Add standby mode for aaudio service stream.
When the stream is stopped, everything will remain open but just stop
writing data. But this will keep the DSP running and using power
until the stream is closed.
To resolve this issue, the solution is to put the stream into standby
mode so that the HAL can release the corresponding resource.
When the HAL releases the resource, the shared file descriptor will also
be released. In that case, when the stream is restarted, AAudioService
needs to recreate shared buffer and the client needs to replace the new
shared buffer.
Test: atest AAudioTests
Test: test_steal_exclusive
Bug: 201000721
Bug: 196394385
Bug: 167345722
Bug: 208619472
Change-Id: Ib4f98e7aee72c2e56acd7f2f0ac378a94ec26241
diff --git a/media/libaaudio/include/aaudio/AAudioTesting.h b/media/libaaudio/include/aaudio/AAudioTesting.h
index 02ec411..0f2d7a2 100644
--- a/media/libaaudio/include/aaudio/AAudioTesting.h
+++ b/media/libaaudio/include/aaudio/AAudioTesting.h
@@ -49,6 +49,12 @@
};
typedef int32_t aaudio_policy_t;
+// Internal error codes. Only used by the framework.
+enum {
+ AAUDIO_INTERNAL_ERROR_BASE = -1000,
+ AAUDIO_ERROR_STANDBY,
+};
+
/**
* Control whether AAudioStreamBuilder_openStream() will use the new MMAP data path
* or the older "Legacy" data path.
diff --git a/media/libaaudio/src/binding/AAudioBinderAdapter.cpp b/media/libaaudio/src/binding/AAudioBinderAdapter.cpp
index 6e3a1c8..42d81ca 100644
--- a/media/libaaudio/src/binding/AAudioBinderAdapter.cpp
+++ b/media/libaaudio/src/binding/AAudioBinderAdapter.cpp
@@ -124,4 +124,16 @@
return result;
}
+aaudio_result_t AAudioBinderAdapter::exitStandby(aaudio_handle_t streamHandle,
+ AudioEndpointParcelable &endpointOut) {
+ aaudio_result_t result;
+ Endpoint endpoint;
+ Status status = mDelegate->exitStandby(streamHandle, &endpoint, &result);
+ if (!status.isOk()) {
+ result = AAudioConvert_androidToAAudioResult(statusTFromBinderStatus(status));
+ }
+ endpointOut = std::move(endpoint);
+ return result;
+}
+
} // namespace aaudio
diff --git a/media/libaaudio/src/binding/AAudioBinderAdapter.h b/media/libaaudio/src/binding/AAudioBinderAdapter.h
index 5e9ab57..d170783 100644
--- a/media/libaaudio/src/binding/AAudioBinderAdapter.h
+++ b/media/libaaudio/src/binding/AAudioBinderAdapter.h
@@ -57,6 +57,9 @@
aaudio_result_t unregisterAudioThread(aaudio_handle_t streamHandle,
pid_t clientThreadId) override;
+ aaudio_result_t exitStandby(aaudio_handle_t streamHandle,
+ AudioEndpointParcelable &parcelable) override;
+
private:
IAAudioService* const mDelegate;
};
diff --git a/media/libaaudio/src/binding/AAudioBinderClient.cpp b/media/libaaudio/src/binding/AAudioBinderClient.cpp
index 135bac3..8e5facc 100644
--- a/media/libaaudio/src/binding/AAudioBinderClient.cpp
+++ b/media/libaaudio/src/binding/AAudioBinderClient.cpp
@@ -201,3 +201,11 @@
return service->unregisterAudioThread(streamHandle, clientThreadId);
}
+
+aaudio_result_t AAudioBinderClient::exitStandby(aaudio_handle_t streamHandle,
+ AudioEndpointParcelable &endpointOut) {
+ std::shared_ptr<AAudioServiceInterface> service = getAAudioService();
+ if (service.get() == nullptr) return AAUDIO_ERROR_NO_SERVICE;
+
+ return service->exitStandby(streamHandle, endpointOut);
+}
diff --git a/media/libaaudio/src/binding/AAudioBinderClient.h b/media/libaaudio/src/binding/AAudioBinderClient.h
index 557ced5..0968f4c 100644
--- a/media/libaaudio/src/binding/AAudioBinderClient.h
+++ b/media/libaaudio/src/binding/AAudioBinderClient.h
@@ -108,6 +108,9 @@
return AAUDIO_ERROR_UNAVAILABLE;
}
+ aaudio_result_t exitStandby(aaudio_handle_t streamHandle,
+ AudioEndpointParcelable &endpointOut) override;
+
void onStreamChange(aaudio_handle_t /*handle*/, int32_t /*opcode*/, int32_t /*value*/) {
// TODO This is just a stub so we can have a client Binder to pass to the service.
