aaudio: fix SHARED MMAP mode in server plus other bugs
Fixed some buffer miscalculations, and some NPEs in the close() code.
Added debugging and some general cleanup.
Fixed data conversion.
Fixed start/pause/flush in server.
Added reference counting in server for endpoints.
Programs can now be ran more than once.
General code cleanup.
Reconnect with service if server dies.
Move stop() logic into server for better synchronization.
Add sleep to prevent race condition when closing an MMAP stream.
Bug: 33398120
Test: two write_sine_callback.cpp can be run simultaneously
Change-Id: Ibb006215a498868c222228d675ff961d7e0bf514
Signed-off-by: Phil Burk <philburk@google.com>
diff --git a/services/oboeservice/AAudioServiceStreamShared.cpp b/services/oboeservice/AAudioServiceStreamShared.cpp
index cd9336b..b5d9927 100644
--- a/services/oboeservice/AAudioServiceStreamShared.cpp
+++ b/services/oboeservice/AAudioServiceStreamShared.cpp
@@ -61,7 +61,7 @@
ALOGD("AAudioServiceStreamShared::open(), direction = %d", direction);
AAudioEndpointManager &mEndpointManager = AAudioEndpointManager::getInstance();
- mServiceEndpoint = mEndpointManager.findEndpoint(mAudioService, deviceId, direction);
+ mServiceEndpoint = mEndpointManager.openEndpoint(mAudioService, deviceId, direction);
ALOGD("AAudioServiceStreamShared::open(), mServiceEndPoint = %p", mServiceEndpoint);
if (mServiceEndpoint == nullptr) {
return AAUDIO_ERROR_UNAVAILABLE;
@@ -72,6 +72,7 @@
if (mAudioFormat == AAUDIO_FORMAT_UNSPECIFIED) {
mAudioFormat = AAUDIO_FORMAT_PCM_FLOAT;
} else if (mAudioFormat != AAUDIO_FORMAT_PCM_FLOAT) {
+ ALOGE("AAudioServiceStreamShared::open(), mAudioFormat = %d, need FLOAT", mAudioFormat);
return AAUDIO_ERROR_INVALID_FORMAT;
}
@@ -79,6 +80,8 @@
if (mSampleRate == AAUDIO_FORMAT_UNSPECIFIED) {
mSampleRate = mServiceEndpoint->getSampleRate();
} else if (mSampleRate != mServiceEndpoint->getSampleRate()) {
+ ALOGE("AAudioServiceStreamShared::open(), mAudioFormat = %d, need %d",
+ mSampleRate, mServiceEndpoint->getSampleRate());
return AAUDIO_ERROR_INVALID_RATE;
}
@@ -86,17 +89,22 @@
if (mSamplesPerFrame == AAUDIO_FORMAT_UNSPECIFIED) {
mSamplesPerFrame = mServiceEndpoint->getSamplesPerFrame();
} else if (mSamplesPerFrame != mServiceEndpoint->getSamplesPerFrame()) {
+ ALOGE("AAudioServiceStreamShared::open(), mSamplesPerFrame = %d, need %d",
+ mSamplesPerFrame, mServiceEndpoint->getSamplesPerFrame());
return AAUDIO_ERROR_OUT_OF_RANGE;
}
// Determine this stream's shared memory buffer capacity.
mFramesPerBurst = mServiceEndpoint->getFramesPerBurst();
int32_t minCapacityFrames = configurationInput.getBufferCapacity();
- int32_t numBursts = (minCapacityFrames + mFramesPerBurst - 1) / mFramesPerBurst;
- if (numBursts < MIN_BURSTS_PER_BUFFER) {
- numBursts = MIN_BURSTS_PER_BUFFER;
- } else if (numBursts > MAX_BURSTS_PER_BUFFER) {
- numBursts = MAX_BURSTS_PER_BUFFER;
+ int32_t numBursts = MAX_BURSTS_PER_BUFFER;
+ if (minCapacityFrames != AAUDIO_UNSPECIFIED) {
+ numBursts = (minCapacityFrames + mFramesPerBurst - 1) / mFramesPerBurst;
+ if (numBursts < MIN_BURSTS_PER_BUFFER) {
+ numBursts = MIN_BURSTS_PER_BUFFER;
+ } else if (numBursts > MAX_BURSTS_PER_BUFFER) {
+ numBursts = MAX_BURSTS_PER_BUFFER;
+ }
}
mCapacityInFrames = numBursts * mFramesPerBurst;
ALOGD("AAudioServiceStreamShared::open(), mCapacityInFrames = %d", mCapacityInFrames);
@@ -122,8 +130,12 @@
* An AAUDIO_SERVICE_EVENT_STARTED will be sent to the client when complete.
