/* ** ** Copyright 2008, 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 "AudioRecord" #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #define WAIT_PERIOD_MS 10 namespace android { using ::android::base::StringPrintf; using android::content::AttributionSourceState; using aidl_utils::statusTFromBinderStatus; // --------------------------------------------------------------------------- // static status_t AudioRecord::getMinFrameCount( size_t* frameCount, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask) { if (frameCount == NULL) { return BAD_VALUE; } size_t size; status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size); if (status != NO_ERROR) { ALOGE("%s(): AudioSystem could not query the input buffer size for" " sampleRate %u, format %#x, channelMask %#x; status %d", __func__, sampleRate, format, channelMask, status); return status; } // We double the size of input buffer for ping pong use of record buffer. const auto frameSize = audio_bytes_per_frame( audio_channel_count_from_in_mask(channelMask), format); if (frameSize == 0 || ((*frameCount = (size * 2) / frameSize) == 0)) { ALOGE("%s(): Unsupported configuration: sampleRate %u, format %#x, channelMask %#x", __func__, sampleRate, format, channelMask); return BAD_VALUE; } return NO_ERROR; } // --------------------------------------------------------------------------- void AudioRecord::MediaMetrics::gather(const AudioRecord *record) { #define MM_PREFIX "android.media.audiorecord." // avoid cut-n-paste errors. // Java API 28 entries, do not change. mMetricsItem->setCString(MM_PREFIX "encoding", toString(record->mFormat).c_str()); mMetricsItem->setCString(MM_PREFIX "source", toString(record->mAttributes.source).c_str()); mMetricsItem->setInt32(MM_PREFIX "latency", (int32_t)record->mLatency); // bad estimate. mMetricsItem->setInt32(MM_PREFIX "samplerate", (int32_t)record->mSampleRate); mMetricsItem->setInt32(MM_PREFIX "channels", (int32_t)record->mChannelCount); // Non-API entries, these can change. mMetricsItem->setInt32(MM_PREFIX "portId", (int32_t)record->mPortId); mMetricsItem->setInt32(MM_PREFIX "frameCount", (int32_t)record->mFrameCount); mMetricsItem->setCString(MM_PREFIX "attributes", toString(record->mAttributes).c_str()); mMetricsItem->setInt64(MM_PREFIX "channelMask", (int64_t)record->mChannelMask); // log total duration recording, including anything currently running. int64_t activeNs = 0; if (mStartedNs != 0) { activeNs = systemTime() - mStartedNs; } mMetricsItem->setDouble(MM_PREFIX "durationMs", (mDurationNs + activeNs) * 1e-6); mMetricsItem->setInt64(MM_PREFIX "startCount", (int64_t)mCount); if (mLastError != NO_ERROR) { mMetricsItem->setInt32(MM_PREFIX "lastError.code", (int32_t)mLastError); mMetricsItem->setCString(MM_PREFIX "lastError.at", mLastErrorFunc.c_str()); } mMetricsItem->setCString(MM_PREFIX "logSessionId", record->mLogSessionId.c_str()); } static const char *stateToString(bool active) { return active ? "ACTIVE" : "STOPPED"; } // hand the user a snapshot of the metrics. status_t AudioRecord::getMetrics(mediametrics::Item * &item) { mMediaMetrics.gather(this); mediametrics::Item *tmp = mMediaMetrics.dup(); if (tmp == nullptr) { return BAD_VALUE; } item = tmp; return NO_ERROR; } AudioRecord::AudioRecord(const AttributionSourceState &client) : mClientAttributionSource(client) { } AudioRecord::AudioRecord( audio_source_t inputSource, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, const AttributionSourceState& client, size_t frameCount, const wp& callback, uint32_t notificationFrames, audio_session_t sessionId, transfer_type transferType, audio_input_flags_t flags, const audio_attributes_t* pAttributes, audio_port_handle_t selectedDeviceId, audio_microphone_direction_t selectedMicDirection, float microphoneFieldDimension) : mClientAttributionSource(client) { uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(mClientAttributionSource.uid)); pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid)); (void)set(inputSource, sampleRate, format, channelMask, frameCount, callback, notificationFrames, false /*threadCanCallJava*/, sessionId, transferType, flags, uid, pid, pAttributes, selectedDeviceId, selectedMicDirection, microphoneFieldDimension); } AudioRecord::~AudioRecord() { mMediaMetrics.gather(this); mediametrics::LogItem(mMetricsId) .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_DTOR) .set(AMEDIAMETRICS_PROP_CALLERNAME, mCallerName.empty() ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN : mCallerName.c_str()) .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)mStatus) .record(); stopAndJoinCallbacks(); // checks mStatus if (mStatus == NO_ERROR) { IInterface::asBinder(mAudioRecord)->unlinkToDeath(mDeathNotifier, this); mAudioRecord.clear(); mCblkMemory.clear(); mBufferMemory.clear(); IPCThreadState::self()->flushCommands(); ALOGV("%s(%d): releasing session id %d", __func__, mPortId, mSessionId); pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid)); AudioSystem::releaseAudioSessionId(mSessionId, pid); } } void AudioRecord::stopAndJoinCallbacks() { // Make sure that callback function exits in the case where // it is looping on buffer empty condition in obtainBuffer(). // Otherwise the callback thread will never exit. stop(); if (mAudioRecordThread != 0) { mAudioRecordThread->requestExit(); // see comment in AudioRecord.h mProxy->interrupt(); mAudioRecordThread->requestExitAndWait(); mAudioRecordThread.clear(); } AutoMutex lock(mLock); if (mDeviceCallback != 0 && mInput != AUDIO_IO_HANDLE_NONE) { // This may not stop all of these device callbacks! // TODO: Add some sort of protection. AudioSystem::removeAudioDeviceCallback(this, mInput, mPortId); mDeviceCallback.clear(); } } status_t AudioRecord::set( audio_source_t inputSource, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, const wp& callback, uint32_t notificationFrames, bool threadCanCallJava, audio_session_t sessionId, transfer_type transferType, audio_input_flags_t flags, uid_t uid, pid_t pid, const audio_attributes_t* pAttributes, audio_port_handle_t selectedDeviceId, audio_microphone_direction_t selectedMicDirection, float microphoneFieldDimension, int32_t maxSharedAudioHistoryMs) { status_t status = NO_ERROR; LOG_ALWAYS_FATAL_IF(mInitialized, "%s: should not be called twice", __func__); mInitialized = true; // Note mPortId is not valid until the track is created, so omit mPortId in ALOG for set. ALOGV("%s(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " "notificationFrames %u, sessionId %d, transferType %d, flags %#x, attributionSource %s" "uid %d, pid %d", __func__, inputSource, sampleRate, format, channelMask, frameCount, notificationFrames, sessionId, transferType, flags, mClientAttributionSource.toString().c_str(), uid, pid); // TODO b/182392553: refactor or remove pid_t callingPid = IPCThreadState::self()->getCallingPid(); pid_t myPid = getpid(); pid_t adjPid = pid; if (pid == -1 || (callingPid != myPid)) { adjPid = callingPid; } mClientAttributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(adjPid)); uid_t adjUid = uid; if (uid == -1 || (callingPid != myPid)) { adjUid = IPCThreadState::self()->getCallingUid(); } mClientAttributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(adjUid)); mTracker.reset(new RecordingActivityTracker()); mSelectedDeviceId = selectedDeviceId; mSelectedMicDirection = selectedMicDirection; mSelectedMicFieldDimension = microphoneFieldDimension; mMaxSharedAudioHistoryMs = maxSharedAudioHistoryMs; std::string errorMessage; // Copy the state variables early so they are available for error reporting. if (pAttributes == nullptr) { mAttributes = AUDIO_ATTRIBUTES_INITIALIZER; mAttributes.source = inputSource; if (inputSource == AUDIO_SOURCE_VOICE_COMMUNICATION || inputSource == AUDIO_SOURCE_CAMCORDER) { mAttributes.flags = static_cast( mAttributes.flags | AUDIO_FLAG_CAPTURE_PRIVATE); } } else { // stream type shouldn't be looked at, this track has audio attributes memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t)); ALOGV("%s: Building AudioRecord with attributes: source=%d flags=0x%x tags=[%s]", __func__, mAttributes.source, mAttributes.flags, mAttributes.tags); } mSampleRate = sampleRate; if (format == AUDIO_FORMAT_DEFAULT) { format = AUDIO_FORMAT_PCM_16_BIT; } if (!audio_is_linear_pcm(format)) { // Compressed capture requires direct flags = (audio_input_flags_t) (flags | AUDIO_INPUT_FLAG_DIRECT); ALOGI("%s(): Format %#x is not linear pcm. Setting DIRECT, using flags %#x", __func__, format, flags); } mFormat = format; mChannelMask = channelMask; mSessionId = sessionId; ALOGV("%s: mSessionId %d", __func__, mSessionId); mOrigFlags = mFlags = flags; mTransfer = transferType; switch (mTransfer) { case TRANSFER_DEFAULT: if (callback == nullptr || threadCanCallJava) { mTransfer = TRANSFER_SYNC; } else { mTransfer = TRANSFER_CALLBACK; } break; case TRANSFER_CALLBACK: if (callback == nullptr) { errorMessage = StringPrintf( "%s: Transfer type TRANSFER_CALLBACK but callback == nullptr", __func__); status = BAD_VALUE; goto error; } break; case TRANSFER_OBTAIN: case TRANSFER_SYNC: break; default: errorMessage = StringPrintf("%s: Invalid transfer type %d", __func__, mTransfer); status = BAD_VALUE; goto error; } // invariant that mAudioRecord != 0 is true only after set() returns successfully if (mAudioRecord != 0) { errorMessage = StringPrintf("%s: Track already in use", __func__); status = INVALID_OPERATION; goto error; } if (!audio_is_valid_format(mFormat)) { errorMessage = StringPrintf("%s: Format %#x is not valid", __func__, mFormat); status = BAD_VALUE; goto error; } if (!audio_is_input_channel(mChannelMask)) { errorMessage = StringPrintf("%s: Invalid channel mask %#x", __func__, mChannelMask); status = BAD_VALUE; goto error; } mChannelCount = audio_channel_count_from_in_mask(mChannelMask); mFrameSize = audio_bytes_per_frame(mChannelCount, mFormat); // mFrameCount is initialized in createRecord_l mReqFrameCount = frameCount; mNotificationFramesReq = notificationFrames; // mNotificationFramesAct is initialized in createRecord_l mCallback = callback; if (mCallback != nullptr) { mAudioRecordThread = new AudioRecordThread(*this); mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO); // thread begins in paused state, and will not reference us until start() } // create the IAudioRecord { AutoMutex lock(mLock); status = createRecord_l(0 /*epoch*/); } ALOGV("%s(%d): status %d", __func__, mPortId, status); if (status != NO_ERROR) { if (mAudioRecordThread != 0) { mAudioRecordThread->requestExit(); // see comment in AudioRecord.h mAudioRecordThread->requestExitAndWait(); mAudioRecordThread.clear(); } // bypass error message to avoid logging twice (createRecord_l logs the error). goto exit; } // TODO: add audio hardware input latency here mLatency = (1000LL * mFrameCount) / mSampleRate; mMarkerPosition = 0; mMarkerReached = false; mNewPosition = 0; mUpdatePeriod = 0; AudioSystem::acquireAudioSessionId(mSessionId, adjPid, adjUid); mSequence = 1; mObservedSequence = mSequence; mInOverrun = false; mFramesRead = 0; mFramesReadServerOffset = 0; error: if (status != NO_ERROR) { mMediaMetrics.markError(status, __FUNCTION__); ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str()); reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str()); } exit: mStatus = status; return status; } // ------------------------------------------------------------------------- status_t AudioRecord::start(AudioSystem::sync_event_t event, audio_session_t triggerSession) { const int64_t beginNs = systemTime(); ALOGV("%s(%d): sync event %d trigger session %d", __func__, mPortId, event, triggerSession); AutoMutex lock(mLock); status_t status = NO_ERROR; mediametrics::Defer defer([&] { mediametrics::LogItem(mMetricsId) .set(AMEDIAMETRICS_PROP_CALLERNAME, mCallerName.empty() ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN : mCallerName.c_str()) .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_START) .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs)) .set(AMEDIAMETRICS_PROP_STATE, stateToString(mActive)) .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status) .record(); }); if (mActive) { return status; } // discard data in buffer const uint32_t framesFlushed = mProxy->flush(); mFramesReadServerOffset -= mFramesRead + framesFlushed; mFramesRead = 0; mProxy->clearTimestamp(); // timestamp is invalid until next server push mPreviousTimestamp.clear(); mTimestampRetrogradePositionReported = false; mTimestampRetrogradeTimeReported = false; // reset current position as seen by client to 0 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); // force refresh of remaining frames by processAudioBuffer() as last // read before stop could be partial. mRefreshRemaining = true; mNewPosition = mProxy->getPosition() + mUpdatePeriod; int32_t flags = android_atomic_acquire_load(&mCblk->mFlags); // we reactivate markers (mMarkerPosition != 0) as the position is reset to 0. // This is legacy behavior. This is not done in stop() to avoid a race condition // where the last marker event is issued twice. mMarkerReached = false; // mActive is checked by restoreRecord_l mActive = true; if (!