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-rw-r--r--media/libmedia/AudioRecord.cpp325
1 files changed, 189 insertions, 136 deletions
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index ccbc5a3..9e7ba88 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -18,7 +18,9 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "AudioRecord"
+#include <inttypes.h>
#include <sys/resource.h>
+
#include <binder/IPCThreadState.h>
#include <media/AudioRecord.h>
#include <utils/Log.h>
@@ -41,37 +43,30 @@ status_t AudioRecord::getMinFrameCount(
return BAD_VALUE;
}
- // default to 0 in case of error
- *frameCount = 0;
-
- size_t size = 0;
+ size_t size;
status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size);
if (status != NO_ERROR) {
- ALOGE("AudioSystem could not query the input buffer size; status %d", status);
- return NO_INIT;
+ ALOGE("AudioSystem could not query the input buffer size for sampleRate %u, format %#x, "
+ "channelMask %#x; status %d", sampleRate, format, channelMask, status);
+ return status;
}
- if (size == 0) {
- ALOGE("Unsupported configuration: sampleRate %u, format %d, channelMask %#x",
+ // 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("Unsupported configuration: sampleRate %u, format %#x, channelMask %#x",
sampleRate, format, channelMask);
return BAD_VALUE;
}
- // We double the size of input buffer for ping pong use of record buffer.
- size <<= 1;
-
- // Assumes audio_is_linear_pcm(format)
- uint32_t channelCount = popcount(channelMask);
- size /= channelCount * audio_bytes_per_sample(format);
-
- *frameCount = size;
return NO_ERROR;
}
// ---------------------------------------------------------------------------
AudioRecord::AudioRecord()
- : mStatus(NO_INIT), mSessionId(0),
+ : mStatus(NO_INIT), mSessionId(AUDIO_SESSION_ALLOCATE),
mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT)
{
}
@@ -81,20 +76,20 @@ AudioRecord::AudioRecord(
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- int frameCount,
+ size_t frameCount,
callback_t cbf,
void* user,
- int notificationFrames,
+ uint32_t notificationFrames,
int sessionId,
transfer_type transferType,
audio_input_flags_t flags)
- : mStatus(NO_INIT), mSessionId(0),
+ : mStatus(NO_INIT), mSessionId(AUDIO_SESSION_ALLOCATE),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mProxy(NULL)
{
mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user,
- notificationFrames, false /*threadCanCallJava*/, sessionId, transferType);
+ notificationFrames, false /*threadCanCallJava*/, sessionId, transferType, flags);
}
AudioRecord::~AudioRecord()
@@ -110,12 +105,12 @@ AudioRecord::~AudioRecord()
mAudioRecordThread->requestExitAndWait();
mAudioRecordThread.clear();
}
- if (mAudioRecord != 0) {
- mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
- mAudioRecord.clear();
- }
+ mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
+ mAudioRecord.clear();
+ mCblkMemory.clear();
+ mBufferMemory.clear();
IPCThreadState::self()->flushCommands();
- AudioSystem::releaseAudioSessionId(mSessionId);
+ AudioSystem::releaseAudioSessionId(mSessionId, -1);
}
}
@@ -124,15 +119,20 @@ status_t AudioRecord::set(
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- int frameCountInt,
+ size_t frameCount,
callback_t cbf,
void* user,
- int notificationFrames,
+ uint32_t notificationFrames,
bool threadCanCallJava,
int sessionId,
transfer_type transferType,
audio_input_flags_t flags)
{
+ ALOGV("set(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
+ "notificationFrames %u, sessionId %d, transferType %d, flags %#x",
+ inputSource, sampleRate, format, channelMask, frameCount, notificationFrames,
+ sessionId, transferType, flags);
+
switch (transferType) {
case TRANSFER_DEFAULT:
if (cbf == NULL || threadCanCallJava) {
@@ -156,23 +156,15 @@ status_t AudioRecord::set(
}
mTransfer = transferType;
- // FIXME "int" here is legacy and will be replaced by size_t later
- if (frameCountInt < 0) {
- ALOGE("Invalid frame count %d", frameCountInt);
- return BAD_VALUE;
- }
- size_t frameCount = frameCountInt;
-
- ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask,
- frameCount);
-
AutoMutex lock(mLock);
+ // invariant that mAudioRecord != 0 is true only after set() returns successfully
if (mAudioRecord != 0) {
ALOGE("Track already in use");
return INVALID_OPERATION;
}
+ // handle default values first.
