| /* |
| ** |
| ** 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 <inttypes.h> |
| #include <android-base/macros.h> |
| #include <sys/resource.h> |
| |
| #include <audiomanager/AudioManager.h> |
| #include <audiomanager/IAudioManager.h> |
| #include <binder/Binder.h> |
| #include <binder/IPCThreadState.h> |
| #include <binder/IServiceManager.h> |
| #include <media/AudioRecord.h> |
| #include <utils/Log.h> |
| #include <private/media/AudioTrackShared.h> |
| #include <processgroup/sched_policy.h> |
| #include <media/IAudioFlinger.h> |
| #include <media/MediaMetricsItem.h> |
| #include <media/TypeConverter.h> |
| |
| #define WAIT_PERIOD_MS 10 |
| |
| namespace android { |
| // --------------------------------------------------------------------------- |
| |
| // 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. |
| // Assumes audio_is_linear_pcm(format) |
| if ((*frameCount = (size * 2) / (audio_channel_count_from_in_mask(channelMask) * |
| audio_bytes_per_sample(format))) == 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()); |
| } |
| } |
| |
| 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 String16 &opPackageName) |
| : mActive(false), mStatus(NO_INIT), mOpPackageName(opPackageName), |
| mSessionId(AUDIO_SESSION_ALLOCATE), |
| mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT), |
| mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE), mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE), |
| mSelectedMicDirection(MIC_DIRECTION_UNSPECIFIED), |
| mSelectedMicFieldDimension(MIC_FIELD_DIMENSION_DEFAULT) |
| { |
| } |
| |
| AudioRecord::AudioRecord( |
| audio_source_t inputSource, |
| uint32_t sampleRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| const String16& opPackageName, |
| size_t frameCount, |
| callback_t cbf, |
| void* user, |
| uint32_t notificationFrames, |
| 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) |
| : mActive(false), |
| mStatus(NO_INIT), |
| mOpPackageName(opPackageName), |
| mSessionId(AUDIO_SESSION_ALLOCATE), |
| mPreviousPriority(ANDROID_PRIORITY_NORMAL), |
| mPreviousSchedulingGroup(SP_DEFAULT), |
| mProxy(NULL) |
| { |
| (void)set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user, |
| 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(); |
| |
| if (mStatus == NO_ERROR) { |
| // 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) { |
| mProxy->interrupt(); |
| mAudioRecordThread->requestExit(); // see comment in AudioRecord.h |
| mAudioRecordThread->requestExitAndWait(); |
| mAudioRecordThread.clear(); |
| } |
| // No lock here: worst case we remove a NULL callback which will be a nop |
| if (mDeviceCallback != 0 && mInput != AUDIO_IO_HANDLE_NONE) { |
| AudioSystem::removeAudioDeviceCallback(this, mInput, mPortId); |
| } |
| 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); |
| AudioSystem::releaseAudioSessionId(mSessionId, mClientPid); |
| } |
| } |
| |
| status_t AudioRecord::set( |
| audio_source_t inputSource, |
| uint32_t sampleRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| size_t frameCount, |
| callback_t cbf, |
| void* user, |
| 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) |
| { |
| status_t status = NO_ERROR; |
| uint32_t channelCount; |
| pid_t callingPid; |
| pid_t myPid; |
| |
| // 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, opPackageName %s " |
| "uid %d, pid %d", |
| __func__, |
| inputSource, sampleRate, format, channelMask, frameCount, notificationFrames, |
| sessionId, transferType, flags, String8(mOpPackageName).string(), uid, pid); |
| |
| mTracker.reset(new RecordingActivityTracker()); |
| |
| mSelectedDeviceId = selectedDeviceId; |
| mSelectedMicDirection = selectedMicDirection; |
| mSelectedMicFieldDimension = microphoneFieldDimension; |
| |
| switch (transferType) { |
| case TRANSFER_DEFAULT: |
| if (cbf == NULL || threadCanCallJava) { |
| transferType = TRANSFER_SYNC; |
| } else { |
| transferType = TRANSFER_CALLBACK; |
| } |
| break; |
| case TRANSFER_CALLBACK: |
| if (cbf == NULL) { |
| ALOGE("%s(): Transfer type TRANSFER_CALLBACK but cbf == NULL", __func__); |
| status = BAD_VALUE; |
| goto exit; |
| } |
| break; |
| case TRANSFER_OBTAIN: |
| case TRANSFER_SYNC: |
| break; |
| default: |
| ALOGE("%s(): Invalid transfer type %d", __func__, transferType); |
| status = BAD_VALUE; |
| goto exit; |
| } |
| mTransfer = transferType; |
| |
| // invariant that mAudioRecord != 0 is true only after set() returns successfully |
| if (mAudioRecord != 0) { |
| ALOGE("%s(): Track already in use", __func__); |
| status = INVALID_OPERATION; |
| goto exit; |
| } |
| |
| if (pAttributes == NULL) { |
| mAttributes = AUDIO_ATTRIBUTES_INITIALIZER; |
| mAttributes.source = inputSource; |
| if (inputSource == AUDIO_SOURCE_VOICE_COMMUNICATION |
| || inputSource == AUDIO_SOURCE_CAMCORDER) { |