(flags & CBLK_INVALID)) { status = statusTFromBinderStatus(mAudioRecord->start(event, triggerSession)); if (status == DEAD_OBJECT) { flags |= CBLK_INVALID; } } if (flags & CBLK_INVALID) { status = restoreRecord_l("start"); } // Call these directly because we are already holding the lock. mAudioRecord->setPreferredMicrophoneDirection(mSelectedMicDirection); mAudioRecord->setPreferredMicrophoneFieldDimension(mSelectedMicFieldDimension); if (status != NO_ERROR) { mActive = false; ALOGE("%s(%d): status %d", __func__, mPortId, status); mMediaMetrics.markError(status, __FUNCTION__); } else { mTracker->recordingStarted(); sp t = mAudioRecordThread; if (t != 0) { t->resume(); } else { mPreviousPriority = getpriority(PRIO_PROCESS, 0); get_sched_policy(0, &mPreviousSchedulingGroup); androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); } // we've successfully started, log that time mMediaMetrics.logStart(systemTime()); } return status; } void AudioRecord::stop() { const int64_t beginNs = systemTime(); AutoMutex lock(mLock); mediametrics::Defer defer([&] { mediametrics::LogItem(mMetricsId) .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_STOP) .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs)) .set(AMEDIAMETRICS_PROP_STATE, stateToString(mActive)) .record(); }); ALOGV("%s(%d): mActive:%d\n", __func__, mPortId, mActive); if (!mActive) { return; } mActive = false; mProxy->interrupt(); mAudioRecord->stop(); mTracker->recordingStopped(); // Note: legacy handling - stop does not clear record marker and // periodic update position; we update those on start(). sp t = mAudioRecordThread; if (t != 0) { t->pause(); } else { setpriority(PRIO_PROCESS, 0, mPreviousPriority); set_sched_policy(0, mPreviousSchedulingGroup); } // we've successfully started, log that time mMediaMetrics.logStop(systemTime()); } bool AudioRecord::stopped() const { AutoMutex lock(mLock); return !mActive; } status_t AudioRecord::setMarkerPosition(uint32_t marker) { AutoMutex lock(mLock); // The only purpose of setting marker position is to get a callback if (mCallback == nullptr) { return INVALID_OPERATION; } mMarkerPosition = marker; mMarkerReached = false; sp t = mAudioRecordThread; if (t != 0) { t->wake(); } return NO_ERROR; } uint32_t AudioRecord::getHalSampleRate() const { return mHalSampleRate; } uint32_t AudioRecord::getHalChannelCount() const { return mHalChannelCount; } audio_format_t AudioRecord::getHalFormat() const { return mHalFormat; } status_t AudioRecord::getMarkerPosition(uint32_t *marker) const { if (marker == NULL) { return BAD_VALUE; } AutoMutex lock(mLock); mMarkerPosition.getValue(marker); return NO_ERROR; } status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod) { AutoMutex lock(mLock); // The only purpose of setting position update period is to get a callback if (mCallback == nullptr) { return INVALID_OPERATION; } mNewPosition = mProxy->getPosition() + updatePeriod; mUpdatePeriod = updatePeriod; sp t = mAudioRecordThread; if (t != 0) { t->wake(); } return NO_ERROR; } status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod) const { if (updatePeriod == NULL) { return BAD_VALUE; } AutoMutex lock(mLock); *updatePeriod = mUpdatePeriod; return NO_ERROR; } status_t AudioRecord::getPosition(uint32_t *position) const { if (position == NULL) { return BAD_VALUE; } AutoMutex lock(mLock); mProxy->getPosition().getValue(position); return NO_ERROR; } uint32_t AudioRecord::getInputFramesLost() const { // no need to check mActive, because if inactive this will return 0, which is what we want return AudioSystem::getInputFramesLost(getInputPrivate()); } status_t AudioRecord::getTimestamp(ExtendedTimestamp *timestamp) { if (timestamp == nullptr) { return BAD_VALUE; } AutoMutex lock(mLock); status_t status = mProxy->getTimestamp(timestamp); if (status == OK) { timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesRead; timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0; if (!audio_is_linear_pcm(mFormat)) { // Don't do retrograde corrections or server offset if track is // compressed return OK; } // server side frame offset in case AudioRecord has been restored. for (int i = ExtendedTimestamp::LOCATION_SERVER; i < ExtendedTimestamp::LOCATION_MAX; ++i) { if (timestamp->mTimeNs[i] >= 0) { timestamp->mPosition[i] += mFramesReadServerOffset; } } bool timestampRetrogradeTimeReported = false; bool timestampRetrogradePositionReported = false; for (int i = 0; i < ExtendedTimestamp::LOCATION_MAX; ++i) { if (timestamp->mTimeNs[i] >= 0 && mPreviousTimestamp.mTimeNs[i] >= 0) { if (timestamp->mTimeNs[i] < mPreviousTimestamp.mTimeNs[i]) { if (!mTimestampRetrogradeTimeReported) { ALOGD("%s: retrograde time adjusting [%d] current:%lld to previous:%lld", __func__, i, (long long)timestamp->mTimeNs[i], (long long)mPreviousTimestamp.mTimeNs[i]); timestampRetrogradeTimeReported = true; } timestamp->mTimeNs[i] = mPreviousTimestamp.mTimeNs[i]; } if (timestamp->mPosition[i] < mPreviousTimestamp.mPosition[i]) { if (!mTimestampRetrogradePositionReported) { ALOGD("%s: retrograde position" " adjusting [%d] current:%lld to previous:%lld", __func__, i, (long long)timestamp->mPosition[i], (long long)mPreviousTimestamp.mPosition[i]); timestampRetrogradePositionReported = true; } timestamp->mPosition[i] = mPreviousTimestamp.mPosition[i]; } } } mPreviousTimestamp = *timestamp; if (timestampRetrogradeTimeReported) { mTimestampRetrogradeTimeReported = true; } if (timestampRetrogradePositionReported) { mTimestampRetrogradePositionReported = true; } } return status; } // ---- Explicit Routing --------------------------------------------------- status_t AudioRecord::setInputDevice(audio_port_handle_t deviceId) { AutoMutex lock(mLock); ALOGV("%s(%d): deviceId=%d mSelectedDeviceId=%d", __func__, mPortId, deviceId, mSelectedDeviceId); if (mSelectedDeviceId != deviceId) { mSelectedDeviceId = deviceId; if (mStatus == NO_ERROR) { if (mActive) { if (mSelectedDeviceId != mRoutedDeviceId) { // stop capture so that audio policy manager does not reject the new instance // start request as only one capture can be active at a time. if (mAudioRecord != 0) { mAudioRecord->stop(); } android_atomic_or(CBLK_INVALID, &mCblk->mFlags); mProxy->interrupt(); } } else { // if the track is idle, try to restore now and // defer to next start if not possible if (restoreRecord_l("setInputDevice") != OK) { android_atomic_or(CBLK_INVALID, &mCblk->mFlags); } } } } return NO_ERROR; } audio_port_handle_t AudioRecord::getInputDevice() { AutoMutex lock(mLock); return mSelectedDeviceId; } // must be called with mLock held void AudioRecord::updateRoutedDeviceId_l() { // if the record is inactive, do not update actual device as the input stream maybe routed // from a device not relevant to this client