*/
aaudio_result_t AAudioServiceStreamShared::start() {
+ AAudioServiceEndpoint *endpoint = mServiceEndpoint;
+ if (endpoint == nullptr) {
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
// Add this stream to the mixer.
- aaudio_result_t result = mServiceEndpoint->startStream(this);
+ aaudio_result_t result = endpoint->startStream(this);
if (result != AAUDIO_OK) {
ALOGE("AAudioServiceStreamShared::start() mServiceEndpoint returned %d", result);
processError();
@@ -139,15 +151,31 @@
* An AAUDIO_SERVICE_EVENT_PAUSED will be sent to the client when complete.
*/
aaudio_result_t AAudioServiceStreamShared::pause() {
+ AAudioServiceEndpoint *endpoint = mServiceEndpoint;
+ if (endpoint == nullptr) {
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
// Add this stream to the mixer.
- aaudio_result_t result = mServiceEndpoint->stopStream(this);
+ aaudio_result_t result = endpoint->stopStream(this);
+ if (result != AAUDIO_OK) {
+ ALOGE("AAudioServiceStreamShared::pause() mServiceEndpoint returned %d", result);
+ processError();
+ }
+ return AAudioServiceStreamBase::pause();
+}
+
+aaudio_result_t AAudioServiceStreamShared::stop() {
+ AAudioServiceEndpoint *endpoint = mServiceEndpoint;
+ if (endpoint == nullptr) {
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+ // Add this stream to the mixer.
+ aaudio_result_t result = endpoint->stopStream(this);
if (result != AAUDIO_OK) {
ALOGE("AAudioServiceStreamShared::stop() mServiceEndpoint returned %d", result);
processError();
- } else {
- result = AAudioServiceStreamBase::start();
}
- return AAUDIO_OK;
+ return AAudioServiceStreamBase::stop();
}
/**
@@ -157,15 +185,21 @@
*/
aaudio_result_t AAudioServiceStreamShared::flush() {
// TODO make sure we are paused
- return AAUDIO_OK;
+ // TODO actually flush the data
+ return AAudioServiceStreamBase::flush() ;
}
aaudio_result_t AAudioServiceStreamShared::close() {
pause();
// TODO wait for pause() to synchronize
- mServiceEndpoint->unregisterStream(this);
- mServiceEndpoint->close();
- mServiceEndpoint = nullptr;
+ AAudioServiceEndpoint *endpoint = mServiceEndpoint;
+ if (endpoint != nullptr) {
+ endpoint->unregisterStream(this);
+
+ AAudioEndpointManager &mEndpointManager = AAudioEndpointManager::getInstance();
+ mEndpointManager.closeEndpoint(endpoint);
+ mServiceEndpoint = nullptr;
+ }
return AAudioServiceStreamBase::close();
}
@@ -189,10 +223,15 @@
mServiceEndpoint = nullptr;
}
+void AAudioServiceStreamShared::markTransferTime(int64_t nanoseconds) {
+ mMarkedPosition = mAudioDataQueue->getFifoBuffer()->getReadCounter();
+ mMarkedTime = nanoseconds;
+}
aaudio_result_t AAudioServiceStreamShared::getFreeRunningPosition(int64_t *positionFrames,
int64_t *timeNanos) {
- *positionFrames = mAudioDataQueue->getFifoBuffer()->getReadCounter();
- *timeNanos = AudioClock::getNanoseconds();
+ // TODO get these two numbers as an atomic pair
+ *positionFrames = mMarkedPosition;
+ *timeNanos = mMarkedTime;
return AAUDIO_OK;
}