diff --git a/media/libaudioclient/tests/Android.bp b/media/libaudioclient/tests/Android.bp
index 891293e..48a5078 100644
--- a/media/libaudioclient/tests/Android.bp
+++ b/media/libaudioclient/tests/Android.bp
@@ -93,3 +93,61 @@
     ],
     data: ["record_test_input_*.txt"],
 }
+
+cc_defaults {
+    name: "libaudioclient_gtests_defaults",
+    cflags: [
+        "-Wall",
+        "-Werror",
+    ],
+    shared_libs: [
+        "capture_state_listener-aidl-cpp",
+        "framework-permission-aidl-cpp",
+        "libbase",
+        "libbinder",
+        "libcgrouprc",
+        "libcutils",
+        "libdl",
+        "liblog",
+        "libmedia",
+        "libmediametrics",
+        "libmediautils",
+        "libmedia_helper",
+        "libnblog",
+        "libprocessgroup",
+        "libshmemcompat",
+        "libstagefright_foundation",
+        "libutils",
+        "libvibrator",
+        "mediametricsservice-aidl-cpp",
+        "packagemanager_aidl-cpp",
+        "shared-file-region-aidl-cpp",
+    ],
+    static_libs: [
+        "android.hardware.audio.common@7.0-enums",
+        "android.media.audio.common.types-V1-cpp",
+        "audioclient-types-aidl-cpp",
+        "audioflinger-aidl-cpp",
+        "audiopolicy-aidl-cpp",
+        "audiopolicy-types-aidl-cpp",
+        "av-types-aidl-cpp",
+        "effect-aidl-cpp",
+        "libaudioclient",
+        "libaudioclient_aidl_conversion",
+        "libaudiofoundation",
+        "libaudiomanager",
+        "libaudiopolicy",
+        "libaudioutils",
+    ],
+    data: ["bbb*.raw"],
+    test_config_template: "audio_test_template.xml",
+}
+
+cc_test {
+    name: "audiorecord_tests",
+    defaults: ["libaudioclient_gtests_defaults"],
+    srcs: [
+        "audiorecord_tests.cpp",
+        "audio_test_utils.cpp",
+    ],
+}
diff --git a/media/libaudioclient/tests/audio_test_template.xml b/media/libaudioclient/tests/audio_test_template.xml
new file mode 100644
index 0000000..ed0cb21
--- /dev/null
+++ b/media/libaudioclient/tests/audio_test_template.xml
@@ -0,0 +1,32 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2022 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.
+-->
+<configuration description="Unit test configuration for {MODULE}">
+    <target_preparer class="com.android.tradefed.targetprep.RootTargetPreparer" />
+
+    <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
+        <option name="cleanup" value="true" />
+        <option name="push-file" key="{MODULE}" value="/data/local/tmp/{MODULE}" />
+
+        <!-- Files used for audio testing -->
+        <option name="push-file" key="bbb_1ch_8kHz_s16le.raw" value="/data/local/tmp/bbb_1ch_8kHz_s16le.raw" />
+        <option name="push-file" key="bbb_2ch_24kHz_s16le.raw" value="/data/local/tmp/bbb_2ch_24kHz_s16le.raw" />
+    </target_preparer>
+
+    <test class="com.android.tradefed.testtype.GTest" >
+        <option name="native-test-device-path" value="/data/local/tmp" />
+        <option name="module-name" value="{MODULE}" />
+    </test>
+</configuration>
diff --git a/media/libaudioclient/tests/audio_test_utils.cpp b/media/libaudioclient/tests/audio_test_utils.cpp
new file mode 100644
index 0000000..df36b62
--- /dev/null
+++ b/media/libaudioclient/tests/audio_test_utils.cpp
@@ -0,0 +1,525 @@
+/*
+ * 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.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "AudioTestUtils"
+
+#include <utils/Log.h>
+
+#include "audio_test_utils.h"
+
+// Generates a random string.
+void CreateRandomFile(int& fd) {
+    std::string filename = "/data/local/tmp/record-XXXXXX";
+    fd = mkstemp(filename.data());
+}
+
+void OnAudioDeviceUpdateNotifier::onAudioDeviceUpdate(audio_io_handle_t audioIo,
+                                                      audio_port_handle_t deviceId) {
+    std::unique_lock<std::mutex> lock{mMutex};
+    ALOGD("%s  audioIo=%d deviceId=%d", __func__, audioIo, deviceId);
+    mAudioIo = audioIo;
+    mDeviceId = deviceId;
+    mCondition.notify_all();
+}
+
+status_t OnAudioDeviceUpdateNotifier::waitForAudioDeviceCb() {
+    std::unique_lock<std::mutex> lock{mMutex};
+    if (mAudioIo == AUDIO_IO_HANDLE_NONE) {
+        mCondition.wait_for(lock, std::chrono::milliseconds(500));
+        if (mAudioIo == AUDIO_IO_HANDLE_NONE) return TIMED_OUT;
+    }
+    return OK;
+}
+
+// AudioTrack callback function.
