| /* |
| * 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 <android-base/file.h> |
| #include <system/audio_config.h> |
| #include <utils/Log.h> |
| |
| #include "audio_test_utils.h" |
| |
| #define WAIT_PERIOD_MS 10 // from AudioTrack.cpp |
| #define MAX_WAIT_TIME_MS 5000 |
| |
| void OnAudioDeviceUpdateNotifier::onAudioDeviceUpdate(audio_io_handle_t audioIo, |
| audio_port_handle_t deviceId) { |
| std::unique_lock<std::mutex> lock{mMutex}; |
| ALOGI("%s: audioIo=%d deviceId=%d", __func__, audioIo, deviceId); |
| mAudioIo = audioIo; |
| mDeviceId = deviceId; |
| mCondition.notify_all(); |
| } |
| |
| status_t OnAudioDeviceUpdateNotifier::waitForAudioDeviceCb(audio_port_handle_t expDeviceId) { |
| std::unique_lock<std::mutex> lock{mMutex}; |
| if (mAudioIo == AUDIO_IO_HANDLE_NONE || |
| (expDeviceId != AUDIO_PORT_HANDLE_NONE && expDeviceId != mDeviceId)) { |
| mCondition.wait_for(lock, std::chrono::milliseconds(500)); |
| if (mAudioIo == AUDIO_IO_HANDLE_NONE || |
| (expDeviceId != AUDIO_PORT_HANDLE_NONE && expDeviceId != mDeviceId)) |
| return TIMED_OUT; |
| } |
| return OK; |
| } |
| |
| 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, audio_offload_info_t* info) |
| : mSampleRate(sampleRate), |
| mFormat(format), |
| mChannelMask(channelMask), |
| mFlags(flags), |
| mSessionId(sessionId), |
| mTransferType(transferType), |
| mAttributes(attributes), |
| mOffloadInfo(info) { |
| 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 /* frameCount */, |
| mFlags, nullptr /* callback */, 0 /* notificationFrames */, |
| nullptr /* sharedBuffer */, false /*canCallJava */, mSessionId, mTransferType, |
| mOffloadInfo, attributionSource, mAttributes); |
| } else if (mTransferType == AudioTrack::TRANSFER_SHARED) { |
| mTrack = new AudioTrack(AUDIO_STREAM_MUSIC, mSampleRate, mFormat, mChannelMask, mMemory, |
| mFlags, wp<AudioTrack::IAudioTrackCallback>::fromExisting(this), 0, |
| mSessionId, mTransferType, nullptr, attributionSource, mAttributes); |
| } else { |
| ALOGE("Test application is not handling transfer type %s", |
| AudioTrack::convertTransferToText(mTransferType)); |
| 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; |
| } |
| |
| void AudioPlayback::onBufferEnd() { |
| std::unique_lock<std::mutex> lock{mMutex}; |
| mStopPlaying = true; |
| mCondition.notify_all(); |
| } |
| |
| status_t AudioPlayback::fillBuffer() { |
| if (PLAY_STARTED != mState) return INVALID_OPERATION; |
| const int maxTries = MAX_WAIT_TIME_MS / WAIT_PERIOD_MS; |
| int counter = 0; |
| 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, 1, &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; |
| counter = 0; |
| } else if (WOULD_BLOCK == status) { |
| // if not received a buffer for MAX_WAIT_TIME_MS, something has gone wrong |
| if (counter == maxTries) return TIMED_OUT; |
| counter++; |
| } |
| } |
| return OK; |
| } |
| |
| status_t AudioPlayback::waitForConsumption(bool testSeek) { |
| if (PLAY_STARTED != mState) return INVALID_OPERATION; |
| |
| const int maxTries = MAX_WAIT_TIME_MS / WAIT_PERIOD_MS; |
| int counter = 0; |
| size_t totalFrameCount = mMemCapacity / mTrack->frameSize(); |
| while (!mStopPlaying && counter < maxTries) { |
| uint32_t currPosition; |
| mTrack->getPosition(&currPosition); |
| if (currPosition >= totalFrameCount) counter++; |
| |
| if (testSeek && (currPosition > totalFrameCount * 0.6)) { |
| testSeek = false; |
| if (!mTrack->hasStarted()) return BAD_VALUE; |
| mTrack->pauseAndWait(std::chrono::seconds(2)); |
| if (mTrack->hasStarted()) return BAD_VALUE; |
| mTrack->reload(); |
| mTrack->getPosition(&currPosition); |
| if (currPosition != 0) return BAD_VALUE; |
