Add audiorecord unit tests
Bug: 215776959
Test: atest audiorecord_tests
Merged-In: I3aea595620424459209e4789125b51ea0afb708e
Change-Id: I3aea595620424459209e4789125b51ea0afb708e
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