+static void AudioTrackCallBackFunction(int event, void* user, void* info __unused) {
+    switch (event) {
+        case AudioTrack::EVENT_BUFFER_END: {
+            AudioPlayback* ap = (AudioPlayback*)user;
+            std::unique_lock<std::mutex> lock{ap->mMutex};
+            ap->mStopPlaying = true;
+            ap->mCondition.notify_all();
+            break;
+        }
+        default:
+            ALOGV("received audiotrack callback %d", event);
+            break;
+    }
+}
+
+AudioPlayback::AudioPlayback(uint32_t sampleRate, audio_format_t format,
+                             audio_channel_mask_t channelMask, audio_output_flags_t flags,
+                             audio_session_t sessionId, AudioTrack::transfer_type transferType,
+                             audio_attributes_t* attributes)
+    : mSampleRate(sampleRate),
+      mFormat(format),
+      mChannelMask(channelMask),
+      mFlags(flags),
+      mSessionId(sessionId),
+      mTransferType(transferType),
+      mAttributes(attributes) {
+    mStopPlaying = false;
+    mBytesUsedSoFar = 0;
+    mState = PLAY_NO_INIT;
+    mMemCapacity = 0;
+    mMemoryDealer = nullptr;
+    mMemory = nullptr;
+}
+
+AudioPlayback::~AudioPlayback() {
+    stop();
+}
+
+status_t AudioPlayback::create() {
+    if (mState != PLAY_NO_INIT) return INVALID_OPERATION;
+    std::string packageName{"AudioPlayback"};
+    AttributionSourceState attributionSource;
+    attributionSource.packageName = packageName;
+    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();
+    if (mTransferType == AudioTrack::TRANSFER_OBTAIN) {
+        mTrack = new AudioTrack(attributionSource);
+        mTrack->set(AUDIO_STREAM_MUSIC, mSampleRate, mFormat, mChannelMask, 0, mFlags, nullptr,
+                    nullptr, 0, 0, false, mSessionId, mTransferType, nullptr, attributionSource,
+                    mAttributes);
+    } else if (mTransferType == AudioTrack::TRANSFER_SHARED) {
+        mTrack = new AudioTrack(AUDIO_STREAM_MUSIC, mSampleRate, mFormat, mChannelMask, mMemory,
+                                mFlags, AudioTrackCallBackFunction, this, 0, mSessionId,
+                                mTransferType, nullptr, attributionSource, mAttributes);
+    } else {
+        ALOGE("Required Transfer type not existed");
+        return INVALID_OPERATION;
+    }
+    mTrack->setCallerName(packageName);
+    status_t status = mTrack->initCheck();
+    if (NO_ERROR == status) mState = PLAY_READY;
+    return status;
+}
+
+status_t AudioPlayback::loadResource(const char* name) {
+    status_t status = OK;
+    FILE* fp = fopen(name, "rbe");
+    struct stat buf {};
+    if (fp && !fstat(fileno(fp), &buf)) {
+        mMemCapacity = buf.st_size;
+        mMemoryDealer = new MemoryDealer(mMemCapacity, "AudioPlayback");
+        if (nullptr == mMemoryDealer.get()) {
+            ALOGE("couldn't get MemoryDealer!");
+            fclose(fp);
+            return NO_MEMORY;
+        }
+        mMemory = mMemoryDealer->allocate(mMemCapacity);
+        if (nullptr == mMemory.get()) {
+            ALOGE("couldn't get IMemory!");
+            fclose(fp);
+            return NO_MEMORY;
+        }
+        uint8_t* ipBuffer = static_cast<uint8_t*>(static_cast<void*>(mMemory->unsecurePointer()));
+        fread(ipBuffer, sizeof(uint8_t), mMemCapacity, fp);
+    } else {
+        ALOGE("unable to open input file %s", name);
+        status = NAME_NOT_FOUND;
+    }
+    if (fp) fclose(fp);
+    return status;
+}
+
+sp<AudioTrack> AudioPlayback::getAudioTrackHandle() {
+    return (PLAY_NO_INIT != mState) ? mTrack : nullptr;
+}
+
+status_t AudioPlayback::start() {
+    status_t status;
+    if (PLAY_READY != mState) {
+        return INVALID_OPERATION;
+    } else {
+        status = mTrack->start();
+        if (OK == status) {
+            mState = PLAY_STARTED;
+            LOG_FATAL_IF(false != mTrack->stopped());
+        }
+    }
+    return status;
+}
+
+status_t AudioPlayback::fillBuffer() {
+    if (PLAY_STARTED != mState && PLAY_STOPPED != mState) return INVALID_OPERATION;
+    int retry = 25;
+    uint8_t* ipBuffer = static_cast<uint8_t*>(static_cast<void*>(mMemory->unsecurePointer()));
+    size_t nonContig = 0;
+    size_t bytesAvailable = mMemCapacity - mBytesUsedSoFar;
+    while (bytesAvailable > 0) {
+        AudioTrack::Buffer trackBuffer;
+        trackBuffer.frameCount = mTrack->frameCount() * 2;
+        status_t status = mTrack->obtainBuffer(&trackBuffer, retry, &nonContig);
+        if (OK == status) {
+            size_t bytesToCopy = std::min(bytesAvailable, trackBuffer.size());
+            if (bytesToCopy > 0) {
+                memcpy(trackBuffer.data(), ipBuffer + mBytesUsedSoFar, bytesToCopy);
+            }
