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LeafOS / LeafOS-Project / android_frameworks_av / 80429eeee07721b0f3904075c86bf57deebd3df3 / . / media / libaudioclient / AudioRecord.cpp
blob: 00c6d63339e49714145935a690b119556873385f [file] [log] [blame]
/*
**
** 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 <android-base/stringprintf.h>
#include <sys/resource.h>
#include <audio_utils/format.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 {
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.
// Assumes audio_is_linear_pcm(format)
const auto sampleSize = audio_channel_count_from_in_mask(channelMask) *
audio_bytes_per_sample(format);
if (sampleSize == 0 || ((*frameCount = (size * 2) / sampleSize) == 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)
: mActive(false), mStatus(NO_INIT), mClientAttributionSource(client),
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 AttributionSourceState& client,
size_t frameCount,
const wp<IAudioRecordCallback>& 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)
: mActive(false),
mStatus(NO_INIT),
mClientAttributionSource(client),
mSessionId(AUDIO_SESSION_ALLOCATE),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mProxy(nullptr)
{
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() {
// Prevent nullptr crash if it did not open properly.
if (mStatus != NO_ERROR) return;
// 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<IAudioRecordCallback>& 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<audio_flags_mask_t>(
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);
if (audio_is_linear_pcm(mFormat)) {
mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat);
} else {
mFrameSize = sizeof(uint8_t);
}
// 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<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)
{
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<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)
{
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<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;
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) {
// 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.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<uint32_t> &epoch)
{
const int64_t beginNs = systemTime();
const sp<IAudioFlinger>& 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);
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<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) {
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<uint8_t[]>(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<IAudioTrack> 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<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) {
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;
memcpy_by_audio_format(buffer, mFormat, audioBuffer.raw, mServerConfig.format,
audioBuffer.mSize / mServerSampleSize);
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<IAudioRecordCallback> 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<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, 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): 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);
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<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::MicrophoneInfoFw>* 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<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