// TODO Implemented in a later CL.
diff --git a/media/libaaudio/src/binding/AAudioServiceInterface.h b/media/libaaudio/src/binding/AAudioServiceInterface.h
index bf94774..e901767 100644
--- a/media/libaaudio/src/binding/AAudioServiceInterface.h
+++ b/media/libaaudio/src/binding/AAudioServiceInterface.h
@@ -95,6 +95,16 @@
virtual aaudio_result_t stopClient(aaudio_handle_t streamHandle,
audio_port_handle_t clientHandle) = 0;
+
+ /**
+ * Exit the standby mode.
+ *
+ * @param streamHandle the stream handle
+ * @param parcelable contains new data queue information
+ * @return the result of the execution
+ */
+ virtual aaudio_result_t exitStandby(aaudio_handle_t streamHandle,
+ AudioEndpointParcelable &parcelable) = 0;
};
} /* namespace aaudio */
diff --git a/media/libaaudio/src/binding/AudioEndpointParcelable.cpp b/media/libaaudio/src/binding/AudioEndpointParcelable.cpp
index dea3e4a..b1262df 100644
--- a/media/libaaudio/src/binding/AudioEndpointParcelable.cpp
+++ b/media/libaaudio/src/binding/AudioEndpointParcelable.cpp
@@ -79,6 +79,22 @@
return index;
}
+void AudioEndpointParcelable::closeDataFileDescriptor() {
+ const int32_t curDataMemoryIndex = mDownDataQueueParcelable.getSharedMemoryIndex();
+ mSharedMemories[curDataMemoryIndex].closeAndReleaseFd();
+}
+
+void AudioEndpointParcelable::updateDataFileDescriptor(
+ AudioEndpointParcelable* endpointParcelable) {
+ const int32_t curDataMemoryIndex = mDownDataQueueParcelable.getSharedMemoryIndex();
+ const int32_t newDataMemoryIndex =
+ endpointParcelable->mDownDataQueueParcelable.getSharedMemoryIndex();
+ mSharedMemories[curDataMemoryIndex].close();
+ mSharedMemories[curDataMemoryIndex].setup(
+ endpointParcelable->mSharedMemories[newDataMemoryIndex]);
+ mDownDataQueueParcelable.updateMemory(endpointParcelable->mDownDataQueueParcelable);
+}
+
aaudio_result_t AudioEndpointParcelable::resolve(EndpointDescriptor *descriptor) {
aaudio_result_t result = mUpMessageQueueParcelable.resolve(mSharedMemories,
&descriptor->upMessageQueueDescriptor);
@@ -92,6 +108,10 @@
return result;
}
+aaudio_result_t AudioEndpointParcelable::resolveDataQueue(RingBufferDescriptor *descriptor) {
+ return mDownDataQueueParcelable.resolve(mSharedMemories, descriptor);
+}
+
aaudio_result_t AudioEndpointParcelable::close() {
int err = 0;
for (int i = 0; i < mNumSharedMemories; i++) {
diff --git a/media/libaaudio/src/binding/AudioEndpointParcelable.h b/media/libaaudio/src/binding/AudioEndpointParcelable.h
index 544aa92..5d2c38f 100644
--- a/media/libaaudio/src/binding/AudioEndpointParcelable.h
+++ b/media/libaaudio/src/binding/AudioEndpointParcelable.h
@@ -52,7 +52,20 @@
*/
int32_t addFileDescriptor(const android::base::unique_fd& fd, int32_t sizeInBytes);
+ /**
+ * Close current data file descriptor. The duplicated file descriptor will be close.