+            mTrack->releaseBuffer(&trackBuffer);
+            mBytesUsedSoFar += bytesToCopy;
+            bytesAvailable = mMemCapacity - mBytesUsedSoFar;
+            if (bytesAvailable == 0) {
+                stop();
+            }
+        } else if (WOULD_BLOCK == status) {
+            if (mStopPlaying)
+                return OK;
+            else
+                return TIMED_OUT;
+        }
+    }
+    return OK;
+}
+
+status_t AudioPlayback::waitForConsumption() {
+    if (PLAY_STARTED != mState) return INVALID_OPERATION;
+    // in static buffer mode, lets not play clips with duration > 30 sec
+    int retry = 30;
+    while (!mStopPlaying && retry > 0) {
+        std::this_thread::sleep_for(std::chrono::milliseconds(300));
+        retry--;
+    }
+    if (!mStopPlaying) return TIMED_OUT;
+    return OK;
+}
+
+status_t AudioPlayback::onProcess() {
+    if (mTransferType == AudioTrack::TRANSFER_SHARED)
+        return waitForConsumption();
+    else if (mTransferType == AudioTrack::TRANSFER_OBTAIN)
+        return fillBuffer();
+    else
+        return INVALID_OPERATION;
+}
+
+void AudioPlayback::stop() {
+    std::unique_lock<std::mutex> lock{mMutex};
+    mStopPlaying = true;
+    if (mState != PLAY_STOPPED) {
+        mTrack->stopAndJoinCallbacks();
+        LOG_FATAL_IF(true != mTrack->stopped());
+        mState = PLAY_STOPPED;
+    }
+}
+
+// hold pcm data sent by AudioRecord
+RawBuffer::RawBuffer(int64_t ptsPipeline, int64_t ptsManual, int32_t capacity)
+    : mData(capacity > 0 ? new uint8_t[capacity] : nullptr),
+      mPtsPipeline(ptsPipeline),
+      mPtsManual(ptsManual),
+      mCapacity(capacity) {}
+
+// Simple AudioCapture
+size_t AudioCapture::onMoreData(const AudioRecord::Buffer& buffer) {
+    if (mState != REC_STARTED) {
+        ALOGE("Unexpected Callback from audiorecord, not reading data");
+        return 0;
+    }
+
+    // no more frames to read
+    if (mNumFramesReceived > mNumFramesToRecord || mStopRecording) {
+        mStopRecording = true;
+        return 0;
+    }
+
+    int64_t timeUs = 0, position = 0, timeNs = 0;
+    ExtendedTimestamp ts;
+    ExtendedTimestamp::Location location;
+    const int32_t usPerSec = 1000000;
+
+    if (mRecord->getTimestamp(&ts) == OK &&
+        ts.getBestTimestamp(&position, &timeNs, ExtendedTimestamp::TIMEBASE_MONOTONIC, &location) ==
+                OK) {
+        // Use audio timestamp.
+        timeUs = timeNs / 1000 -
+                 (position - mNumFramesReceived + mNumFramesLost) * usPerSec / mSampleRate;
+    } else {
+        // This should not happen in normal case.
+        ALOGW("Failed to get audio timestamp, fallback to use systemclock");
+        timeUs = systemTime() / 1000LL;
+        // Estimate the real sampling time of the 1st sample in this buffer
+        // from AudioRecord's latency. (Apply this adjustment first so that
+        // the start time logic is not affected.)
+        timeUs -= mRecord->latency() * 1000LL;
+    }
+
+    ALOGV("dataCallbackTimestamp: %" PRId64 " us", timeUs);
+
+    const size_t frameSize = mRecord->frameSize();
+    uint64_t numLostBytes = (uint64_t)mRecord->getInputFramesLost() * frameSize;
+    if (numLostBytes > 0) {
+        ALOGW("Lost audio record data: %" PRIu64 " bytes", numLostBytes);
+    }
+    std::deque<RawBuffer> tmpQueue;
+    while (numLostBytes > 0) {
+        uint64_t bufferSize = numLostBytes;
+        if (numLostBytes > mMaxBytesPerCallback) {
+            numLostBytes -= mMaxBytesPerCallback;
+            bufferSize = mMaxBytesPerCallback;
+        } else {
+            numLostBytes = 0;
+        }
+        const int64_t timestampUs =
+                ((1000000LL * mNumFramesReceived) + (mRecord->getSampleRate() >> 1)) /
+                mRecord->getSampleRate();
+        RawBuffer emptyBuffer{timeUs, timestampUs, static_cast<int32_t>(bufferSize)};
+        memset(emptyBuffer.mData.get(), 0, bufferSize);
+        mNumFramesLost += bufferSize / frameSize;
+        mNumFramesReceived += bufferSize / frameSize;
+        tmpQueue.push_back(std::move(emptyBuffer));
+    }
+
+    if (buffer.size() == 0) {
+        ALOGW("Nothing is available from AudioRecord callback buffer");
+    } else {
+        const size_t bufferSize = buffer.size();
+        const int64_t timestampUs =
+                ((1000000LL * mNumFramesReceived) + (mRecord->getSampleRate() >> 1)) /
+                mRecord->getSampleRate();
+        RawBuffer audioBuffer{timeUs, timestampUs, static_cast<int32_t>(bufferSize)};
+        memcpy(audioBuffer.mData.get(), buffer.data(), bufferSize);
+        mNumFramesReceived += bufferSize / frameSize;
+        tmpQueue.push_back(std::move(audioBuffer));
+    }
+
+    if (tmpQueue.size() > 0) {
+        std::unique_lock<std::mutex> lock{mMutex};