if (inputSource == AUDIO_SOURCE_DEFAULT) {
inputSource = AUDIO_SOURCE_MIC;
}
@@ -191,12 +183,12 @@ status_t AudioRecord::set(
// validate parameters
if (!audio_is_valid_format(format)) {
- ALOGE("Invalid format %d", format);
+ ALOGE("Invalid format %#x", format);
return BAD_VALUE;
}
// Temporary restriction: AudioFlinger currently supports 16-bit PCM only
if (format != AUDIO_FORMAT_PCM_16_BIT) {
- ALOGE("Format %d is not supported", format);
+ ALOGE("Format %#x is not supported", format);
return BAD_VALUE;
}
mFormat = format;
@@ -206,61 +198,50 @@ status_t AudioRecord::set(
return BAD_VALUE;
}
mChannelMask = channelMask;
- uint32_t channelCount = popcount(channelMask);
+ uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
mChannelCount = channelCount;
- // Assumes audio_is_linear_pcm(format), else sizeof(uint8_t)
- mFrameSize = channelCount * audio_bytes_per_sample(format);
-
- // validate framecount
- size_t minFrameCount = 0;
- status_t status = AudioRecord::getMinFrameCount(&minFrameCount,
- sampleRate, format, channelMask);
- if (status != NO_ERROR) {
- ALOGE("getMinFrameCount() failed; status %d", status);
- return status;
+ if (audio_is_linear_pcm(format)) {
+ mFrameSize = channelCount * audio_bytes_per_sample(format);
+ } else {
+ mFrameSize = sizeof(uint8_t);
}
- ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);
- if (frameCount == 0) {
- frameCount = minFrameCount;
- } else if (frameCount < minFrameCount) {
- ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
- return BAD_VALUE;
- }
- mFrameCount = frameCount;
+ // mFrameCount is initialized in openRecord_l
+ mReqFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
- mNotificationFramesAct = 0;
+ // mNotificationFramesAct is initialized in openRecord_l
- if (sessionId == 0 ) {
- mSessionId = AudioSystem::newAudioSessionId();
+ if (sessionId == AUDIO_SESSION_ALLOCATE) {
+ mSessionId = AudioSystem::newAudioUniqueId();
} else {
mSessionId = sessionId;
}
ALOGV("set(): mSessionId %d", mSessionId);
mFlags = flags;
-
- // create the IAudioRecord
- status = openRecord_l(0 /*epoch*/);
- if (status) {
- return status;
- }
+ mCbf = cbf;
if (cbf != NULL) {
mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava);
mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
}
- mStatus = NO_ERROR;
+ // create the IAudioRecord
+ status_t status = openRecord_l(0 /*epoch*/);
- // Update buffer size in case it has been limited by AudioFlinger during track creation
- mFrameCount = mCblk->frameCount_;
+ if (status != NO_ERROR) {
+ if (mAudioRecordThread != 0) {
+ mAudioRecordThread->requestExit(); // see comment in AudioRecord.h
+ mAudioRecordThread->requestExitAndWait();
+ mAudioRecordThread.clear();
+ }
+ return status;
+ }
+ mStatus = NO_ERROR;
mActive = false;
- mCbf = cbf;
- mRefreshRemaining = true;
mUserData = user;
// TODO: add audio hardware input latency here
mLatency = (1000*mFrameCount) / sampleRate;
@@ -268,7 +249,7 @@ status_t AudioRecord::set(
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
- AudioSystem::acquireAudioSessionId(mSessionId);
+ AudioSystem::acquireAudioSessionId(mSessionId, -1);
mSequence = 1;
mObservedSequence = mSequence;
mInOverrun = false;
@@ -289,6 +270,9 @@ status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession)
// 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);
@@ -352,6 +336,7 @@ bool AudioRecord::stopped() const
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;
}
@@ -377,6 +362,7 @@ status_t AudioRecord::getMarkerPosition(uint32_t *marker) const
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;
}
@@ -412,7 +398,7 @@ status_t AudioRecord::getPosition(uint32_t *position) const
return NO_ERROR;
}
-unsigned int AudioRecord::getInputFramesLost() const
+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(getInput());
@@ -430,56 +416,79 @@ status_t AudioRecord::openRecord_l(size_t epoch)
return NO_INIT;
}
- IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;
- pid_t tid = -1;
+ // Fast tracks must be at the primary _output_ [sic] sampling rate,
+ // because there is currently no concept of a primary input sampling rate
+ uint32_t afSampleRate = AudioSystem::getPrimaryOutputSamplingRate();
+ if (afSampleRate == 0) {
+ ALOGW("getPrimaryOutputSamplingRate failed");
+ }
// Client can only express a preference for FAST. Server will perform additional tests.