+ */
+ void closeDataFileDescriptor();
+
+ /**
+ * Update current data file descriptor with given endpoint parcelable.
+ * @param endpointParcelable an endpoint parcelable that contains new data file
+ * descriptor information
+ */
+ void updateDataFileDescriptor(AudioEndpointParcelable* endpointParcelable);
+
aaudio_result_t resolve(EndpointDescriptor *descriptor);
+ aaudio_result_t resolveDataQueue(RingBufferDescriptor *descriptor);
aaudio_result_t close();
diff --git a/media/libaaudio/src/binding/RingBufferParcelable.cpp b/media/libaaudio/src/binding/RingBufferParcelable.cpp
index fa7ca72..3bc51d0 100644
--- a/media/libaaudio/src/binding/RingBufferParcelable.cpp
+++ b/media/libaaudio/src/binding/RingBufferParcelable.cpp
@@ -33,6 +33,7 @@
: mReadCounterParcelable(parcelable.readCounterParcelable),
mWriteCounterParcelable(parcelable.writeCounterParcelable),
mDataParcelable(parcelable.dataParcelable),
+ mSharedMemoryIndex(parcelable.sharedMemoryIndex),
mBytesPerFrame(parcelable.bytesPerFrame),
mFramesPerBurst(parcelable.framesPerBurst),
mCapacityInFrames(parcelable.capacityInFrames),
@@ -45,6 +46,7 @@
result.readCounterParcelable = mReadCounterParcelable.parcelable();
result.writeCounterParcelable = mWriteCounterParcelable.parcelable();
result.dataParcelable = mDataParcelable.parcelable();
+ result.sharedMemoryIndex = mSharedMemoryIndex;
result.bytesPerFrame = mBytesPerFrame;
result.framesPerBurst = mFramesPerBurst;
result.capacityInFrames = mCapacityInFrames;
@@ -60,6 +62,7 @@
int32_t readCounterOffset,
int32_t writeCounterOffset,
int32_t counterSizeBytes) {
+ mSharedMemoryIndex = sharedMemoryIndex;
mReadCounterParcelable.setup(sharedMemoryIndex, readCounterOffset, counterSizeBytes);
mWriteCounterParcelable.setup(sharedMemoryIndex, writeCounterOffset, counterSizeBytes);
mDataParcelable.setup(sharedMemoryIndex, dataMemoryOffset, dataSizeInBytes);
@@ -68,12 +71,13 @@
void RingBufferParcelable::setupMemory(int32_t sharedMemoryIndex,
int32_t dataMemoryOffset,
int32_t dataSizeInBytes) {
+ mSharedMemoryIndex = sharedMemoryIndex;
mReadCounterParcelable.setup(sharedMemoryIndex, 0, 0);
mWriteCounterParcelable.setup(sharedMemoryIndex, 0, 0);
mDataParcelable.setup(sharedMemoryIndex, dataMemoryOffset, dataSizeInBytes);
}
-int32_t RingBufferParcelable::getBytesPerFrame() {
+int32_t RingBufferParcelable::getBytesPerFrame() const {
return mBytesPerFrame;
}
@@ -81,7 +85,7 @@
mBytesPerFrame = bytesPerFrame;
}
-int32_t RingBufferParcelable::getFramesPerBurst() {
+int32_t RingBufferParcelable::getFramesPerBurst() const {
return mFramesPerBurst;
}
@@ -89,7 +93,7 @@
mFramesPerBurst = framesPerBurst;
}
-int32_t RingBufferParcelable::getCapacityInFrames() {
+int32_t RingBufferParcelable::getCapacityInFrames() const {
return mCapacityInFrames;
}
@@ -124,6 +128,14 @@
return AAUDIO_OK;
}
+void RingBufferParcelable::updateMemory(const RingBufferParcelable& parcelable) {
+ setupMemory(mSharedMemoryIndex, 0,
+ parcelable.getCapacityInFrames() * parcelable.getBytesPerFrame());
+ setBytesPerFrame(parcelable.getBytesPerFrame());
+ setFramesPerBurst(parcelable.getFramesPerBurst());
+ setCapacityInFrames(parcelable.getCapacityInFrames());
+}
+