+        for (auto it = tmpQueue.begin(); it != tmpQueue.end(); it++)
+            mBuffersReceived.push_back(std::move(*it));
+        mCondition.notify_all();
+    }
+    return buffer.size();
+}
+
+void AudioCapture::onOverrun() {
+    ALOGV("received event overrun");
+    mBufferOverrun = true;
+}
+
+void AudioCapture::onMarker(uint32_t markerPosition) {
+    ALOGV("received Callback at position %d", markerPosition);
+    mReceivedCbMarkerAtPosition = markerPosition;
+}
+
+void AudioCapture::onNewPos(uint32_t markerPosition) {
+    ALOGV("received Callback at position %d", markerPosition);
+    mReceivedCbMarkerCount++;
+}
+
+void AudioCapture::onNewIAudioRecord() {
+    ALOGV("IAudioRecord is re-created");
+}
+
+AudioCapture::AudioCapture(audio_source_t inputSource, uint32_t sampleRate, audio_format_t format,
+                           audio_channel_mask_t channelMask, audio_input_flags_t flags,
+                           audio_session_t sessionId, AudioRecord::transfer_type transferType)
+    : mInputSource(inputSource),
+      mSampleRate(sampleRate),
+      mFormat(format),
+      mChannelMask(channelMask),
+      mFlags(flags),
+      mSessionId(sessionId),
+      mTransferType(transferType) {
+    mFrameCount = 0;
+    mNotificationFrames = 0;
+    mNumFramesToRecord = 0;
+    mNumFramesReceived = 0;
+    mNumFramesLost = 0;
+    mBufferOverrun = false;
+    mMarkerPosition = 0;
+    mMarkerPeriod = 0;
+    mReceivedCbMarkerAtPosition = -1;
+    mReceivedCbMarkerCount = 0;
+    mState = REC_NO_INIT;
+    mStopRecording = false;
+#if RECORD_TO_FILE
+    CreateRandomFile(mOutFileFd);
+#endif
+}
+
+AudioCapture::~AudioCapture() {
+    if (mOutFileFd > 0) close(mOutFileFd);
+    stop();
+}
+
+status_t AudioCapture::create() {
+    if (mState != REC_NO_INIT) return INVALID_OPERATION;
+    // get Min Frame Count
+    size_t minFrameCount;
+    status_t status =
+            AudioRecord::getMinFrameCount(&minFrameCount, mSampleRate, mFormat, mChannelMask);
+    if (NO_ERROR != status) return status;
+    // Limit notificationFrames basing on client bufferSize
+    const int samplesPerFrame = audio_channel_count_from_in_mask(mChannelMask);
+    const int bytesPerSample = audio_bytes_per_sample(mFormat);
+    mNotificationFrames = mMaxBytesPerCallback / (samplesPerFrame * bytesPerSample);
+    // select frameCount to be at least minFrameCount
+    mFrameCount = 2 * mNotificationFrames;
+    while (mFrameCount < minFrameCount) {
+        mFrameCount += mNotificationFrames;
+    }
+    if (mFlags & AUDIO_INPUT_FLAG_FAST) {
+        ALOGW("Overriding all previous computations");
+        const uint32_t kMinNormalCaptureBufferSizeMs = 12;
+        size_t maxFrameCount = kMinNormalCaptureBufferSizeMs * mSampleRate / 1000;
+        mMaxBytesPerCallback = maxFrameCount * samplesPerFrame * bytesPerSample / 2;
+        mNotificationFrames = maxFrameCount / 2;
+        mFrameCount = 2 * mNotificationFrames;
+    }
+    mNumFramesToRecord = (mSampleRate * 0.25);  // record .25 sec
+    std::string packageName{"AudioCapture"};
+    AttributionSourceState attributionSource;
+    attributionSource.packageName = packageName;
+    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();
+    if (mTransferType == AudioRecord::TRANSFER_OBTAIN) {
+        mRecord = new AudioRecord(attributionSource);
+        status = mRecord->set(mInputSource, mSampleRate, mFormat, mChannelMask, mFrameCount,
+                              nullptr, nullptr, 0, false, mSessionId, mTransferType, mFlags,
+                              attributionSource.uid, attributionSource.pid);
+        if (NO_ERROR != status) return status;
+    } else if (mTransferType == AudioRecord::TRANSFER_CALLBACK) {
+        mRecord = new AudioRecord(mInputSource, mSampleRate, mFormat, mChannelMask,
+                                  attributionSource, mFrameCount, this, mNotificationFrames,
+                                  mSessionId, mTransferType, mFlags);
+    } else {
+        ALOGE("Test application is not handling transfer type %s",
+              AudioRecord::convertTransferToText(mTransferType));
+        return NO_INIT;
+    }
+    mRecord->setCallerName(packageName);
+    status = mRecord->initCheck();
+    if (NO_ERROR == status) mState = REC_READY;
+    return status;
+}
+
+sp<AudioRecord> AudioCapture::getAudioRecordHandle() {
+    return (REC_NO_INIT == mState) ? nullptr : mRecord;
+}
+
+status_t AudioCapture::start(AudioSystem::sync_event_t event, audio_session_t triggerSession) {
+    status_t status;
+    if (REC_READY != mState) {
+        return INVALID_OPERATION;
+    } else {
+        status = mRecord->start(event, triggerSession);
+        if (OK == status) {
+            mState = REC_STARTED;