because of other active use cases. if (!mActive) { return; } if (mInput != AUDIO_IO_HANDLE_NONE) { audio_port_handle_t deviceId = AudioSystem::getDeviceIdForIo(mInput); if (deviceId != AUDIO_PORT_HANDLE_NONE) { mRoutedDeviceId = deviceId; } } } audio_port_handle_t AudioRecord::getRoutedDeviceId() { AutoMutex lock(mLock); updateRoutedDeviceId_l(); return mRoutedDeviceId; } status_t AudioRecord::dump(int fd, const Vector& args __unused) const { String8 result; result.append(" AudioRecord::dump\n"); result.appendFormat(" id(%d) status(%d), active(%d), session Id(%d)\n", mPortId, mStatus, mActive, mSessionId); result.appendFormat(" flags(%#x), req. flags(%#x), audio source(%d)\n", mFlags, mOrigFlags, mAttributes.source); result.appendFormat(" format(%#x), channel mask(%#x), channel count(%u), sample rate(%u)\n", mFormat, mChannelMask, mChannelCount, mSampleRate); result.appendFormat(" frame count(%zu), req. frame count(%zu)\n", mFrameCount, mReqFrameCount); result.appendFormat(" notif. frame count(%u), req. notif. frame count(%u)\n", mNotificationFramesAct, mNotificationFramesReq); result.appendFormat(" input(%d), latency(%u), selected device Id(%d), routed device Id(%d)\n", mInput, mLatency, mSelectedDeviceId, mRoutedDeviceId); result.appendFormat(" mic direction(%d) mic field dimension(%f)", mSelectedMicDirection, mSelectedMicFieldDimension); ::write(fd, result.c_str(), result.size()); return NO_ERROR; } // ------------------------------------------------------------------------- // TODO Move this macro to a common header file for enum to string conversion in audio framework. #define MEDIA_CASE_ENUM(name) case name: return #name const char * AudioRecord::convertTransferToText(transfer_type transferType) { switch (transferType) { MEDIA_CASE_ENUM(TRANSFER_DEFAULT); MEDIA_CASE_ENUM(TRANSFER_CALLBACK); MEDIA_CASE_ENUM(TRANSFER_OBTAIN); MEDIA_CASE_ENUM(TRANSFER_SYNC); default: return "UNRECOGNIZED"; } } // must be called with mLock held status_t AudioRecord::createRecord_l(const Modulo &epoch) { const int64_t beginNs = systemTime(); const sp& audioFlinger = AudioSystem::get_audio_flinger(); IAudioFlinger::CreateRecordInput input; IAudioFlinger::CreateRecordOutput output; [[maybe_unused]] audio_session_t originalSessionId; void *iMemPointer; audio_track_cblk_t* cblk; status_t status; static const int32_t kMaxCreateAttempts = 3; int32_t remainingAttempts = kMaxCreateAttempts; std::string errorMessage; if (audioFlinger == 0) { errorMessage = StringPrintf("%s(%d): Could not get audioflinger", __func__, mPortId); status = NO_INIT; goto exit; } // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted. // After fast request is denied, we will request again if IAudioRecord is re-created. // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, // we must release it ourselves if anything goes wrong. // Client can only express a preference for FAST. Server will perform additional tests. if (mFlags & AUDIO_INPUT_FLAG_FAST) { bool useCaseAllowed = // any of these use cases: // use case 1: callback transfer mode (mTransfer == TRANSFER_CALLBACK) || // use case 2: blocking read mode // The default buffer capacity at 48 kHz is 2048 frames, or ~42.6 ms. // That's enough for double-buffering with our standard 20 ms rule of thumb for // the minimum period of a non-SCHED_FIFO thread. // This is needed so that AAudio apps can do a low latency non-blocking read from a // callback running with SCHED_FIFO. (mTransfer == TRANSFER_SYNC) || // use case 3: obtain/release mode (mTransfer == TRANSFER_OBTAIN); if (!useCaseAllowed) { ALOGD("%s(%d): AUDIO_INPUT_FLAG_FAST denied, incompatible transfer = %s", __func__, mPortId, convertTransferToText(mTransfer)); mFlags = (audio_input_flags_t) (mFlags & ~(AUDIO_INPUT_FLAG_FAST | AUDIO_INPUT_FLAG_RAW)); } } input.attr = mAttributes; input.config.sample_rate = mSampleRate; input.config.channel_mask = mChannelMask; input.config.format = mFormat; input.clientInfo.attributionSource = mClientAttributionSource; input.clientInfo.clientTid = -1; if (mFlags & AUDIO_INPUT_FLAG_FAST) { if (mAudioRecordThread != 0) { input.clientInfo.clientTid = mAudioRecordThread->getTid(); } } input.riid = mTracker->getRiid(); input.flags = mFlags; // The notification frame count is the period between callbacks, as suggested by the client // but moderated by the server. For record, the calculations are done entirely on server side. input.frameCount = mReqFrameCount; input.notificationFrameCount = mNotificationFramesReq; input.selectedDeviceId = mSelectedDeviceId; input.sessionId = mSessionId; originalSessionId = mSessionId; input.maxSharedAudioHistoryMs = mMaxSharedAudioHistoryMs; do { media::CreateRecordResponse response; status = audioFlinger->createRecord(VALUE_OR_FATAL(input.toAidl()), response); output = VALUE_OR_FATAL(IAudioFlinger::CreateRecordOutput::fromAidl(response)); if (status == NO_ERROR) { break; } if (status != FAILED_TRANSACTION || --remainingAttempts <= 0) { errorMessage = StringPrintf( "%s(%d): AudioFlinger could not create record track, status: %d", __func__, mPortId, status); goto exit; } // FAILED_TRANSACTION happens under very specific conditions causing a state mismatch // between audio policy manager and audio flinger during the input stream open sequence // and can be recovered by retrying. // Leave time for race condition to clear before retrying and randomize delay // to reduce the probability of concurrent retries in locked steps. usleep((20 + rand() % 30) * 10000); } while (1); ALOG_ASSERT(output.audioRecord != 0); // AudioFlinger now owns the reference to the I/O handle, // so we are no longer responsible for releasing it. mAwaitBoost = false; if (output.flags & AUDIO_INPUT_FLAG_FAST) { ALOGI("%s(%d): AUDIO_INPUT_FLAG_FAST successful; frameCount %zu -> %zu", __func__, mPortId, mReqFrameCount, output.frameCount); mAwaitBoost = true; } mFlags = output.flags; mRoutedDeviceId = output.selectedDeviceId; mSessionId = output.sessionId; mSampleRate = output.sampleRate; mServerConfig = output.serverConfig; mServerFrameSize = audio_bytes_per_frame( audio_channel_count_from_in_mask(mServerConfig.channel_mask), mServerConfig.format); mServerSampleSize = audio_bytes_per_sample(mServerConfig.format); mHalSampleRate = output.halConfig.sample_rate; mHalChannelCount = audio_channel_count_from_in_mask(output.halConfig.channel_mask); mHalFormat = output.halConfig.format; if (output.cblk == 0) { errorMessage = StringPrintf("%s(%d): Could not get control block", __func__, mPortId); status = NO_INIT; goto exit; } // TODO: Using unsecurePointer() has some associated security pitfalls // (see declaration for