| 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; |
| |
| // these below should probably come from the audioFlinger too... |
| if (format == AUDIO_FORMAT_DEFAULT) { |
| format = AUDIO_FORMAT_PCM_16_BIT; |
| } |
| |
| // validate parameters |
| // AudioFlinger capture only supports linear PCM |
| if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { |
| ALOGE("%s(): Format %#x is not linear pcm", __func__, format); |
| status = BAD_VALUE; |
| goto exit; |
| } |
| mFormat = format; |
| |
| if (!audio_is_input_channel(channelMask)) { |
| ALOGE("%s(): Invalid channel mask %#x", __func__, channelMask); |
| status = BAD_VALUE; |
| goto exit; |
| } |
| mChannelMask = channelMask; |
| channelCount = audio_channel_count_from_in_mask(channelMask); |
| mChannelCount = channelCount; |
| |
| if (audio_is_linear_pcm(format)) { |
| mFrameSize = channelCount * audio_bytes_per_sample(format); |
| } else { |
| mFrameSize = sizeof(uint8_t); |
| } |
| |
| // mFrameCount is initialized in createRecord_l |
| mReqFrameCount = frameCount; |
| |
| mNotificationFramesReq = notificationFrames; |
| // mNotificationFramesAct is initialized in createRecord_l |
| |
| mSessionId = sessionId; |
| ALOGV("%s(): mSessionId %d", __func__, mSessionId); |
| |
| callingPid = IPCThreadState::self()->getCallingPid(); |
| myPid = getpid(); |
| if (uid == AUDIO_UID_INVALID || (callingPid != myPid)) { |
| mClientUid = IPCThreadState::self()->getCallingUid(); |
| } else { |
| mClientUid = uid; |
| } |
| if (pid == -1 || (callingPid != myPid)) { |
| mClientPid = callingPid; |
| } else { |
| mClientPid = pid; |
| } |
| |
| mOrigFlags = mFlags = flags; |
| mCbf = cbf; |
| |
| if (cbf != NULL) { |
| 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*/, mOpPackageName); |
| } |
| |
| 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(); |
| } |
| goto exit; |
| } |
| |
| mUserData = user; |
| // TODO: add audio hardware input latency here |
| mLatency = (1000LL * mFrameCount) / mSampleRate; |
| mMarkerPosition = 0; |
| mMarkerReached = false; |
| mNewPosition = 0; |
| mUpdatePeriod = 0; |
| AudioSystem::acquireAudioSessionId(mSessionId, mClientPid, mClientUid); |
| mSequence = 1; |
| mObservedSequence = mSequence; |
| mInOverrun = false; |
| mFramesRead = 0; |
| mFramesReadServerOffset = 0; |
| |
| exit: |
| mStatus = status; |
| if (status != NO_ERROR) { |
| mMediaMetrics.markError(status, __FUNCTION__); |
| } |
| 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 = mAudioRecord->start(event, triggerSession).transactionError(); |
| 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<AudioRecordThread> 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<AudioRecordThread> 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) |
| { |
| // The only purpose of setting marker position is to get a callback |
| if (mCbf == NULL) { |
| return INVALID_OPERATION; |
| } |
| |
| AutoMutex lock(mLock); |
| mMarkerPosition = marker; |
| mMarkerReached = false; |
| |
| sp<AudioRecordThread> t = mAudioRecordThread; |
| if (t != 0) { |
| t->wake(); |
| } |
| return NO_ERROR; |
| } |
| |
| 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) |
| { |
| // The only purpose of setting position update period is to get a callback |
| if (mCbf == NULL) { |
| return INVALID_OPERATION; |
| } |
| |
| AutoMutex lock(mLock); |
| mNewPosition = mProxy->getPosition() + updatePeriod; |
| mUpdatePeriod = updatePeriod; |
| |
| sp<AudioRecordThread> 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; |
| // 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); |
| if (mSelectedDeviceId != deviceId) { |
| mSelectedDeviceId = deviceId; |
| if (mStatus == NO_ERROR) { |
| // 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 && mActive) { |
| mAudioRecord->stop(); |
| } |
| android_atomic_or(CBLK_INVALID, &mCblk->mFlags); |
| mProxy->interrupt(); |
| } |
| } |
| 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<String16>& 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.string(), 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<uint32_t> &epoch, const String16& opPackageName) |
| { |
| const int64_t beginNs = systemTime(); |
| const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); |
| IAudioFlinger::CreateRecordInput input; |
| IAudioFlinger::CreateRecordOutput output; |
| audio_session_t originalSessionId; |
| sp<media::IAudioRecord> record; |
| void *iMemPointer; |
| audio_track_cblk_t* cblk; |
| status_t status; |
| static const int32_t kMaxCreateAttempts = 3; |
| int32_t remainingAttempts = kMaxCreateAttempts; |
| |
| if (audioFlinger == 0) { |
| ALOGE("%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.clientUid = mClientUid; |
| input.clientInfo.clientPid = mClientPid; |
| input.clientInfo.clientTid = -1; |
| if (mFlags & AUDIO_INPUT_FLAG_FAST) { |
| if (mAudioRecordThread != 0) { |