+            LOG_FATAL_IF(false != mRecord->stopped());
+        }
+    }
+    return status;
+}
+
+status_t AudioCapture::stop() {
+    status_t status = OK;
+    mStopRecording = true;
+    if (mState != REC_STOPPED) {
+        uint32_t position;
+        status = mRecord->getPosition(&position);
+        if (OK == status && mTransferType == AudioRecord::TRANSFER_CALLBACK) {
+            if (position - mNumFramesToRecord > mFrameCount)
+                if (mBufferOverrun == false) status = BAD_VALUE;
+        }
+        mRecord->stopAndJoinCallbacks();
+        mState = REC_STOPPED;
+        LOG_FATAL_IF(true != mRecord->stopped());
+    }
+    return status;
+}
+
+status_t AudioCapture::obtainBuffer(RawBuffer& buffer) {
+    if (REC_STARTED != mState && REC_STOPPED != mState) return INVALID_OPERATION;
+    int retry = 25;
+    AudioRecord::Buffer recordBuffer;
+    recordBuffer.frameCount = mNotificationFrames;
+    size_t nonContig = 0;
+    status_t status = mRecord->obtainBuffer(&recordBuffer, retry, &nonContig);
+    if (OK == status) {
+        const int64_t timestampUs =
+                ((1000000LL * mNumFramesReceived) + (mRecord->getSampleRate() >> 1)) /
+                mRecord->getSampleRate();
+        RawBuffer buff{-1, timestampUs, static_cast<int32_t>(recordBuffer.size())};
+        memcpy(buff.mData.get(), recordBuffer.data(), recordBuffer.size());
+        buffer = std::move(buff);
+        mNumFramesReceived += recordBuffer.size() / mRecord->frameSize();
+        mRecord->releaseBuffer(&recordBuffer);
+        if (mNumFramesReceived > mNumFramesToRecord) {
+            stop();
+        }
+    } else if (status == WOULD_BLOCK) {
+        if (mStopRecording)
+            return WOULD_BLOCK;
+        else
+            return TIMED_OUT;
+    }
+    return OK;
+}
+
+status_t AudioCapture::obtainBufferCb(RawBuffer& buffer) {
+    if (REC_STARTED != mState) return INVALID_OPERATION;
+    int retry = 10;
+    std::unique_lock<std::mutex> lock{mMutex};
+    while (mBuffersReceived.empty() && !mStopRecording && retry > 0) {
+        mCondition.wait_for(lock, std::chrono::milliseconds(100));
+        retry--;
+    }
+    if (!mBuffersReceived.empty()) {
+        auto it = mBuffersReceived.begin();
+        buffer = std::move(*it);
+        mBuffersReceived.erase(it);
+    } else {
+        if (retry == 0) return TIMED_OUT;
+        if (mStopRecording)
+            return WOULD_BLOCK;
+        else
+            return UNKNOWN_ERROR;
+    }
+    return OK;
+}
+
+status_t AudioCapture::audioProcess() {
+    RawBuffer buffer;
+    while (true) {
+        status_t status;
+        if (mTransferType == AudioRecord::TRANSFER_CALLBACK)
+            status = obtainBufferCb(buffer);
+        else
+            status = obtainBuffer(buffer);
+        switch (status) {
+            case OK:
+                if (mOutFileFd > 0) {
+                    const char* ptr =
+                            static_cast<const char*>(static_cast<void*>(buffer.mData.get()));
+                    write(mOutFileFd, ptr, buffer.mCapacity);
+                }
+                break;
+            case WOULD_BLOCK:
+                return OK;
+            case TIMED_OUT:          // "recorder application timed out from receiving buffers"
+            case NO_INIT:            // "recorder not initialized"
+            case INVALID_OPERATION:  // "recorder not started"
+            case UNKNOWN_ERROR:      // "Unknown error"
+            default:
+                return status;
+        }
+    }
+}
diff --git a/media/libaudioclient/tests/audio_test_utils.h b/media/libaudioclient/tests/audio_test_utils.h
new file mode 100644
index 0000000..2e0d14d
--- /dev/null
+++ b/media/libaudioclient/tests/audio_test_utils.h
@@ -0,0 +1,171 @@
+/*
+ * 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.
+ */
+
+#ifndef AUDIO_TEST_UTILS_H_
+#define AUDIO_TEST_UTILS_H_
+
+#include <sys/stat.h>
+#include <unistd.h>
+#include <atomic>
+#include <chrono>
+#include <cinttypes>
+#include <deque>
+#include <memory>
+#include <mutex>
+#include <thread>
+
+#include <binder/MemoryDealer.h>
+#include <media/AidlConversion.h>
+#include <media/AudioRecord.h>
+#include <media/AudioTrack.h>
+
+#define RECORD_TO_FILE 0
+
+using namespace android;
+
+void CreateRandomFile(int& fd);
+
+class OnAudioDeviceUpdateNotifier : public AudioSystem::AudioDeviceCallback {
+  public:
+    audio_io_handle_t mAudioIo = AUDIO_IO_HANDLE_NONE;
+    audio_port_handle_t mDeviceId = AUDIO_PORT_HANDLE_NONE;
+    std::mutex mMutex;
+    std::condition_variable mCondition;
+
+    void onAudioDeviceUpdate(audio_io_handle_t audioIo, audio_port_handle_t deviceId);
+    status_t waitForAudioDeviceCb();
+};
+
+// Simple AudioPlayback class.