details). // Either document why it is safe in this case or address the // issue (e.g. by copying). iMemPointer = output.cblk ->unsecurePointer(); if (iMemPointer == NULL) { errorMessage = StringPrintf( "%s(%d): Could not get control block pointer", __func__, mPortId); status = NO_INIT; goto exit; } cblk = static_cast(iMemPointer); // Starting address of buffers in shared memory. // The buffers are either immediately after the control block, // or in a separate area at discretion of server. void *buffers; if (output.buffers == 0) { buffers = cblk + 1; } else { // TODO: Using unsecurePointer() has some associated security pitfalls // (see declaration for details). // Either document why it is safe in this case or address the // issue (e.g. by copying). buffers = output.buffers->unsecurePointer(); if (buffers == NULL) { errorMessage = StringPrintf( "%s(%d): Could not get buffer pointer", __func__, mPortId); status = NO_INIT; goto exit; } } // invariant that mAudioRecord != 0 is true only after set() returns successfully if (mAudioRecord != 0) { IInterface::asBinder(mAudioRecord)->unlinkToDeath(mDeathNotifier, this); mDeathNotifier.clear(); } mAudioRecord = output.audioRecord; mCblkMemory = output.cblk; mBufferMemory = output.buffers; IPCThreadState::self()->flushCommands(); mCblk = cblk; // note that output.frameCount is the (possibly revised) value of mReqFrameCount if (output.frameCount < mReqFrameCount || (mReqFrameCount == 0 && output.frameCount == 0)) { ALOGW("%s(%d): Requested frameCount %zu but received frameCount %zu", __func__, output.portId, mReqFrameCount, output.frameCount); } // Make sure that application is notified with sufficient margin before overrun. // The computation is done on server side. if (mNotificationFramesReq > 0 && output.notificationFrameCount != mNotificationFramesReq) { ALOGW("%s(%d): Server adjusted notificationFrames from %u to %zu for frameCount %zu", __func__, output.portId, mNotificationFramesReq, output.notificationFrameCount, output.frameCount); } mNotificationFramesAct = (uint32_t)output.notificationFrameCount; if (mServerConfig.format != mFormat && mCallback != nullptr) { mFormatConversionBufRaw = std::make_unique(mNotificationFramesAct * mFrameSize); mFormatConversionBuffer.raw = mFormatConversionBufRaw.get(); } //mInput != input includes the case where mInput == AUDIO_IO_HANDLE_NONE for first creation if (mDeviceCallback != 0) { if (mInput != AUDIO_IO_HANDLE_NONE) { AudioSystem::removeAudioDeviceCallback(this, mInput, mPortId); } AudioSystem::addAudioDeviceCallback(this, output.inputId, output.portId); } if (!mSharedAudioPackageName.empty()) { mAudioRecord->shareAudioHistory(mSharedAudioPackageName, mSharedAudioStartMs); } mPortId = output.portId; // We retain a copy of the I/O handle, but don't own the reference mInput = output.inputId; mRefreshRemaining = true; mFrameCount = output.frameCount; // If IAudioRecord is re-created, don't let the requested frameCount // decrease. This can confuse clients that cache frameCount(). if (mFrameCount > mReqFrameCount) { mReqFrameCount = mFrameCount; } // update proxy mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mServerFrameSize); mProxy->setEpoch(epoch); mProxy->setMinimum(mNotificationFramesAct); mDeathNotifier = new DeathNotifier(this); IInterface::asBinder(mAudioRecord)->linkToDeath(mDeathNotifier, this); mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_RECORD) + std::to_string(mPortId); mediametrics::LogItem(mMetricsId) .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE) .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs)) // the following are immutable (at least until restore) .set(AMEDIAMETRICS_PROP_FLAGS, toString(mFlags).c_str()) .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str()) .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId) .set(AMEDIAMETRICS_PROP_TRACKID, mPortId) .set(AMEDIAMETRICS_PROP_LOGSESSIONID, mLogSessionId) .set(AMEDIAMETRICS_PROP_SOURCE, toString(mAttributes.source).c_str()) .set(AMEDIAMETRICS_PROP_THREADID, (int32_t)output.inputId) .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId) .set(AMEDIAMETRICS_PROP_ROUTEDDEVICEID, (int32_t)mRoutedDeviceId) .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str()) .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask) .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount) .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate) // the following are NOT immutable .set(AMEDIAMETRICS_PROP_STATE, stateToString(mActive)) .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status) .set(AMEDIAMETRICS_PROP_SELECTEDMICDIRECTION, (int32_t)mSelectedMicDirection) .set(AMEDIAMETRICS_PROP_SELECTEDMICFIELDDIRECTION, (double)mSelectedMicFieldDimension) .record(); exit: if (status != NO_ERROR) { ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str()); reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str()); } mStatus = status; // sp track destructor will cause releaseOutput() to be called by AudioFlinger return status; } // Report error associated with the event and some configuration details. void AudioRecord::reportError(status_t status, const char *event, const char *message) const { if (status == NO_ERROR) return; // We report error on the native side because some callers do not come // from Java. // Ensure these variables are initialized in set(). mediametrics::LogItem(AMEDIAMETRICS_KEY_AUDIO_RECORD_ERROR) .set(AMEDIAMETRICS_PROP_EVENT, event) .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status) .set(AMEDIAMETRICS_PROP_STATUSMESSAGE, message) .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str()) .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId) .set(AMEDIAMETRICS_PROP_SOURCE, toString(mAttributes.source).c_str()) .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId) .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str()) .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask) .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount) .