| mTrack->start(); |
| while (currPosition < totalFrameCount * 0.3) { |
| mTrack->getPosition(&currPosition); |
| } |
| mTrack->pauseAndWait(std::chrono::seconds(2)); |
| uint32_t setPosition = totalFrameCount * 0.9; |
| mTrack->setPosition(setPosition); |
| uint32_t bufferPosition; |
| mTrack->getBufferPosition(&bufferPosition); |
| if (bufferPosition != setPosition) return BAD_VALUE; |
| mTrack->start(); |
| } |
| std::this_thread::sleep_for(std::chrono::milliseconds(WAIT_PERIOD_MS)); |
| } |
| if (!mStopPlaying && counter == maxTries) return TIMED_OUT; |
| return OK; |
| } |
| |
| status_t AudioPlayback::onProcess(bool testSeek) { |
| if (mTransferType == AudioTrack::TRANSFER_SHARED) |
| return waitForConsumption(testSeek); |
| 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 && mState != PLAY_NO_INIT) { |
| int32_t msec = 0; |
| (void)mTrack->pendingDuration(&msec); |
| mTrack->stopAndJoinCallbacks(); |
| LOG_FATAL_IF(true != mTrack->stopped()); |
| mState = PLAY_STOPPED; |
| if (msec > 0) { |
| ALOGD("deleting recycled track, waiting for data drain (%d msec)", msec); |
| usleep(msec * 1000LL); |
| } |
| } |
| } |
| |
| // 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, |
| const audio_attributes_t* attributes) |
| : mInputSource(inputSource), |
| mSampleRate(sampleRate), |
| mFormat(format), |
| mChannelMask(channelMask), |
| mFlags(flags), |
| mSessionId(sessionId), |
| mTransferType(transferType), |
| mAttributes(attributes) { |
| 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; |
| } |
| |
| 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"); |
| mFrameCount = 0; |
| mNotificationFrames = 0; |
| } |
| 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) { |
| if (mSampleRate == 48000) { // test all available constructors |
| mRecord = new AudioRecord(mInputSource, mSampleRate, mFormat, mChannelMask, |
| attributionSource, mFrameCount, nullptr /* callback */, |
| mNotificationFrames, mSessionId, mTransferType, mFlags, |
| mAttributes); |
| } else { |
| mRecord = new AudioRecord(attributionSource); |
| status = mRecord->set(mInputSource, mSampleRate, mFormat, mChannelMask, mFrameCount, |
| nullptr /* callback */, 0 /* notificationFrames */, |
| false /* canCallJava */, mSessionId, mTransferType, mFlags, |
| attributionSource.uid, attributionSource.pid, mAttributes); |
| } |
| 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, mAttributes); |
| } 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; |
| if (mFlags & AUDIO_INPUT_FLAG_FAST) { |
| mFrameCount = mRecord->frameCount(); |
| mNotificationFrames = mRecord->getNotificationPeriodInFrames(); |
| mMaxBytesPerCallback = mNotificationFrames * samplesPerFrame * bytesPerSample; |
| } |
| return status; |
| } |
| |
| status_t AudioCapture::setRecordDuration(float durationInSec) { |
| if (REC_READY != mState) { |
| return INVALID_OPERATION; |
| } |
| uint32_t sampleRate = mSampleRate == 0 ? mRecord->getSampleRate() : mSampleRate; |
| mNumFramesToRecord = (sampleRate * durationInSec); |
| return OK; |
| } |
| |
| status_t AudioCapture::enableRecordDump() { |
| if (mOutFileFd != -1) { |
| return INVALID_OPERATION; |
| } |
| TemporaryFile tf("/data/local/tmp"); |
| tf.DoNotRemove(); |
| mOutFileFd = tf.release(); |
| mFileName = std::string{tf.path}; |
| return OK; |
| } |
| |
| 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 && mState != REC_NO_INIT) { |
| if (mInputSource != AUDIO_SOURCE_DEFAULT) { |
| bool state = false; |
| status = AudioSystem::isSourceActive(mInputSource, &state); |
| if (status == OK && !state) 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) return INVALID_OPERATION; |