+class AudioPlayback {
+  public:
+    AudioPlayback(uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask,
+                  audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+                  audio_session_t sessionId = AUDIO_SESSION_NONE,
+                  AudioTrack::transfer_type transferType = AudioTrack::TRANSFER_SHARED,
+                  audio_attributes_t* attributes = nullptr);
+    ~AudioPlayback();
+    status_t loadResource(const char* name);
+    status_t create();
+    sp<AudioTrack> getAudioTrackHandle();
+    status_t start();
+    status_t waitForConsumption();
+    status_t fillBuffer();
+    status_t onProcess();
+    void stop();
+
+    bool mStopPlaying;
+    std::mutex mMutex;
+    std::condition_variable mCondition;
+
+    enum State {
+        PLAY_NO_INIT,
+        PLAY_READY,
+        PLAY_STARTED,
+        PLAY_STOPPED,
+    };
+
+  private:
+    const uint32_t mSampleRate;
+    const audio_format_t mFormat;
+    const audio_channel_mask_t mChannelMask;
+    const audio_output_flags_t mFlags;
+    const audio_session_t mSessionId;
+    const AudioTrack::transfer_type mTransferType;
+    const audio_attributes_t* mAttributes;
+
+    size_t mBytesUsedSoFar;
+    State mState;
+    size_t mMemCapacity;
+    sp<MemoryDealer> mMemoryDealer;
+    sp<IMemory> mMemory;
+
+    sp<AudioTrack> mTrack;
+};
+
+// hold pcm data sent by AudioRecord
+class RawBuffer {
+  public:
+    RawBuffer(int64_t ptsPipeline = -1, int64_t ptsManual = -1, int32_t capacity = 0);
+
+    std::unique_ptr<uint8_t[]> mData;
+    int64_t mPtsPipeline;
+    int64_t mPtsManual;
+    int32_t mCapacity;
+};
+
+// Simple AudioCapture
+class AudioCapture : public AudioRecord::IAudioRecordCallback {
+  public:
+    AudioCapture(audio_source_t inputSource, uint32_t sampleRate, audio_format_t format,
+                 audio_channel_mask_t channelMask,
+                 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
+                 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
+                 AudioRecord::transfer_type transferType = AudioRecord::TRANSFER_CALLBACK);
+    ~AudioCapture();
+    size_t onMoreData(const AudioRecord::Buffer& buffer) override;
+    void onOverrun() override;
+    void onMarker(uint32_t markerPosition) override;
+    void onNewPos(uint32_t newPos) override;
+    void onNewIAudioRecord() override;
+    status_t create();
+    sp<AudioRecord> getAudioRecordHandle();
+    status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
+                   audio_session_t triggerSession = AUDIO_SESSION_NONE);
+    status_t obtainBufferCb(RawBuffer& buffer);
+    status_t obtainBuffer(RawBuffer& buffer);
+    status_t audioProcess();
+    status_t stop();
+
+    uint32_t mFrameCount;
+    uint32_t mNotificationFrames;
+    int64_t mNumFramesToRecord;
+    int64_t mNumFramesReceived;
+    int64_t mNumFramesLost;
+    uint32_t mMarkerPosition;
+    uint32_t mMarkerPeriod;
+    uint32_t mReceivedCbMarkerAtPosition;
+    uint32_t mReceivedCbMarkerCount;
+    bool mBufferOverrun;
+
+    enum State {
+        REC_NO_INIT,
+        REC_READY,
+        REC_STARTED,
+        REC_STOPPED,
+    };
+
+  private:
+    const audio_source_t mInputSource;
+    const uint32_t mSampleRate;
+    const audio_format_t mFormat;
+    const audio_channel_mask_t mChannelMask;
+    const audio_input_flags_t mFlags;
+    const audio_session_t mSessionId;
+    const AudioRecord::transfer_type mTransferType;
+
+    size_t mMaxBytesPerCallback = 2048;
+    sp<AudioRecord> mRecord;
+    State mState;
+    bool mStopRecording;
+    int mOutFileFd = -1;
+
+    std::mutex mMutex;
+    std::condition_variable mCondition;
+    std::deque<RawBuffer> mBuffersReceived;
+};
+
+#endif  // AUDIO_TEST_UTILS_H_
diff --git a/media/libaudioclient/tests/audiorecord_tests.cpp b/media/libaudioclient/tests/audiorecord_tests.cpp
new file mode 100644
index 0000000..0002384
--- /dev/null
+++ b/media/libaudioclient/tests/audiorecord_tests.cpp
@@ -0,0 +1,237 @@
+/*
+ * 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.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "AudioRecordTest"
+
+#include <gtest/gtest.h>
+
+#include "audio_test_utils.h"
+
+using namespace android;
+
+class AudioRecordTest : public ::testing::Test {
+  public:
+    virtual void SetUp() override {
+        mAC = new AudioCapture(AUDIO_SOURCE_DEFAULT, 44100, AUDIO_FORMAT_PCM_16_BIT,
+                               AUDIO_CHANNEL_IN_FRONT);
+        ASSERT_NE(nullptr, mAC);
+        ASSERT_EQ(OK, mAC->create()) << "record creation failed";
+    }
+
+    virtual void TearDown() override {
+        if (mAC) ASSERT_EQ(OK, mAC->stop());
+    }
+
+    sp<AudioCapture> mAC;
+};
+
+class AudioRecordCreateTest
+    : public ::testing::TestWithParam<
+              std::tuple<uint32_t, audio_format_t, audio_channel_mask_t, audio_input_flags_t,
+                         audio_session_t, audio_source_t>> {
+  public:
+    AudioRecordCreateTest()
+        : mSampleRate(std::get<0>(GetParam())),
+          mFormat(std::get<1>(GetParam())),
+          mChannelMask(std::get<2>(GetParam())),
+          mFlags(std::get<3>(GetParam())),