aaudio_result_t RingBufferParcelable::validate() const {
if (mCapacityInFrames < 0 || mCapacityInFrames >= 32 * 1024) {
ALOGE("invalid mCapacityInFrames = %d", mCapacityInFrames);
diff --git a/media/libaaudio/src/binding/RingBufferParcelable.h b/media/libaaudio/src/binding/RingBufferParcelable.h
index 2508cea..29d0d86 100644
--- a/media/libaaudio/src/binding/RingBufferParcelable.h
+++ b/media/libaaudio/src/binding/RingBufferParcelable.h
@@ -46,15 +46,15 @@
int32_t dataMemoryOffset,
int32_t dataSizeInBytes);
- int32_t getBytesPerFrame();
+ int32_t getBytesPerFrame() const;
void setBytesPerFrame(int32_t bytesPerFrame);
- int32_t getFramesPerBurst();
+ int32_t getFramesPerBurst() const;
void setFramesPerBurst(int32_t framesPerBurst);
- int32_t getCapacityInFrames();
+ int32_t getCapacityInFrames() const;
void setCapacityInFrames(int32_t capacityInFrames);
@@ -62,6 +62,12 @@
aaudio_result_t resolve(SharedMemoryParcelable *memoryParcels, RingBufferDescriptor *descriptor);
+ void updateMemory(const RingBufferParcelable& parcelable);
+
+ int32_t getSharedMemoryIndex() const {
+ return mSharedMemoryIndex;
+ }
+
void dump();
// Extract a parcelable representation of this object.
@@ -71,6 +77,7 @@
SharedRegionParcelable mReadCounterParcelable;
SharedRegionParcelable mWriteCounterParcelable;
SharedRegionParcelable mDataParcelable;
+ int32_t mSharedMemoryIndex = -1;
int32_t mBytesPerFrame = 0; // index is in frames
int32_t mFramesPerBurst = 0; // for ISOCHRONOUS queues
int32_t mCapacityInFrames = 0; // zero if unused
diff --git a/media/libaaudio/src/binding/SharedMemoryParcelable.cpp b/media/libaaudio/src/binding/SharedMemoryParcelable.cpp
index 3a49655..741aefc 100644
--- a/media/libaaudio/src/binding/SharedMemoryParcelable.cpp
+++ b/media/libaaudio/src/binding/SharedMemoryParcelable.cpp
@@ -64,6 +64,10 @@
mSizeInBytes = sizeInBytes;
}
+void SharedMemoryParcelable::setup(const SharedMemoryParcelable &sharedMemoryParcelable) {
+ setup(sharedMemoryParcelable.mFd, sharedMemoryParcelable.mSizeInBytes);
+}
+
aaudio_result_t SharedMemoryParcelable::close() {
if (mResolvedAddress != MMAP_UNRESOLVED_ADDRESS) {
int err = munmap(mResolvedAddress, mSizeInBytes);
@@ -76,6 +80,14 @@
return AAUDIO_OK;
}
+aaudio_result_t SharedMemoryParcelable::closeAndReleaseFd() {
+ aaudio_result_t result = close();
+ if (result == AAUDIO_OK) {
+ mFd.reset();
+ }
+ return result;
+}
+
aaudio_result_t SharedMemoryParcelable::resolveSharedMemory(const unique_fd& fd) {
mResolvedAddress = (uint8_t *) mmap(nullptr, mSizeInBytes, PROT_READ | PROT_WRITE,
MAP_SHARED, fd.get(), 0);
diff --git a/media/libaaudio/src/binding/SharedMemoryParcelable.h b/media/libaaudio/src/binding/SharedMemoryParcelable.h
index 1f2c335..7762fef 100644
--- a/media/libaaudio/src/binding/SharedMemoryParcelable.h
+++ b/media/libaaudio/src/binding/SharedMemoryParcelable.h
@@ -52,12 +52,16 @@
*/
void setup(const android::base::unique_fd& fd, int32_t sizeInBytes);
+ void setup(const SharedMemoryParcelable& sharedMemoryParcelable);
+
// mmap() shared memory
aaudio_result_t resolve(int32_t offsetInBytes, int32_t sizeInBytes, void **regionAddressPtr);
// munmap() any mapped memory
aaudio_result_t close();