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate) .record(); } status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig) { if (audioBuffer == NULL) { if (nonContig != NULL) { *nonContig = 0; } return BAD_VALUE; } if (mTransfer != TRANSFER_OBTAIN) { audioBuffer->frameCount = 0; audioBuffer->mSize = 0; audioBuffer->raw = NULL; if (nonContig != NULL) { *nonContig = 0; } return INVALID_OPERATION; } const struct timespec *requested; struct timespec timeout; if (waitCount == -1) { requested = &ClientProxy::kForever; } else if (waitCount == 0) { requested = &ClientProxy::kNonBlocking; } else if (waitCount > 0) { time_t ms = WAIT_PERIOD_MS * (time_t) waitCount; timeout.tv_sec = ms / 1000; timeout.tv_nsec = (long) (ms % 1000) * 1000000; requested = &timeout; } else { ALOGE("%s(%d): invalid waitCount %d", __func__, mPortId, waitCount); requested = NULL; } return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig); } status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, struct timespec *elapsed, size_t *nonContig) { // previous and new IAudioRecord sequence numbers are used to detect track re-creation uint32_t oldSequence = 0; Proxy::Buffer buffer; status_t status = NO_ERROR; static const int32_t kMaxTries = 5; int32_t tryCounter = kMaxTries; do { // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to // keep them from going away if another thread re-creates the track during obtainBuffer() sp proxy; sp iMem; sp bufferMem; { // start of lock scope AutoMutex lock(mLock); uint32_t newSequence = mSequence; // did previous obtainBuffer() fail due to media server death or voluntary invalidation? if (status == DEAD_OBJECT) { // re-create track, unless someone else has already done so if (newSequence == oldSequence) { if (!audio_is_linear_pcm(mFormat)) { // If compressed capture, don't attempt to restore the track. // Return a DEAD_OBJECT error and let the caller recreate. tryCounter = 0; } else { status = restoreRecord_l("obtainBuffer"); } if (status != NO_ERROR) { buffer.mFrameCount = 0; buffer.mRaw = NULL; buffer.mNonContig = 0; break; } } } oldSequence = newSequence; // Keep the extra references proxy = mProxy; iMem = mCblkMemory; bufferMem = mBufferMemory; // Non-blocking if track is stopped if (!mActive) { requested = &ClientProxy::kNonBlocking; } } // end of lock scope buffer.mFrameCount = audioBuffer->frameCount; // FIXME starts the requested timeout and elapsed over from scratch status = proxy->obtainBuffer(&buffer, requested, elapsed); } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); audioBuffer->frameCount = buffer.mFrameCount; audioBuffer->mSize = buffer.mFrameCount * mServerFrameSize; audioBuffer->raw = buffer.mRaw; audioBuffer->sequence = oldSequence; if (nonContig != NULL) { *nonContig = buffer.mNonContig; } return status; } void AudioRecord::releaseBuffer(const Buffer* audioBuffer) { // FIXME add error checking on mode, by adding an internal version size_t stepCount = audioBuffer->frameCount; if (stepCount == 0) { return; } Proxy::Buffer buffer; buffer.mFrameCount = stepCount; buffer.mRaw = audioBuffer->raw; AutoMutex lock(mLock); if (audioBuffer->sequence != mSequence) { // This Buffer came from a different IAudioRecord instance, so ignore the releaseBuffer ALOGD("%s is no-op due to IAudioRecord sequence mismatch %u != %u", __func__, audioBuffer->sequence, mSequence); return; } mInOverrun = false; mProxy->releaseBuffer(&buffer); // the server does not automatically disable recorder on overrun, so no need to restart } audio_io_handle_t AudioRecord::getInputPrivate() const { AutoMutex lock(mLock); return mInput; } // ------------------------------------------------------------------------- ssize_t AudioRecord::read(void* buffer, size_t userSize, bool blocking) { if (mTransfer != TRANSFER_SYNC) { return INVALID_OPERATION; } if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { // Validation. user is most-likely passing an error code, and it would // make the return value ambiguous (actualSize vs error). ALOGE("%s(%d) (buffer=%p, size=%zu (%zu)", __func__, mPortId, buffer, userSize, userSize); return BAD_VALUE; } ssize_t read = 0; Buffer audioBuffer; while (userSize >= mFrameSize) { audioBuffer.frameCount = userSize / mFrameSize; status_t err = obtainBuffer(&audioBuffer, blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); if (err < 0) { if (read > 0) { break; } if (err == TIMED_OUT || err == -EINTR) { err = WOULD_BLOCK; } return ssize_t(err); } size_t bytesRead = audioBuffer.frameCount * mFrameSize; if (audio_is_linear_pcm(mFormat)) { memcpy_by_audio_format(buffer, mFormat, audioBuffer.raw, mServerConfig.format, audioBuffer.mSize / mServerSampleSize); } else { memcpy(buffer, audioBuffer.raw, audioBuffer.mSize); } buffer = ((char *) buffer) + bytesRead; userSize -= bytesRead; read += bytesRead; releaseBuffer(&audioBuffer); } if (read > 0) { mFramesRead += read / mFrameSize; // mFramesReadTime = systemTime(SYSTEM_TIME_MONOTONIC); // not provided at this time. } return read; } // ------------------------------------------------------------------------- nsecs_t AudioRecord::processAudioBuffer() { mLock.lock(); const sp callback = mCallback.promote(); if (!callback) { mCallback = nullptr; mLock.unlock(); return NS_NEVER; } if (mAwaitBoost) { mAwaitBoost = false; mLock.unlock(); static const int32_t kMaxTries = 5; int32_t tryCounter = kMaxTries; uint32_t pollUs = 10000; do { int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK; if (policy == SCHED_FIFO || policy == SCHED_RR) { break; } usleep(pollUs); pollUs <<= 1; } while (tryCounter-- > 0); if (tryCounter < 0) { ALOGE("%s(%d): did not receive expected priority boost on time", __func__, mPortId); } // Run again immediately return 0; } // Can only reference mCblk while locked int32_t flags = android_atomic_and(~CBLK_OVERRUN, &mCblk->mFlags); // Check for track invalidation if (flags & CBLK_INVALID) { (void) restoreRecord_l("processAudioBuffer"); mLock.unlock(); // Run again immediately, but with a new IAudioRecord return 0; } bool active = mActive; // Manage overrun callback, must be done under lock to avoid race with releaseBuffer() bool newOverrun = false; if (flags & CBLK_OVERRUN) { if (!mInOverrun) { mInOverrun = true; newOverrun = true; } } // Get current position of server Modulo position(mProxy->getPosition()); // Manage marker callback bool markerReached = false; Modulo markerPosition(mMarkerPosition); // FIXME fails for wraparound, need 64 bits if (!mMarkerReached && markerPosition.value() > 0 && position >= markerPosition) { mMarkerReached = markerReached = true; } // Determine the number of new position callback(s) that will be needed, while locked size_t newPosCount = 0; Modulo newPosition(mNewPosition); uint32_t updatePeriod = mUpdatePeriod; // FIXME fails for wraparound, need 64 bits if (updatePeriod > 0 && position >= newPosition) { newPosCount = ((position - newPosition).value() / updatePeriod) + 1; mNewPosition += updatePeriod * newPosCount; } // Cache other fields that will be needed soon uint32_t notificationFrames = mNotificationFramesAct; if (mRefreshRemaining) { mRefreshRemaining = false; mRemainingFrames = notificationFrames; mRetryOnPartialBuffer = false; } size_t misalignment = mProxy->getMisalignment(); uint32_t sequence = mSequence; // These