+          mSessionId(std::get<4>(GetParam())),
+          mInputSource(std::get<5>(GetParam())){};
+
+    const uint32_t mSampleRate;
+    const audio_format_t mFormat;
+    const audio_channel_mask_t mChannelMask;
+    const audio_input_flags_t mFlags;
+    const audio_session_t mSessionId;
+    const audio_source_t mInputSource;
+    const AudioRecord::transfer_type mTransferType = AudioRecord::TRANSFER_OBTAIN;
+
+    sp<AudioCapture> mAC;
+
+    virtual void SetUp() override {
+        mAC = new AudioCapture(mInputSource, mSampleRate, mFormat, mChannelMask, mFlags, mSessionId,
+                               mTransferType);
+        ASSERT_NE(nullptr, mAC);
+        ASSERT_EQ(OK, mAC->create()) << "record creation failed";
+    }
+
+    virtual void TearDown() override {
+        if (mAC) ASSERT_EQ(OK, mAC->stop());
+    }
+};
+
+TEST_F(AudioRecordTest, TestSimpleRecord) {
+    EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+}
+
+TEST_F(AudioRecordTest, TestAudioCbNotifier) {
+    EXPECT_EQ(BAD_VALUE, mAC->getAudioRecordHandle()->addAudioDeviceCallback(nullptr));
+    sp<OnAudioDeviceUpdateNotifier> cb = new OnAudioDeviceUpdateNotifier();
+    sp<OnAudioDeviceUpdateNotifier> cbOld = new OnAudioDeviceUpdateNotifier();
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->addAudioDeviceCallback(cbOld));
+    EXPECT_EQ(INVALID_OPERATION, mAC->getAudioRecordHandle()->addAudioDeviceCallback(cbOld));
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->addAudioDeviceCallback(cb));
+    EXPECT_EQ(OK, mAC->start()) << "record creation failed";
+    EXPECT_EQ(OK, cb->waitForAudioDeviceCb());
+    EXPECT_EQ(AUDIO_IO_HANDLE_NONE, cbOld->mAudioIo);
+    EXPECT_EQ(AUDIO_PORT_HANDLE_NONE, cbOld->mDeviceId);
+    EXPECT_NE(AUDIO_IO_HANDLE_NONE, cb->mAudioIo);
+    EXPECT_NE(AUDIO_PORT_HANDLE_NONE, cb->mDeviceId);
+    EXPECT_EQ(BAD_VALUE, mAC->getAudioRecordHandle()->removeAudioDeviceCallback(nullptr));
+    EXPECT_EQ(INVALID_OPERATION, mAC->getAudioRecordHandle()->removeAudioDeviceCallback(cbOld));
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->removeAudioDeviceCallback(cb));
+    mAC->stop();
+}
+
+TEST_F(AudioRecordTest, TestEventRecordTrackPause) {
+    std::unique_ptr<AudioPlayback> playback{
+            new AudioPlayback(8000, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_MONO)};
+    ASSERT_EQ(OK, playback->loadResource("/data/local/tmp/bbb_1ch_8kHz_s16le.raw"))
+            << "Unable to open Resource";
+    EXPECT_EQ(OK, playback->create()) << "AudioTrack Creation failed";
+    audio_session_t audioTrackSession = playback->getAudioTrackHandle()->getSessionId();
+    EXPECT_EQ(OK, mAC->start(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE, audioTrackSession))
+            << "record creation failed";
+    EXPECT_EQ(OK, playback->start());
+    RawBuffer buffer;
+    status_t status = mAC->obtainBufferCb(buffer);
+    EXPECT_EQ(status, TIMED_OUT) << "Not expecting any callbacks until track sends Sync event";
+    playback->getAudioTrackHandle()->pause();
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+    playback->stop();
+}
+
+TEST_F(AudioRecordTest, TestEventRecordTrackStop) {
+    std::unique_ptr<AudioPlayback> playback{
+            new AudioPlayback(8000, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_MONO)};
+    ASSERT_EQ(OK, playback->loadResource("/data/local/tmp/bbb_1ch_8kHz_s16le.raw"))
+            << "Unable to open Resource";
+    EXPECT_EQ(OK, playback->create()) << "AudioTrack Creation failed";
+    audio_session_t audioTrackSession = playback->getAudioTrackHandle()->getSessionId();
+    EXPECT_EQ(OK, mAC->start(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE, audioTrackSession))
+            << "record creation failed";
+    EXPECT_EQ(OK, playback->start());
+    RawBuffer buffer;
+    status_t status = mAC->obtainBufferCb(buffer);
+    EXPECT_EQ(status, TIMED_OUT) << "Not expecting any callbacks until track sends Sync event";
+    playback->stop();
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+}
+
+TEST_F(AudioRecordTest, TestGetSetMarker) {
+    mAC->mMarkerPosition = (mAC->mNotificationFrames << 3) + (mAC->mNotificationFrames >> 1);
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->setMarkerPosition(mAC->mMarkerPosition))
+            << "setMarkerPosition() failed";
+    uint32_t marker;
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getMarkerPosition(&marker))
+            << "getMarkerPosition() failed";
+    EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+    EXPECT_EQ(marker, mAC->mMarkerPosition)
+            << "configured marker and received marker are different";
+    EXPECT_EQ(mAC->mReceivedCbMarkerAtPosition, mAC->mMarkerPosition)
+            << "configured marker and received cb marker are different";
+}
+
+TEST_F(AudioRecordTest, TestGetSetMarkerPeriodical) {
+    mAC->mMarkerPeriod = (mAC->mNotificationFrames << 3) + (mAC->mNotificationFrames >> 1);
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->setPositionUpdatePeriod(mAC->mMarkerPeriod))
+            << "setPositionUpdatePeriod() failed";
+    uint32_t marker;
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getPositionUpdatePeriod(&marker))