- // The only supported use case for FAST is callback transfer mode.
- if (mFlags & AUDIO_INPUT_FLAG_FAST) {
- if ((mTransfer != TRANSFER_CALLBACK) || (mAudioRecordThread == 0)) {
- ALOGW("AUDIO_INPUT_FLAG_FAST denied by client");
- // once denied, do not request again if IAudioRecord is re-created
- mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
- } else {
- trackFlags |= IAudioFlinger::TRACK_FAST;
- tid = mAudioRecordThread->getTid();
- }
+ if ((mFlags & AUDIO_INPUT_FLAG_FAST) && !(
+ // use case: callback transfer mode
+ (mTransfer == TRANSFER_CALLBACK) &&
+ // matching sample rate
+ (mSampleRate == afSampleRate))) {
+ ALOGW("AUDIO_INPUT_FLAG_FAST denied by client");
+ // once denied, do not request again if IAudioRecord is re-created
+ mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
}
- mNotificationFramesAct = mNotificationFramesReq;
+ IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;
- if (!(mFlags & AUDIO_INPUT_FLAG_FAST)) {
- // Make sure that application is notified with sufficient margin before overrun
- if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount/2) {
- mNotificationFramesAct = mFrameCount/2;
+ pid_t tid = -1;
+ if (mFlags & AUDIO_INPUT_FLAG_FAST) {
+ trackFlags |= IAudioFlinger::TRACK_FAST;
+ if (mAudioRecordThread != 0) {
+ tid = mAudioRecordThread->getTid();
}
}
audio_io_handle_t input = AudioSystem::getInput(mInputSource, mSampleRate, mFormat,
- mChannelMask, mSessionId);
- if (input == 0) {
- ALOGE("Could not get audio input for record source %d", mInputSource);
+ mChannelMask, mSessionId, mFlags);
+ if (input == AUDIO_IO_HANDLE_NONE) {
+ ALOGE("Could not get audio input for record source %d, sample rate %u, format %#x, "
+ "channel mask %#x, session %d, flags %#x",
+ mInputSource, mSampleRate, mFormat, mChannelMask, mSessionId, mFlags);
return BAD_VALUE;
}
+ {
+ // 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.
+ size_t frameCount = mReqFrameCount;
+ size_t temp = frameCount; // temp may be replaced by a revised value of frameCount,
+ // but we will still need the original value also
int originalSessionId = mSessionId;
+
+ // The notification frame count is the period between callbacks, as suggested by the server.
+ size_t notificationFrames = mNotificationFramesReq;
+
+ sp<IMemory> iMem; // for cblk
+ sp<IMemory> bufferMem;
sp<IAudioRecord> record = audioFlinger->openRecord(input,
mSampleRate, mFormat,
mChannelMask,
- mFrameCount,
+ &temp,
&trackFlags,
tid,
&mSessionId,
+ &notificationFrames,
+ iMem,
+ bufferMem,
&status);
- ALOGE_IF(originalSessionId != 0 && mSessionId != originalSessionId,
+ ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId,
"session ID changed from %d to %d", originalSessionId, mSessionId);
- if (record == 0 || status != NO_ERROR) {
+ if (status != NO_ERROR) {
ALOGE("AudioFlinger could not create record track, status: %d", status);
- AudioSystem::releaseInput(input);
- return status;
+ goto release;
}
- sp<IMemory> iMem = record->getCblk();
+ ALOG_ASSERT(record != 0);
+
+ // AudioFlinger now owns the reference to the I/O handle,
+ // so we are no longer responsible for releasing it.