+ aaudio_result_t closeAndReleaseFd();
+
int32_t getSizeInBytes();
void dump();
diff --git a/media/libaaudio/src/binding/aidl/aaudio/IAAudioService.aidl b/media/libaaudio/src/binding/aidl/aaudio/IAAudioService.aidl
index 44d2211..485c2e2 100644
--- a/media/libaaudio/src/binding/aidl/aaudio/IAAudioService.aidl
+++ b/media/libaaudio/src/binding/aidl/aaudio/IAAudioService.aidl
@@ -78,4 +78,6 @@
int unregisterAudioThread(int streamHandle,
int clientThreadId);
+
+ int exitStandby(int streamHandle, out Endpoint endpoint);
}
diff --git a/media/libaaudio/src/binding/aidl/aaudio/RingBuffer.aidl b/media/libaaudio/src/binding/aidl/aaudio/RingBuffer.aidl
index a58b33a..dd64493 100644
--- a/media/libaaudio/src/binding/aidl/aaudio/RingBuffer.aidl
+++ b/media/libaaudio/src/binding/aidl/aaudio/RingBuffer.aidl
@@ -26,4 +26,5 @@
int framesPerBurst; // for ISOCHRONOUS queues
int capacityInFrames; // zero if unused
int /* RingbufferFlags */ flags; // = RingbufferFlags::NONE;
+ int sharedMemoryIndex;
}
\ No newline at end of file
diff --git a/media/libaaudio/src/client/AudioEndpoint.cpp b/media/libaaudio/src/client/AudioEndpoint.cpp
index 24888de..e780f4f 100644
--- a/media/libaaudio/src/client/AudioEndpoint.cpp
+++ b/media/libaaudio/src/client/AudioEndpoint.cpp
@@ -139,38 +139,49 @@
);
// ============================ data queue =============================
- descriptor = &pEndpointDescriptor->dataQueueDescriptor;
- ALOGV("configure() data framesPerBurst = %d", descriptor->framesPerBurst);
+ result = configureDataQueue(pEndpointDescriptor->dataQueueDescriptor, direction);
+
+ return result;
+}
+
+aaudio_result_t AudioEndpoint::configureDataQueue(const RingBufferDescriptor& descriptor,
+ aaudio_direction_t direction) {
+ aaudio_result_t result = AudioEndpoint_validateQueueDescriptor("data", &descriptor);
+ if (result != AAUDIO_OK) {
+ return result;
+ }
+
+ ALOGV("configure() data framesPerBurst = %d", descriptor.framesPerBurst);
ALOGV("configure() data readCounterAddress = %p",
- descriptor->readCounterAddress);
+ descriptor.readCounterAddress);
// An example of free running is when the other side is read or written by hardware DMA
// or a DSP. It does not update its counter so we have to update it.
int64_t *remoteCounter = (direction == AAUDIO_DIRECTION_OUTPUT)
- ? descriptor->readCounterAddress // read by other side
- : descriptor->writeCounterAddress; // written by other side
+ ? descriptor.readCounterAddress // read by other side
+ : descriptor.writeCounterAddress; // written by other side
mFreeRunning = (remoteCounter == nullptr);
ALOGV("configure() mFreeRunning = %d", mFreeRunning ? 1 : 0);
- int64_t *readCounterAddress = (descriptor->readCounterAddress == nullptr)
+ int64_t *readCounterAddress = (descriptor.readCounterAddress == nullptr)
? &mDataReadCounter
- : descriptor->readCounterAddress;
- int64_t *writeCounterAddress = (descriptor->writeCounterAddress == nullptr)
+ : descriptor.readCounterAddress;
+ int64_t *writeCounterAddress = (descriptor.writeCounterAddress == nullptr)
? &mDataWriteCounter
- : descriptor->writeCounterAddress;
+ : descriptor.writeCounterAddress;
// Clear buffer to avoid an initial glitch on some devices.