| input.clientInfo.clientTid = mAudioRecordThread->getTid(); |
| } |
| } |
| input.opPackageName = opPackageName; |
| 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; |
| |
| do { |
| record = audioFlinger->createRecord(input, output, &status); |
| if (status == NO_ERROR) { |
| break; |
| } |
| if (status != FAILED_TRANSACTION || --remainingAttempts <= 0) { |
| ALOGE("%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(record != 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; |
| |
| if (output.cblk == 0) { |
| ALOGE("%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) { |
| ALOGE("%s(%d): Could not get control block pointer", __func__, mPortId); |
| status = NO_INIT; |
| goto exit; |
| } |
| cblk = static_cast<audio_track_cblk_t*>(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) { |
| ALOGE("%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 = record; |
| 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; |
| |
| //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); |
| } |
| |
| 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, mFrameSize); |
| 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_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_SELECTEDMICDIRECTION, (int32_t)mSelectedMicDirection) |
| .set(AMEDIAMETRICS_PROP_SELECTEDMICFIELDDIRECTION, (double)mSelectedMicFieldDimension) |
| .record(); |
| |
| exit: |
| mStatus = status; |
| // sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger |
| return status; |
| } |
| |
| 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->size = 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<AudioRecordClientProxy> proxy; |
| sp<IMemory> iMem; |
| sp<IMemory> 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) { |
| 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->size = buffer.mFrameCount * mFrameSize; |
| 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->size / mFrameSize; |
| 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)) { |
| // sanity-check. 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.size; |
| memcpy(buffer, audioBuffer.i8, bytesRead); |
| 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(); |
| 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<uint32_t> position(mProxy->getPosition()); |
| |
| // Manage marker callback |
| bool markerReached = false; |
| Modulo<uint32_t> 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<uint32_t> 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, mCbf, mUserData, mSampleRate, mFrameSize |
| |
| mLock.unlock(); |
| |
| // perform callbacks while unlocked |
| if (newOverrun) { |
| mCbf(EVENT_OVERRUN, mUserData, NULL); |
| } |
| if (markerReached) { |
| mCbf(EVENT_MARKER, mUserData, &markerPosition); |
| } |
| while (newPosCount > 0) { |
| size_t temp = newPosition.value(); // FIXME size_t != uint32_t |
| mCbf(EVENT_NEW_POS, mUserData, &temp); |
| newPosition += updatePeriod; |
| newPosCount--; |
| } |
| if (mObservedSequence != sequence) { |
| mObservedSequence = sequence; |
| mCbf(EVENT_NEW_IAUDIORECORD, mUserData, NULL); |
| } |
| |
| // 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; |
| } |
| } |
| |
| size_t reqSize = audioBuffer.size; |
| mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); |
| size_t readSize = audioBuffer.size; |
| |
| // Sanity check 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): dead IAudioRecord, creating a new one 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<uint32_t> position(mProxy->getPosition()); |
| mNewPosition = position + mUpdatePeriod; |
| result = createRecord_l(position, mOpPackageName); |
| |
| 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 = mAudioRecord->start( |
| AudioSystem::SYNC_EVENT_SAME, AUDIO_SESSION_NONE).transactionError(); |
| } |
| 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<AudioSystem::AudioDeviceCallback>& 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<AudioSystem::AudioDeviceCallback>& 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<AudioSystem::AudioDeviceCallback> 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<media::MicrophoneInfo>* activeMicrophones) |
| { |
| AutoMutex lock(mLock); |
| return mAudioRecord->getActiveMicrophones(activeMicrophones).transactionError(); |
| } |
| |
| 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 mAudioRecord->setPreferredMicrophoneDirection(direction).transactionError(); |
| } |
| } |
| |
| 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 mAudioRecord->setPreferredMicrophoneFieldDimension(zoom).transactionError(); |
| } |
| } |
| |
| // ========================================================================= |
| |
| void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who __unused) |
| { |
| sp<AudioRecord> 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 |