+
if (iMem == 0) {
ALOGE("Could not get control block");
return NO_INIT;
@@ -489,37 +498,67 @@ status_t AudioRecord::openRecord_l(size_t epoch)
ALOGE("Could not get control block pointer");
return NO_INIT;
}
+ audio_track_cblk_t* 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 (bufferMem == 0) {
+ buffers = cblk + 1;
+ } else {
+ buffers = bufferMem->pointer();
+ if (buffers == NULL) {
+ ALOGE("Could not get buffer pointer");
+ return NO_INIT;
+ }
+ }
+
+ // invariant that mAudioRecord != 0 is true only after set() returns successfully
if (mAudioRecord != 0) {
mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
mDeathNotifier.clear();
}
- mInput = input;
mAudioRecord = record;
mCblkMemory = iMem;
- audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
+ mBufferMemory = bufferMem;
+ IPCThreadState::self()->flushCommands();
+
mCblk = cblk;
- // FIXME missing fast track frameCount logic
+ // note that temp is the (possibly revised) value of frameCount
+ if (temp < frameCount || (frameCount == 0 && temp == 0)) {
+ ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp);
+ }
+ frameCount = temp;
+
mAwaitBoost = false;
if (mFlags & AUDIO_INPUT_FLAG_FAST) {
if (trackFlags & IAudioFlinger::TRACK_FAST) {
- ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %u", mFrameCount);
+ ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %zu", frameCount);
mAwaitBoost = true;
- // double-buffering is not required for fast tracks, due to tighter scheduling
- if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount) {
- mNotificationFramesAct = mFrameCount;
- }
} else {
- ALOGV("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %u", mFrameCount);
+ ALOGV("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %zu", frameCount);
// once denied, do not request again if IAudioRecord is re-created
mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
- if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount/2) {
- mNotificationFramesAct = mFrameCount/2;
- }
}
}
- // starting address of buffers in shared memory
- void *buffers = (char*)cblk + sizeof(audio_track_cblk_t);
+ // Make sure that application is notified with sufficient margin before overrun
+ if (notificationFrames == 0 || notificationFrames > frameCount) {
+ ALOGW("Received notificationFrames %zu for frameCount %zu", notificationFrames, frameCount);
+ }
+ mNotificationFramesAct = notificationFrames;
+
+ // We retain a copy of the I/O handle, but don't own the reference
+ mInput = input;
+ mRefreshRemaining = true;
+
+ mFrameCount = frameCount;
+ // If IAudioRecord is re-created, don't let the requested frameCount
+ // decrease. This can confuse clients that cache frameCount().
+ if (frameCount > mReqFrameCount) {
+ mReqFrameCount = frameCount;
+ }
// update proxy
mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize);
@@ -530,6 +569,14 @@ status_t AudioRecord::openRecord_l(size_t epoch)
mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this);
return NO_ERROR;
+ }
+
+release:
+ AudioSystem::releaseInput(input, (audio_session_t)mSessionId);
+ if (status == NO_ERROR) {
+ status = NO_INIT;
+ }
+ return status;
}
status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
@@ -580,6 +627,7 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *r
// 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);
@@ -591,6 +639,9 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *r
if (newSequence == oldSequence) {
status = restoreRecord_l("obtainBuffer");
if (status != NO_ERROR) {
+ buffer.mFrameCount = 0;
+ buffer.mRaw = NULL;
+ buffer.mNonContig = 0;
break;
}
}
@@ -600,6 +651,7 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *r
// Keep the extra references
proxy = mProxy;
iMem = mCblkMemory;
+ bufferMem = mBufferMemory;
// Non-blocking if track is stopped
if (!mActive) {
@@ -660,7 +712,7 @@ ssize_t AudioRecord::read(void* buffer, size_t userSize)
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("AudioRecord::read(buffer=%p, size=%u (%d)", buffer, userSize, userSize);
+ ALOGE("AudioRecord::read(buffer=%p, size=%zu (%zu)", buffer, userSize, userSize);
return BAD_VALUE;
}
@@ -692,7 +744,7 @@ ssize_t AudioRecord::read(void* buffer, size_t userSize)
// -------------------------------------------------------------------------
-nsecs_t AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread)
+nsecs_t AudioRecord::processAudioBuffer()
{
mLock.lock();