- size_t bufferSizeBytes = descriptor->capacityInFrames * descriptor->bytesPerFrame;
- memset(descriptor->dataAddress, 0, bufferSizeBytes);
+ size_t bufferSizeBytes = descriptor.capacityInFrames * descriptor.bytesPerFrame;
+ memset(descriptor.dataAddress, 0, bufferSizeBytes);
mDataQueue = std::make_unique<FifoBufferIndirect>(
- descriptor->bytesPerFrame,
- descriptor->capacityInFrames,
+ descriptor.bytesPerFrame,
+ descriptor.capacityInFrames,
readCounterAddress,
writeCounterAddress,
- descriptor->dataAddress
+ descriptor.dataAddress
);
- uint32_t threshold = descriptor->capacityInFrames / 2;
+ uint32_t threshold = descriptor.capacityInFrames / 2;
mDataQueue->setThreshold(threshold);
return result;
}
@@ -181,47 +192,66 @@
}
int32_t AudioEndpoint::getEmptyFramesAvailable(WrappingBuffer *wrappingBuffer) {
- return mDataQueue->getEmptyRoomAvailable(wrappingBuffer);
+ return mDataQueue == nullptr ? 0 : mDataQueue->getEmptyRoomAvailable(wrappingBuffer);
}
int32_t AudioEndpoint::getEmptyFramesAvailable() {
- return mDataQueue->getEmptyFramesAvailable();
+ return mDataQueue == nullptr ? 0 : mDataQueue->getEmptyFramesAvailable();
}
int32_t AudioEndpoint::getFullFramesAvailable(WrappingBuffer *wrappingBuffer) {
- return mDataQueue->getFullDataAvailable(wrappingBuffer);
+ return mDataQueue == nullptr ? 0 : mDataQueue->getFullDataAvailable(wrappingBuffer);
}
int32_t AudioEndpoint::getFullFramesAvailable() {
- return mDataQueue->getFullFramesAvailable();
+ return mDataQueue == nullptr ? 0 : mDataQueue->getFullFramesAvailable();
+}
+
+android::fifo_frames_t AudioEndpoint::read(void *buffer, android::fifo_frames_t numFrames) {
+ return mDataQueue == nullptr ? 0 : mDataQueue->read(buffer, numFrames);
+}
+
+android::fifo_frames_t AudioEndpoint::write(void *buffer, android::fifo_frames_t numFrames) {
+ return mDataQueue == nullptr ? 0 : mDataQueue->write(buffer, numFrames);
}
void AudioEndpoint::advanceWriteIndex(int32_t deltaFrames) {
- mDataQueue->advanceWriteIndex(deltaFrames);
+ if (mDataQueue != nullptr) {
+ mDataQueue->advanceWriteIndex(deltaFrames);
+ }
}
void AudioEndpoint::advanceReadIndex(int32_t deltaFrames) {
- mDataQueue->advanceReadIndex(deltaFrames);
+ if (mDataQueue != nullptr) {
+ mDataQueue->advanceReadIndex(deltaFrames);
+ }
}
void AudioEndpoint::setDataReadCounter(fifo_counter_t framesRead) {
- mDataQueue->setReadCounter(framesRead);
+ if (mDataQueue != nullptr) {
+ mDataQueue->setReadCounter(framesRead);
+ }
}
fifo_counter_t AudioEndpoint::getDataReadCounter() const {
- return mDataQueue->getReadCounter();
+ return mDataQueue == nullptr ? 0 : mDataQueue->getReadCounter();
}
void AudioEndpoint::setDataWriteCounter(fifo_counter_t framesRead) {
- mDataQueue->setWriteCounter(framesRead);
+ if (mDataQueue != nullptr) {
+ mDataQueue->setWriteCounter(framesRead);
+ }
}
fifo_counter_t AudioEndpoint::getDataWriteCounter() const {
- return mDataQueue->getWriteCounter();
+ return mDataQueue == nullptr ? 0 : mDataQueue->getWriteCounter();
}
int32_t AudioEndpoint::setBufferSizeInFrames(int32_t requestedFrames,
int32_t *actualFrames) {
+ if (mDataQueue == nullptr) {
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
if (requestedFrames < ENDPOINT_DATA_QUEUE_SIZE_MIN) {
requestedFrames = ENDPOINT_DATA_QUEUE_SIZE_MIN;
}
@@ -231,11 +261,11 @@
}
int32_t AudioEndpoint::getBufferSizeInFrames() const {
- return mDataQueue->getThreshold();