+            << "getPositionUpdatePeriod() failed";
+    EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+    EXPECT_EQ(marker, mAC->mMarkerPeriod) << "configured marker and received marker are different";
+    EXPECT_EQ(mAC->mReceivedCbMarkerCount, mAC->mNumFramesToRecord / mAC->mMarkerPeriod)
+            << "configured marker and received cb marker are different";
+}
+
+TEST_F(AudioRecordTest, TestMicDirectionConfiguration) {
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->setPreferredMicrophoneDirection(MIC_DIRECTION_FRONT))
+            << "setPreferredMicrophoneDirection() Failed";
+    EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+}
+
+TEST_F(AudioRecordTest, TestMicFieldConfiguration) {
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->setPreferredMicrophoneFieldDimension(0.5f))
+            << "setPreferredMicrophoneFieldDimension() Failed";
+    EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+}
+
+TEST_F(AudioRecordTest, TestGetPosition) {
+    uint32_t position;
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getPosition(&position)) << "getPosition() failed";
+    EXPECT_EQ(0, position);
+    EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+    EXPECT_EQ(OK, mAC->stop());
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getPosition(&position)) << "getPosition() failed";
+}
+
+// TODO: Add checkPatchCapture(), verify the information of patch via dumpPort() and dumpPatch()
+TEST_P(AudioRecordCreateTest, TestCreateRecord) {
+    EXPECT_EQ(mFormat, mAC->getAudioRecordHandle()->format());
+    EXPECT_EQ(audio_channel_count_from_in_mask(mChannelMask),
+              mAC->getAudioRecordHandle()->channelCount());
+    if (mAC->mFrameCount != 0)
+        EXPECT_LE(mAC->mFrameCount, mAC->getAudioRecordHandle()->frameCount());
+    EXPECT_EQ(mInputSource, mAC->getAudioRecordHandle()->inputSource());
+    if (mSampleRate != 0) EXPECT_EQ(mSampleRate, mAC->getAudioRecordHandle()->getSampleRate());
+    if (mSessionId != AUDIO_SESSION_NONE)
+        EXPECT_EQ(mSessionId, mAC->getAudioRecordHandle()->getSessionId());
+    if (mTransferType != AudioRecord::TRANSFER_CALLBACK) {
+        uint32_t marker;
+        mAC->mMarkerPosition = (mAC->mNotificationFrames << 3) + (mAC->mNotificationFrames >> 1);
+        EXPECT_EQ(INVALID_OPERATION,
+                  mAC->getAudioRecordHandle()->setMarkerPosition(mAC->mMarkerPosition));
+        EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getMarkerPosition(&marker));
+        EXPECT_EQ(INVALID_OPERATION,
+                  mAC->getAudioRecordHandle()->setPositionUpdatePeriod(mAC->mMarkerPosition));
+        EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getPositionUpdatePeriod(&marker));
+    }
+    EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+}
+
+// for port primary input
+INSTANTIATE_TEST_SUITE_P(AudioRecordPrimaryInput, AudioRecordCreateTest,
+                         ::testing::Combine(::testing::Values(8000, 11025, 12000, 16000, 22050,
+                                                              24000, 32000, 44100, 48000),
+                                            ::testing::Values(AUDIO_FORMAT_PCM_8_24_BIT),
+                                            ::testing::Values(AUDIO_CHANNEL_IN_MONO,
+                                                              AUDIO_CHANNEL_IN_STEREO,
+                                                              AUDIO_CHANNEL_IN_FRONT_BACK),
+                                            ::testing::Values(AUDIO_INPUT_FLAG_NONE),
+                                            ::testing::Values(AUDIO_SESSION_NONE),
+                                            ::testing::Values(AUDIO_SOURCE_DEFAULT)));
+
+// misc
+INSTANTIATE_TEST_SUITE_P(AudioRecordMiscInput, AudioRecordCreateTest,
+                         ::testing::Combine(::testing::Values(48000),
+                                            ::testing::Values(AUDIO_FORMAT_PCM_16_BIT),
+                                            ::testing::Values(AUDIO_CHANNEL_IN_MONO),
+                                            ::testing::Values(AUDIO_INPUT_FLAG_NONE),
+                                            ::testing::Values(AUDIO_SESSION_NONE),
+                                            ::testing::Values(AUDIO_SOURCE_MIC,
+                                                              AUDIO_SOURCE_CAMCORDER,
+                                                              AUDIO_SOURCE_VOICE_RECOGNITION,
+                                                              AUDIO_SOURCE_VOICE_COMMUNICATION,
+                                                              AUDIO_SOURCE_UNPROCESSED)));
diff --git a/media/libaudioclient/tests/bbb_1ch_8kHz_s16le.raw b/media/libaudioclient/tests/bbb_1ch_8kHz_s16le.raw
new file mode 100644
index 0000000..2d1e4bf
--- /dev/null
+++ b/media/libaudioclient/tests/bbb_1ch_8kHz_s16le.raw
Binary files differ
diff --git a/media/libaudioclient/tests/bbb_2ch_24kHz_s16le.raw b/media/libaudioclient/tests/bbb_2ch_24kHz_s16le.raw
new file mode 100644
index 0000000..c8ac5f7
--- /dev/null
+++ b/media/libaudioclient/tests/bbb_2ch_24kHz_s16le.raw
Binary files differ