if (mAwaitBoost) {
@@ -760,17 +812,17 @@ nsecs_t AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread)
}
// Cache other fields that will be needed soon
- size_t notificationFrames = mNotificationFramesAct;
+ uint32_t notificationFrames = mNotificationFramesAct;
if (mRefreshRemaining) {
mRefreshRemaining = false;
mRemainingFrames = notificationFrames;
mRetryOnPartialBuffer = false;
}
size_t misalignment = mProxy->getMisalignment();
- int32_t sequence = mSequence;
+ uint32_t sequence = mSequence;
// These fields don't need to be cached, because they are assigned only by set():
- // mTransfer, mCbf, mUserData, mSampleRate
+ // mTransfer, mCbf, mUserData, mSampleRate, mFrameSize
mLock.unlock();
@@ -841,11 +893,11 @@ nsecs_t AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread)
size_t nonContig;
status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
- "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount);
+ "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount);
requested = &ClientProxy::kNonBlocking;
size_t avail = audioBuffer.frameCount + nonContig;
- ALOGV("obtainBuffer(%u) returned %u = %u + %u",
- mRemainingFrames, avail, audioBuffer.frameCount, nonContig);
+ ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d",
+ mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
if (err != NO_ERROR) {
if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) {
break;
@@ -872,8 +924,8 @@ nsecs_t AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread)
// Sanity check on returned size
if (ssize_t(readSize) < 0 || readSize > reqSize) {
- ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes",
- reqSize, (int) readSize);
+ ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
+ reqSize, ssize_t(readSize));
return NS_NEVER;
}
@@ -932,7 +984,7 @@ status_t AudioRecord::restoreRecord_l(const char *from)
status_t result;
// if the new IAudioRecord is created, openRecord_l() will modify the
- // following member variables: mAudioRecord, mCblkMemory and mCblk.
+ // following member variables: mAudioRecord, mCblkMemory, mCblk, mBufferMemory.
// It will also delete the strong references on previous IAudioRecord and IMemory
size_t position = mProxy->getPosition();
mNewPosition = position + mUpdatePeriod;
@@ -954,7 +1006,7 @@ status_t AudioRecord::restoreRecord_l(const char *from)
// =========================================================================
-void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who)
+void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
{
sp<AudioRecord> audioRecord = mAudioRecord.promote();
if (audioRecord != 0) {
@@ -966,7 +1018,8 @@ void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who)
// =========================================================================
AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver, bool bCanCallJava)
- : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL)
+ : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
+ mIgnoreNextPausedInt(false)
{
}
@@ -983,6 +1036,10 @@ bool AudioRecord::AudioRecordThread::threadLoop()
// caller will check for exitPending()
return true;
}
+ if (mIgnoreNextPausedInt) {
+ mIgnoreNextPausedInt = false;
+ mPausedInt = false;
+ }
if (mPausedInt) {
if (mPausedNs > 0) {
(void) mMyCond.waitRelative(mMyLock, mPausedNs);
@@ -993,7 +1050,7 @@ bool AudioRecord::AudioRecordThread::threadLoop()
return true;
}
}
- nsecs_t ns = mReceiver.processAudioBuffer(this);
+ nsecs_t ns = mReceiver.processAudioBuffer();
switch (ns) {
case 0:
return true;
@@ -1007,7 +1064,7 @@ bool AudioRecord::AudioRecordThread::threadLoop()
ns = 1000000000LL;
// fall through
default:
- LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns);
+ LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns);
pauseInternal(ns);
return true;
}
@@ -1017,12 +1074,7 @@ void AudioRecord::AudioRecordThread::requestExit()
{
// must be in this order to avoid a race condition
Thread::requestExit();
- AutoMutex _l(mMyLock);
- if (mPaused || mPausedInt) {
- mPaused = false;
- mPausedInt = false;
- mMyCond.signal();
- }
+ resume();
}
void AudioRecord::AudioRecordThread::pause()
@@ -1034,6 +1086,7 @@ void AudioRecord::AudioRecordThread::pause()
void AudioRecord::AudioRecordThread::resume()
{
AutoMutex _l(mMyLock);
+ mIgnoreNextPausedInt = true;
if (mPaused || mPausedInt) {
mPaused = false;
mPausedInt = false;
generated by cgit v1.1 at 2024-05-31 21:18:23 +0000