+ return mDataQueue == nullptr ? 0 : mDataQueue->getThreshold();
}
int32_t AudioEndpoint::getBufferCapacityInFrames() const {
- return (int32_t)mDataQueue->getBufferCapacityInFrames();
+ return mDataQueue == nullptr ? 0 : (int32_t)mDataQueue->getBufferCapacityInFrames();
}
void AudioEndpoint::dump() const {
@@ -244,5 +274,7 @@
}
void AudioEndpoint::eraseDataMemory() {
- mDataQueue->eraseMemory();
+ if (mDataQueue != nullptr) {
+ mDataQueue->eraseMemory();
+ }
}
diff --git a/media/libaaudio/src/client/AudioEndpoint.h b/media/libaaudio/src/client/AudioEndpoint.h
index b3dbc20..01dd05a 100644
--- a/media/libaaudio/src/client/AudioEndpoint.h
+++ b/media/libaaudio/src/client/AudioEndpoint.h
@@ -17,6 +17,8 @@
#ifndef ANDROID_AAUDIO_AUDIO_ENDPOINT_H
#define ANDROID_AAUDIO_AUDIO_ENDPOINT_H
+#include <mutex>
+
#include <aaudio/AAudio.h>
#include "binding/AAudioServiceMessage.h"
@@ -42,6 +44,9 @@
aaudio_result_t configure(const EndpointDescriptor *pEndpointDescriptor,
aaudio_direction_t direction);
+ aaudio_result_t configureDataQueue(const RingBufferDescriptor &descriptor,
+ aaudio_direction_t direction);
+
/**
* Read from a command passed up from the Server.
* @return 1 if command received, 0 for no command, or negative error.
@@ -56,6 +61,10 @@
int32_t getFullFramesAvailable();
+ android::fifo_frames_t read(void* buffer, android::fifo_frames_t numFrames);
+
+ android::fifo_frames_t write(void* buffer, android::fifo_frames_t numFrames);
+
void advanceReadIndex(int32_t deltaFrames);
void advanceWriteIndex(int32_t deltaFrames);
@@ -85,11 +94,21 @@
int32_t getBufferCapacityInFrames() const;
+ void setThreshold(int32_t frames) {
+ mDataQueue->setThreshold(frames);
+ }
+
+ int32_t getThreshold() {
+ return mDataQueue->getThreshold();
+ }
+
/**
* Write zeros to the data queue memory.
*/
void eraseDataMemory();
+ void freeDataQueue();
+
void dump() const;
private:
@@ -98,6 +117,8 @@
bool mFreeRunning{false};
android::fifo_counter_t mDataReadCounter{0}; // only used if free-running
android::fifo_counter_t mDataWriteCounter{0}; // only used if free-running
+
+ std::mutex mDataQueueLock;
};
} // namespace aaudio
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index afdc2ac..9f0564f 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -81,8 +81,6 @@
aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) {
aaudio_result_t result = AAUDIO_OK;
- int32_t framesPerBurst;
- int32_t framesPerHardwareBurst;
AAudioStreamRequest request;
AAudioStreamConfiguration configurationOutput;
@@ -97,9 +95,6 @@
return result;
}
- const int32_t burstMinMicros = android::AudioSystem::getAAudioHardwareBurstMinUsec();
- int32_t burstMicros = 0;
-
const audio_format_t requestedFormat = getFormat();
// We have to do volume scaling. So we prefer FLOAT format.
if (requestedFormat == AUDIO_FORMAT_DEFAULT) {
@@ -215,12 +210,28 @@
goto error;
}
- framesPerHardwareBurst = mEndpointDescriptor.dataQueueDescriptor.framesPerBurst;
+ if ((result = configureDataInformation(builder.getFramesPerDataCallback())) != AAUDIO_OK) {
+ goto error;
+ }
+
+ setState(AAUDIO_STREAM_STATE_OPEN);
+
+ return result;
+
+error:
+ safeReleaseClose();
+ return result;
+}
+
+aaudio_result_t AudioStreamInternal::configureDataInformation(int32_t callbackFrames) {
+ int32_t framesPerHardwareBurst = mEndpointDescriptor.dataQueueDescriptor.framesPerBurst;
// Scale up the burst size to meet the minimum equivalent in microseconds.
