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-rw-r--r--media/libmedia/AudioRecord.cpp72
1 files changed, 49 insertions, 23 deletions
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 07ca14f..100a914 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -112,7 +112,9 @@ AudioRecord::~AudioRecord()
mCblkMemory.clear();
mBufferMemory.clear();
IPCThreadState::self()->flushCommands();
- AudioSystem::releaseAudioSessionId(mSessionId, -1);
+ ALOGV("~AudioRecord, releasing session id %d",
+ mSessionId);
+ AudioSystem::releaseAudioSessionId(mSessionId, -1 /*pid*/);
}
}
@@ -159,8 +161,6 @@ status_t AudioRecord::set(
}
mTransfer = transferType;
- AutoMutex lock(mLock);
-
// invariant that mAudioRecord != 0 is true only after set() returns successfully
if (mAudioRecord != 0) {
ALOGE("Track already in use");
@@ -233,6 +233,7 @@ status_t AudioRecord::set(
if (cbf != NULL) {
mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava);
mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
+ // thread begins in paused state, and will not reference us until start()
}
// create the IAudioRecord
@@ -286,7 +287,6 @@ status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession)
status_t status = NO_ERROR;
if (!(flags & CBLK_INVALID)) {
- ALOGV("mAudioRecord->start()");
status = mAudioRecord->start(event, triggerSession);
if (status == DEAD_OBJECT) {
flags |= CBLK_INVALID;
@@ -352,6 +352,10 @@ status_t AudioRecord::setMarkerPosition(uint32_t marker)
mMarkerPosition = marker;
mMarkerReached = false;
+ sp<AudioRecordThread> t = mAudioRecordThread;
+ if (t != 0) {
+ t->wake();
+ }
return NO_ERROR;
}
@@ -378,6 +382,10 @@ status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
mNewPosition = mProxy->getPosition() + updatePeriod;
mUpdatePeriod = updatePeriod;
+ sp<AudioRecordThread> t = mAudioRecordThread;
+ if (t != 0) {
+ t->wake();
+ }
return NO_ERROR;
}
@@ -408,7 +416,7 @@ status_t AudioRecord::getPosition(uint32_t *position) 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());
+ return AudioSystem::getInputFramesLost(getInputPrivate());
}
// -------------------------------------------------------------------------
@@ -416,7 +424,6 @@ uint32_t AudioRecord::getInputFramesLost() const
// must be called with mLock held
status_t AudioRecord::openRecord_l(size_t epoch)
{
- status_t status;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
if (audioFlinger == 0) {
ALOGE("Could not get audioflinger");
@@ -431,12 +438,16 @@ status_t AudioRecord::openRecord_l(size_t epoch)
}
// Client can only express a preference for FAST. Server will perform additional tests.
- if ((mFlags & AUDIO_INPUT_FLAG_FAST) && !(
- // use case: callback transfer mode
- (mTransfer == TRANSFER_CALLBACK) &&
+ if ((mFlags & AUDIO_INPUT_FLAG_FAST) && !((
+ // either of these use cases:
+ // use case 1: callback transfer mode
+ (mTransfer == TRANSFER_CALLBACK) ||
+ // use case 2: obtain/release mode
+ (mTransfer == TRANSFER_OBTAIN)) &&
// matching sample rate
(mSampleRate == afSampleRate))) {
- ALOGW("AUDIO_INPUT_FLAG_FAST denied by client");
+ ALOGW("AUDIO_INPUT_FLAG_FAST denied by client; transfer %d, track %u Hz, primary %u Hz",
+ mTransfer, mSampleRate, afSampleRate);
// once denied, do not request again if IAudioRecord is re-created
mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
}
@@ -452,7 +463,8 @@ status_t AudioRecord::openRecord_l(size_t epoch)
}
audio_io_handle_t input;
- status = AudioSystem::getInputForAttr(&mAttributes, &input, (audio_session_t)mSessionId,