| const int maxTries = MAX_WAIT_TIME_MS / WAIT_PERIOD_MS; |
| int counter = 0; |
| size_t nonContig = 0; |
| while (mNumFramesReceived < mNumFramesToRecord) { |
| AudioRecord::Buffer recordBuffer; |
| recordBuffer.frameCount = mNotificationFrames; |
| status_t status = mRecord->obtainBuffer(&recordBuffer, 1, &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); |
| counter = 0; |
| } else if (WOULD_BLOCK == status) { |
| // if not received a buffer for MAX_WAIT_TIME_MS, something has gone wrong |
| if (counter == maxTries) return TIMED_OUT; |
| counter++; |
| } |
| } |
| return OK; |
| } |
| |
| status_t AudioCapture::obtainBufferCb(RawBuffer& buffer) { |
| if (REC_STARTED != mState) return INVALID_OPERATION; |
| const int maxTries = MAX_WAIT_TIME_MS / WAIT_PERIOD_MS; |
| int counter = 0; |
| std::unique_lock<std::mutex> lock{mMutex}; |
| while (mBuffersReceived.empty() && !mStopRecording && counter < maxTries) { |
| mCondition.wait_for(lock, std::chrono::milliseconds(WAIT_PERIOD_MS)); |
| counter++; |
| } |
| if (!mBuffersReceived.empty()) { |
| auto it = mBuffersReceived.begin(); |
| buffer = std::move(*it); |
| mBuffersReceived.erase(it); |
| } else { |
| if (!mStopRecording && counter == maxTries) return TIMED_OUT; |
| } |
| return OK; |
| } |
| |
| status_t AudioCapture::audioProcess() { |
| RawBuffer buffer; |
| status_t status = OK; |
| while (mNumFramesReceived < mNumFramesToRecord && status == OK) { |
| if (mTransferType == AudioRecord::TRANSFER_CALLBACK) |
| status = obtainBufferCb(buffer); |
| else |
| status = obtainBuffer(buffer); |
| if (OK == status && mOutFileFd > 0) { |
| const char* ptr = static_cast<const char*>(static_cast<void*>(buffer.mData.get())); |
| write(mOutFileFd, ptr, buffer.mCapacity); |
| } |
| } |
| return OK; |
| } |
| |
| status_t listAudioPorts(std::vector<audio_port_v7>& portsVec) { |
| int attempts = 5; |
| status_t status; |
| unsigned int generation1, generation; |
| unsigned int numPorts; |
| do { |
| if (attempts-- < 0) { |
| status = TIMED_OUT; |
| break; |
| } |
| // query for number of ports. |
| numPorts = 0; |
| status = AudioSystem::listAudioPorts(AUDIO_PORT_ROLE_NONE, AUDIO_PORT_TYPE_NONE, &numPorts, |
| nullptr, &generation1); |
| if (status != NO_ERROR) { |
| ALOGE("AudioSystem::listAudioPorts returned error %d", status); |
| break; |
| } |
| portsVec.resize(numPorts); |
| status = AudioSystem::listAudioPorts(AUDIO_PORT_ROLE_NONE, AUDIO_PORT_TYPE_NONE, &numPorts, |
| portsVec.data(), &generation); |
| } while (generation1 != generation && status == NO_ERROR); |
| if (status != NO_ERROR) { |
| numPorts = 0; |
| portsVec.clear(); |
| } |
| return status; |
| } |
| |
| status_t getPortById(const audio_port_handle_t portId, audio_port_v7& port) { |
| std::vector<struct audio_port_v7> ports; |
| status_t status = listAudioPorts(ports); |
| if (status != OK) return status; |
| for (auto i = 0; i < ports.size(); i++) { |
| if (ports[i].id == portId) { |
| port = ports[i]; |
| return OK; |
| } |
| } |
| return BAD_VALUE; |
| } |
| |
| status_t getPortByAttributes(audio_port_role_t role, audio_port_type_t type, |
| audio_devices_t deviceType, const std::string& address, |
| audio_port_v7& port) { |
| std::vector<struct audio_port_v7> ports; |
| status_t status = listAudioPorts(ports); |
| if (status != OK) return status; |
| for (auto i = 0; i < ports.size(); i++) { |
| if (ports[i].role == role && ports[i].type == type && |
| ports[i].ext.device.type == deviceType && |
| !strncmp(ports[i].ext.device.address, address.c_str(), AUDIO_DEVICE_MAX_ADDRESS_LEN)) { |
| port = ports[i]; |
| return OK; |
| } |
| } |
| return BAD_VALUE; |
| } |
| |