fields don't need to be cached, because they are assigned only by set(): // mTransfer, mCallback, mUserData, mSampleRate, mFrameSize mLock.unlock(); // perform callbacks while unlocked if (newOverrun) { callback->onOverrun(); } if (markerReached) { callback->onMarker(markerPosition.value()); } while (newPosCount > 0) { callback->onNewPos(newPosition.value()); newPosition += updatePeriod; newPosCount--; } if (mObservedSequence != sequence) { mObservedSequence = sequence; callback->onNewIAudioRecord(); } // if inactive, then don't run me again until re-started if (!active) { return NS_INACTIVE; } // Compute the estimated time until the next timed event (position, markers) uint32_t minFrames = ~0; if (!markerReached && position < markerPosition) { minFrames = (markerPosition - position).value(); } if (updatePeriod > 0) { uint32_t remaining = (newPosition - position).value(); if (remaining < minFrames) { minFrames = remaining; } } // If > 0, poll periodically to recover from a stuck server. A good value is 2. static const uint32_t kPoll = 0; if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { minFrames = kPoll * notificationFrames; } // Convert frame units to time units nsecs_t ns = NS_WHENEVER; if (minFrames != (uint32_t) ~0) { // This "fudge factor" avoids soaking CPU, and compensates for late progress by server static const nsecs_t kFudgeNs = 10000000LL; // 10 ms ns = ((minFrames * 1000000000LL) / mSampleRate) + kFudgeNs; } // If not supplying data by EVENT_MORE_DATA, then we're done if (mTransfer != TRANSFER_CALLBACK) { return ns; } struct timespec timeout; const struct timespec *requested = &ClientProxy::kForever; if (ns != NS_WHENEVER) { timeout.tv_sec = ns / 1000000000LL; timeout.tv_nsec = ns % 1000000000LL; ALOGV("%s(%d): timeout %ld.%03d", __func__, mPortId, timeout.tv_sec, (int) timeout.tv_nsec / 1000000); requested = &timeout; } size_t readFrames = 0; while (mRemainingFrames > 0) { Buffer audioBuffer; audioBuffer.frameCount = mRemainingFrames; size_t nonContig; status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), "%s(%d): obtainBuffer() err=%d frameCount=%zu", __func__, mPortId, err, audioBuffer.frameCount); requested = &ClientProxy::kNonBlocking; size_t avail = audioBuffer.frameCount + nonContig; ALOGV("%s(%d): obtainBuffer(%u) returned %zu = %zu + %zu err %d", __func__, mPortId, mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); if (err != NO_ERROR) { if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) { break; } ALOGE("%s(%d): Error %d obtaining an audio buffer, giving up.", __func__, mPortId, err); return NS_NEVER; } if (mRetryOnPartialBuffer) { mRetryOnPartialBuffer = false; if (avail < mRemainingFrames) { int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / mSampleRate; if (ns < 0 || myns < ns) { ns = myns; } return ns; } } Buffer* buffer = &audioBuffer; if (mServerConfig.format != mFormat) { buffer = &mFormatConversionBuffer; buffer->frameCount = audioBuffer.frameCount; buffer->mSize = buffer->frameCount * mFrameSize; buffer->sequence = audioBuffer.sequence; memcpy_by_audio_format(buffer->raw, mFormat, audioBuffer.raw, mServerConfig.format, audioBuffer.size() / mServerSampleSize); } const size_t reqSize = buffer->size(); const size_t readSize = callback->onMoreData(*buffer); buffer->mSize = readSize; // Validate on returned size if (ssize_t(readSize) < 0 || readSize > reqSize) { ALOGE("%s(%d): EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes", __func__, mPortId, reqSize, ssize_t(readSize)); return NS_NEVER; } if (readSize == 0) { // The callback is done consuming buffers // Keep this thread going to handle timed events and // still try to provide more data in intervals of WAIT_PERIOD_MS // but don't just loop and block the CPU, so wait return WAIT_PERIOD_MS * 1000000LL; } size_t releasedFrames = readSize / mFrameSize; audioBuffer.frameCount = releasedFrames; mRemainingFrames -= releasedFrames; if (misalignment >= releasedFrames) { misalignment -= releasedFrames; } else { misalignment = 0; } releaseBuffer(&audioBuffer); readFrames += releasedFrames; // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer // if callback doesn't like to accept the full chunk if (readSize < reqSize) { continue; } // There could be enough non-contiguous frames available to satisfy the remaining request if (mRemainingFrames <= nonContig) { continue; } #if 0 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA // that total to a sum == notificationFrames. if (0 < misalignment && misalignment <= mRemainingFrames) { mRemainingFrames = misalignment; return (mRemainingFrames * 1100000000LL) / mSampleRate; } #endif } if (readFrames > 0) { AutoMutex lock(mLock); mFramesRead += readFrames; // mFramesReadTime = systemTime(SYSTEM_TIME_MONOTONIC); // not provided at this time. } mRemainingFrames = notificationFrames; mRetryOnPartialBuffer = true; // A lot has transpired since ns was calculated, so run again immediately and re-calculate return 0; } status_t AudioRecord::restoreRecord_l(const char *from) { status_t result = NO_ERROR; // logged: make sure to set this before returning. const int64_t beginNs = systemTime(); mediametrics::Defer defer([&] { mediametrics::LogItem(mMetricsId) .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_RESTORE) .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs)) .set(AMEDIAMETRICS_PROP_STATE, stateToString(mActive)) .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)result) .set(AMEDIAMETRICS_PROP_WHERE, from) .record(); }); ALOGW("%s(%d) called from %s()", __func__, mPortId, from); ++mSequence; const int INITIAL_RETRIES = 3; int retries = INITIAL_RETRIES; retry: if (retries < INITIAL_RETRIES) { // refresh the audio configuration cache in this process to make sure we get new // input parameters and new IAudioRecord in createRecord_l() AudioSystem::clearAudioConfigCache(); } mFlags = mOrigFlags; // if the new IAudioRecord is created, createRecord_l() will modify the // following member variables: mAudioRecord, mCblkMemory, mCblk, mBufferMemory. // It will also delete the strong references on previous IAudioRecord and IMemory Modulo position(mProxy->getPosition()); mNewPosition = position + mUpdatePeriod; result = createRecord_l(position); if (result == NO_ERROR) { if (mActive) { // callback thread or sync event hasn't changed // FIXME this fails if we have a new AudioFlinger instance result = statusTFromBinderStatus(mAudioRecord->start( AudioSystem::SYNC_EVENT_SAME, AUDIO_SESSION_NONE)); } mFramesReadServerOffset = mFramesRead; // server resets to zero so we need an offset. } if (result != NO_ERROR) { ALOGW("%s(%d): failed status %d, retries %d", __func__, mPortId, result, retries); if (--retries > 0) { // leave time for an eventual race condition to clear before retrying usleep(500000); goto retry; } // if no retries left, set invalid bit to force restoring at next occasion // and avoid inconsistent active state on client and server sides if (mCblk != nullptr) { android_atomic_or(CBLK_INVALID, &mCblk->mFlags); } } return result; } status_t AudioRecord::addAudioDeviceCallback(const sp& callback) { if (callback == 0) { ALOGW("%s(%d): adding NULL callback!", __func__, mPortId); return BAD_VALUE; } AutoMutex lock(mLock); if (mDeviceCallback.unsafe_get() == callback.get()) { ALOGW("%s(%d): adding same callback!", __func__, mPortId); return INVALID_OPERATION; } status_t status = NO_ERROR; if (mInput != AUDIO_IO_HANDLE_NONE) { if (mDeviceCallback != 0) { ALOGW("%s(%d): callback already present!", __func__, mPortId); AudioSystem::removeAudioDeviceCallback(this, mInput, mPortId); } status = AudioSystem::addAudioDeviceCallback(this, mInput, mPortId); } mDeviceCallback = callback; return status; } status_t AudioRecord::removeAudioDeviceCallback( const sp& callback) { if (callback == 0) { ALOGW("%s(%d): removing NULL callback!", __func__, mPortId); return BAD_VALUE; } AutoMutex lock(mLock); if (mDeviceCallback.unsafe_get() != callback.get()) { ALOGW("%s(%d): removing different callback!", __func__, mPortId); return INVALID_OPERATION; } mDeviceCallback.clear(); if (mInput != AUDIO_IO_HANDLE_NONE) { AudioSystem::removeAudioDeviceCallback(this, mInput, mPortId); } return NO_ERROR; } void AudioRecord::onAudioDeviceUpdate(audio_io_handle_t audioIo, audio_port_handle_t deviceId) { sp callback; { AutoMutex lock(mLock); if (audioIo != mInput) { return; } callback = mDeviceCallback.promote(); // only update device if the record is active as route changes due to other use cases are // irrelevant for this client if (mActive) { mRoutedDeviceId = deviceId; } } if (callback.get() != nullptr) { callback->onAudioDeviceUpdate(mInput, mRoutedDeviceId); } } // ------------------------------------------------------------------------- status_t AudioRecord::getActiveMicrophones(std::vector* activeMicrophones) { AutoMutex lock(mLock); return statusTFromBinderStatus(mAudioRecord->getActiveMicrophones(activeMicrophones)); } status_t AudioRecord::setPreferredMicrophoneDirection(audio_microphone_direction_t direction) { AutoMutex lock(mLock); if (mSelectedMicDirection == direction) { // NOP return OK; } mSelectedMicDirection = direction; if (mAudioRecord == 0) { // the internal AudioRecord hasn't be created yet, so just stash the attribute. return OK; } else { return statusTFromBinderStatus(mAudioRecord->setPreferredMicrophoneDirection(direction)); } } status_t AudioRecord::setPreferredMicrophoneFieldDimension(float zoom) { AutoMutex lock(mLock); if (mSelectedMicFieldDimension == zoom) { // NOP return OK; } mSelectedMicFieldDimension = zoom; if (mAudioRecord == 0) { // the internal AudioRecord hasn't be created yet, so just stash the attribute. return OK; } else { return statusTFromBinderStatus(mAudioRecord->setPreferredMicrophoneFieldDimension(zoom)); } } void AudioRecord::setLogSessionId(const char *logSessionId) { AutoMutex lock(mLock); if (logSessionId == nullptr) logSessionId = ""; // an empty string is an unset session id. if (mLogSessionId == logSessionId) return; mLogSessionId = logSessionId; mediametrics::LogItem(mMetricsId) .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETLOGSESSIONID) .set(AMEDIAMETRICS_PROP_LOGSESSIONID, logSessionId) .record(); } status_t AudioRecord::shareAudioHistory(const std::string& sharedPackageName, int64_t sharedStartMs) { AutoMutex lock(mLock); if (mAudioRecord == 0) { return NO_INIT; } status_t status = statusTFromBinderStatus( mAudioRecord->shareAudioHistory(sharedPackageName, sharedStartMs)); if (status == NO_ERROR) { mSharedAudioPackageName = sharedPackageName; mSharedAudioStartMs = sharedStartMs; } return status; } // ========================================================================= void AudioRecord::DeathNotifier::binderDied(const wp& who __unused) { sp audioRecord = mAudioRecord.promote(); if (audioRecord != 0) { AutoMutex lock(audioRecord->mLock); audioRecord->mProxy->binderDied(); } } // ========================================================================= AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver) : Thread(true /* bCanCallJava */) // binder recursion on restoreRecord_l() may call Java. , mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), mIgnoreNextPausedInt(false) { } AudioRecord::AudioRecordThread::~AudioRecordThread() { } bool AudioRecord::AudioRecordThread::threadLoop() { { AutoMutex _l(mMyLock); if (mPaused) { // TODO check return value and handle or log mMyCond.wait(mMyLock); // caller will check for exitPending() return true; } if (mIgnoreNextPausedInt) { mIgnoreNextPausedInt = false; mPausedInt = false; } if (mPausedInt) { if (mPausedNs > 0) { // TODO check return value and handle or log (void) mMyCond.waitRelative(mMyLock, mPausedNs); } else { // TODO check return value and handle or log mMyCond.wait(mMyLock); } mPausedInt = false; return true; } } if (exitPending()) { return false; } nsecs_t ns = mReceiver.processAudioBuffer(); switch (ns) { case 0: return true; case NS_INACTIVE: pauseInternal(); return true; case NS_NEVER: return false; case NS_WHENEVER: // Event driven: call wake() when callback notifications conditions change. ns = INT64_MAX; FALLTHROUGH_INTENDED; default: LOG_ALWAYS_FATAL_IF(ns < 0, "%s() returned %lld", __func__, (long long)ns); pauseInternal(ns); return true; } } void AudioRecord::AudioRecordThread::requestExit() { // must be in this order to avoid a race condition Thread::requestExit(); resume(); } void AudioRecord::AudioRecordThread::pause() { AutoMutex _l(mMyLock); mPaused = true; } void AudioRecord::AudioRecordThread::resume() { AutoMutex _l(mMyLock); mIgnoreNextPausedInt = true; if (mPaused || mPausedInt) { mPaused = false; mPausedInt = false; mMyCond.signal(); } } void AudioRecord::AudioRecordThread::wake() { AutoMutex _l(mMyLock); if (!mPaused) { // wake() might be called while servicing a callback - ignore the next // pause time and call processAudioBuffer. mIgnoreNextPausedInt = true; if (mPausedInt && mPausedNs > 0) { // audio record is active and internally paused with timeout. mPausedInt = false; mMyCond.signal(); } } } void AudioRecord::AudioRecordThread::pauseInternal(nsecs_t ns) { AutoMutex _l(mMyLock); mPausedInt = true; mPausedNs = ns; } // ------------------------------------------------------------------------- } // namespace android