count = mFdp.ConsumeIntegral<uint32_t>();
String8 address = static_cast<String8>(mFdp.ConsumeRandomLengthString().c_str());
config.channel_mask = getValue(&mFdp, kChannelMasks);
config.offload_info = AUDIO_INFO_INITIALIZER;
config.offload_info.bit_rate = mFdp.ConsumeIntegral<uint32_t>();
config.offload_info.bit_width = mFdp.ConsumeIntegral<uint32_t>();
config.offload_info.channel_mask = getValue(&mFdp, kChannelMasks);
config.offload_info.content_id = mFdp.ConsumeIntegral<uint32_t>();
config.offload_info.duration_us = mFdp.ConsumeIntegral<int64_t>();
config.offload_info.encapsulation_mode = getValue(&mFdp, kEncapsulation);
config.offload_info.format = getValue(&mFdp, kFormats);
config.offload_info.has_video = mFdp.ConsumeBool();
config.offload_info.is_streaming = mFdp.ConsumeBool();
config.offload_info.sample_rate = getSampleRate(&mFdp);
config.offload_info.stream_type = getValue(&mFdp, kStreamtypes);
config.offload_info.sync_id = mFdp.ConsumeIntegral<uint32_t>();
config.offload_info.usage = getValue(&mFdp, kUsages);
config.format = getValue(&mFdp, kFormats);
config.sample_rate = getSampleRate(&mFdp);
sp<DeviceDescriptorBase> device = new DeviceDescriptorBase(getValue(&mFdp, kDevices));
audio_output_flags_t flags = getValue(&mFdp, kOutputFlags);
audio_config_base_t mixerConfig = AUDIO_CONFIG_BASE_INITIALIZER;
media::OpenOutputRequest request{};
media::OpenOutputResponse response{};
request.module = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_module_handle_t_int32_t(module));
request.halConfig = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_config_t_AudioConfig(config, false /*isInput*/));
request.mixerConfig = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_config_base_t_AudioConfigBase(mixerConfig, false /*isInput*/));
request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_DeviceDescriptorBase(device));
request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
status_t status = af->openOutput(request, &response);
if (status != NO_ERROR) {
return NO_ERROR;
}
output = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_io_handle_t(response.output));
audio_io_handle_t output1 = mFdp.ConsumeIntegral<int32_t>();
af->openDuplicateOutput(output, output1);
af->suspendOutput(output);
af->restoreOutput(output);
af->closeOutput(output);
return NO_ERROR;
}
void AudioFlingerFuzzer::invokeAudioPatch() {
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (!af) {
return;
}
struct audio_patch patch = {};
audio_patch_handle_t handle = mFdp.ConsumeIntegral<int32_t>();
patch.id = mFdp.ConsumeIntegral<int32_t>();
patch.num_sources = mFdp.ConsumeIntegral<uint32_t>();
patch.num_sinks = mFdp.ConsumeIntegral<uint32_t>();
for (int i = 0; i < AUDIO_PATCH_PORTS_MAX; ++i) {
patch.sources[i].config_mask = mFdp.ConsumeIntegral<uint32_t>();
patch.sources[i].channel_mask = getValue(&mFdp, kChannelMasks);
patch.sources[i].format = getValue(&mFdp, kFormats);
patch.sources[i].gain.channel_mask = getValue(&mFdp, kChannelMasks);
patch.sources[i].gain.index = mFdp.ConsumeIntegral<int32_t>();
patch.sources[i].gain.mode = getValue(&mFdp, kGainModes);
patch.sources[i].gain.ramp_duration_ms = mFdp.ConsumeIntegral<uint32_t>();
patch.sources[i].id = static_cast<audio_format_t>(mFdp.ConsumeIntegral<int32_t>());
patch.sources[i].role = getValue(&mFdp, kPortRoles);
patch.sources[i].sample_rate = getSampleRate(&mFdp);
patch.sources[i].type = getValue(&mFdp, kPortTypes);
patch.sinks[i].config_mask = mFdp.ConsumeIntegral<uint32_t>();
patch.sinks[i].channel_mask = getValue(&mFdp, kChannelMasks);
patch.sinks[i].format = getValue(&mFdp, kFormats);
patch.sinks[i].gain.channel_mask = getValue(&mFdp, kChannelMasks);
patch.sinks[i].gain.index = mFdp.ConsumeIntegral<int32_t>();
patch.sinks[i].gain.mode = getValue(&mFdp, kGainModes);
patch.sinks[i].gain.ramp_duration_ms = mFdp.ConsumeIntegral<uint32_t>();
patch.sinks[i].id = static_cast<audio_format_t>(mFdp.ConsumeIntegral<int32_t>());
patch.sinks[i].role = getValue(&mFdp, kPortRoles);
patch.sinks[i].sample_rate = getSampleRate(&mFdp);
patch.sinks[i].type = getValue(&mFdp, kPortTypes);
}
status_t status = af->createAudioPatch(&patch, &handle);
if (status != NO_ERROR) {
return;
}
unsigned int num_patches = mFdp.ConsumeIntegral<uint32_t>();
struct audio_patch patches = {};
af->listAudioPatches(&num_patches, &patches);
af->releaseAudioPatch(handle);
}
void AudioFlingerFuzzer::process() {
invokeAudioEffect();
invokeAudioInputDevice();
invokeAudioOutputDevice();
invokeAudioPatch();
invokeAudioRecord();
invokeAudioSystem();
invokeAudioTrack();
}
extern "C" int LLVMFuzzerTestOneInput(const uint8_t *data, size_t size) {
if (size < 1) {
return 0;
}
AudioFlingerFuzzer audioFuzzer(data, size);
audioFuzzer.process();
return 0;
}