| status_t listAudioPatches(std::vector<struct audio_patch>& patchesVec) { |
| int attempts = 5; |
| status_t status; |
| unsigned int generation1, generation; |
| unsigned int numPatches; |
| do { |
| if (attempts-- < 0) { |
| status = TIMED_OUT; |
| break; |
| } |
| // query for number of patches. |
| numPatches = 0; |
| status = AudioSystem::listAudioPatches(&numPatches, nullptr, &generation1); |
| if (status != NO_ERROR) { |
| ALOGE("AudioSystem::listAudioPatches returned error %d", status); |
| break; |
| } |
| patchesVec.resize(numPatches); |
| status = AudioSystem::listAudioPatches(&numPatches, patchesVec.data(), &generation); |
| } while (generation1 != generation && status == NO_ERROR); |
| if (status != NO_ERROR) { |
| numPatches = 0; |
| patchesVec.clear(); |
| } |
| return status; |
| } |
| |
| status_t getPatchForOutputMix(audio_io_handle_t audioIo, audio_patch& patch) { |
| std::vector<struct audio_patch> patches; |
| status_t status = listAudioPatches(patches); |
| if (status != OK) return status; |
| |
| for (auto i = 0; i < patches.size(); i++) { |
| for (auto j = 0; j < patches[i].num_sources; j++) { |
| if (patches[i].sources[j].type == AUDIO_PORT_TYPE_MIX && |
| patches[i].sources[j].ext.mix.handle == audioIo) { |
| patch = patches[i]; |
| return OK; |
| } |
| } |
| } |
| return BAD_VALUE; |
| } |
| |
| status_t getPatchForInputMix(audio_io_handle_t audioIo, audio_patch& patch) { |
| std::vector<struct audio_patch> patches; |
| status_t status = listAudioPatches(patches); |
| if (status != OK) return status; |
| |
| for (auto i = 0; i < patches.size(); i++) { |
| for (auto j = 0; j < patches[i].num_sinks; j++) { |
| if (patches[i].sinks[j].type == AUDIO_PORT_TYPE_MIX && |
| patches[i].sinks[j].ext.mix.handle == audioIo) { |
| patch = patches[i]; |
| return OK; |
| } |
| } |
| } |
| return BAD_VALUE; |
| } |
| |
| bool patchContainsOutputDevice(audio_port_handle_t deviceId, audio_patch patch) { |
| for (auto j = 0; j < patch.num_sinks; j++) { |
| if (patch.sinks[j].type == AUDIO_PORT_TYPE_DEVICE && patch.sinks[j].id == deviceId) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| bool patchContainsInputDevice(audio_port_handle_t deviceId, audio_patch patch) { |
| for (auto j = 0; j < patch.num_sources; j++) { |
| if (patch.sources[j].type == AUDIO_PORT_TYPE_DEVICE && patch.sources[j].id == deviceId) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| bool checkPatchPlayback(audio_io_handle_t audioIo, audio_port_handle_t deviceId) { |
| struct audio_patch patch; |
| if (getPatchForOutputMix(audioIo, patch) == OK) { |
| return patchContainsOutputDevice(deviceId, patch); |
| } |
| return false; |
| } |
| |
| bool checkPatchCapture(audio_io_handle_t audioIo, audio_port_handle_t deviceId) { |
| struct audio_patch patch; |
| if (getPatchForInputMix(audioIo, patch) == OK) { |
| return patchContainsInputDevice(deviceId, patch); |
| } |
| return false; |
| } |
| |
| std::string dumpPortConfig(const audio_port_config& port) { |
| auto aidlPortConfig = legacy2aidl_audio_port_config_AudioPortConfigFw(port); |
| return aidlPortConfig.ok() ? aidlPortConfig.value().toString() |
| : "Error while converting audio port config to AIDL"; |
| } |
| |
| std::string dumpPatch(const audio_patch& patch) { |
| auto aidlPatch = legacy2aidl_audio_patch_AudioPatchFw(patch); |
| return aidlPatch.ok() ? aidlPatch.value().toString() : "Error while converting patch to AIDL"; |
| } |
| |
| std::string dumpPort(const audio_port_v7& port) { |
| auto aidlPort = legacy2aidl_audio_port_v7_AudioPortFw(port); |
| return aidlPort.ok() ? aidlPort.value().toString() : "Error while converting port to AIDL"; |
| } |