+ status_t status = AudioSystem::getInputForAttr(&mAttributes, &input,
+ (audio_session_t)mSessionId,
mSampleRate, mFormat, mChannelMask, mFlags);
if (status != NO_ERROR) {
@@ -684,9 +696,9 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *r
return status;
}
-void AudioRecord::releaseBuffer(Buffer* audioBuffer)
+void AudioRecord::releaseBuffer(const Buffer* audioBuffer)
{
- // all TRANSFER_* are valid
+ // FIXME add error checking on mode, by adding an internal version
size_t stepCount = audioBuffer->size / mFrameSize;
if (stepCount == 0) {
@@ -704,7 +716,7 @@ void AudioRecord::releaseBuffer(Buffer* audioBuffer)
// the server does not automatically disable recorder on overrun, so no need to restart
}
-audio_io_handle_t AudioRecord::getInput() const
+audio_io_handle_t AudioRecord::getInputPrivate() const
{
AutoMutex lock(mLock);
return mInput;
@@ -712,7 +724,7 @@ audio_io_handle_t AudioRecord::getInput() const
// -------------------------------------------------------------------------
-ssize_t AudioRecord::read(void* buffer, size_t userSize)
+ssize_t AudioRecord::read(void* buffer, size_t userSize, bool blocking)
{
if (mTransfer != TRANSFER_SYNC) {
return INVALID_OPERATION;
@@ -731,7 +743,8 @@ ssize_t AudioRecord::read(void* buffer, size_t userSize)
while (userSize >= mFrameSize) {
audioBuffer.frameCount = userSize / mFrameSize;
- status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever);
+ status_t err = obtainBuffer(&audioBuffer,
+ blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
if (err < 0) {
if (read > 0) {
break;
@@ -863,8 +876,11 @@ nsecs_t AudioRecord::processAudioBuffer()
if (!markerReached && position < markerPosition) {
minFrames = markerPosition - position;
}
- if (updatePeriod > 0 && updatePeriod < minFrames) {
- minFrames = updatePeriod;
+ if (updatePeriod > 0) {
+ uint32_t remaining = newPosition - position;
+ if (remaining < minFrames) {
+ minFrames = remaining;
+ }
}
// If > 0, poll periodically to recover from a stuck server. A good value is 2.
@@ -990,14 +1006,13 @@ status_t AudioRecord::restoreRecord_l(const char *from)
{
ALOGW("dead IAudioRecord, creating a new one from %s()", from);
++mSequence;
- status_t result;
// if the new IAudioRecord is created, openRecord_l() will modify the
// 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;
- result = openRecord_l(position);
+ status_t result = openRecord_l(position);
if (result == NO_ERROR) {
if (mActive) {
// callback thread or sync event hasn't changed
@@ -1069,8 +1084,8 @@ bool AudioRecord::AudioRecordThread::threadLoop()
case NS_NEVER:
return false;
case NS_WHENEVER:
- // FIXME increase poll interval, or make event-driven
- ns = 1000000000LL;
+ // Event driven: call wake() when callback notifications conditions change.
+ ns = INT64_MAX;
// fall through
default:
LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns);
@@ -1103,6 +1118,17 @@ void AudioRecord::AudioRecordThread::resume()
}
}
+void AudioRecord::AudioRecordThread::wake()
+{
+ AutoMutex _l(mMyLock);
+ if (!mPaused && mPausedInt && mPausedNs > 0) {
+ // audio record is active and internally paused with timeout.
+ mIgnoreNextPausedInt = true;
+ mPausedInt = false;
+ mMyCond.signal();
+ }
+}
+
void AudioRecord::AudioRecordThread::pauseInternal(nsecs_t ns)
{
AutoMutex _l(mMyLock);
@@ -1112,4 +1138,4 @@ void AudioRecord::AudioRecordThread::pauseInternal(nsecs_t ns)
// -------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
generated by cgit v1.1 at 2024-05-27 19:10:17 +0000