// This is to avoid waking the CPU too often when the HW burst is very small
// or at high sample rates.
- framesPerBurst = framesPerHardwareBurst;
+ int32_t framesPerBurst = framesPerHardwareBurst;
+ int32_t burstMicros = 0;
+ const int32_t burstMinMicros = android::AudioSystem::getAAudioHardwareBurstMinUsec();
do {
if (burstMicros > 0) { // skip first loop
framesPerBurst *= 2;
@@ -233,8 +244,7 @@
// Validate final burst size.
if (framesPerBurst < MIN_FRAMES_PER_BURST || framesPerBurst > MAX_FRAMES_PER_BURST) {
ALOGE("%s - framesPerBurst out of range = %d", __func__, framesPerBurst);
- result = AAUDIO_ERROR_OUT_OF_RANGE;
- goto error;
+ return AAUDIO_ERROR_OUT_OF_RANGE;
}
setFramesPerBurst(framesPerBurst); // only save good value
@@ -242,26 +252,21 @@
if (mBufferCapacityInFrames < getFramesPerBurst()
|| mBufferCapacityInFrames > MAX_BUFFER_CAPACITY_IN_FRAMES) {
ALOGE("%s - bufferCapacity out of range = %d", __func__, mBufferCapacityInFrames);
- result = AAUDIO_ERROR_OUT_OF_RANGE;
- goto error;
+ return AAUDIO_ERROR_OUT_OF_RANGE;
}
mClockModel.setSampleRate(getSampleRate());
mClockModel.setFramesPerBurst(framesPerHardwareBurst);
if (isDataCallbackSet()) {
- mCallbackFrames = builder.getFramesPerDataCallback();
+ mCallbackFrames = callbackFrames;
if (mCallbackFrames > getBufferCapacity() / 2) {
ALOGW("%s - framesPerCallback too big = %d, capacity = %d",
__func__, mCallbackFrames, getBufferCapacity());
- result = AAUDIO_ERROR_OUT_OF_RANGE;
- goto error;
-
+ return AAUDIO_ERROR_OUT_OF_RANGE;
} else if (mCallbackFrames < 0) {
ALOGW("%s - framesPerCallback negative", __func__);
- result = AAUDIO_ERROR_OUT_OF_RANGE;
- goto error;
-
+ return AAUDIO_ERROR_OUT_OF_RANGE;
}
if (mCallbackFrames == AAUDIO_UNSPECIFIED) {
mCallbackFrames = getFramesPerBurst();
@@ -302,14 +307,7 @@
}
setBufferSize(mBufferCapacityInFrames / 2); // Default buffer size to match Q
-
- setState(AAUDIO_STREAM_STATE_OPEN);
-
- return result;
-
-error:
- safeReleaseClose();
- return result;
+ return AAUDIO_OK;
}
// This must be called under mStreamLock.
@@ -357,6 +355,60 @@
}
}
+aaudio_result_t AudioStreamInternal::exitStandby_l() {
+ AudioEndpointParcelable endpointParcelable;
+ // The stream is in standby mode, copy all available data and then close the duplicated
+ // shared file descriptor so that it won't cause issue when the HAL try to reallocate new
+ // shared file descriptor when exiting from standby.
+ // Cache current read counter, which will be reset to new read and write counter
+ // when the new data queue and endpoint are reconfigured.
+ const android::fifo_counter_t readCounter = mAudioEndpoint->getDataReadCounter();
+ // Cache the buffer size which may be from client.
+ const int32_t previousBufferSize = mBufferSizeInFrames;
+ // Copy all available data from current data queue.
+ uint8_t buffer[getBufferCapacity() * getBytesPerFrame()];
+ android::fifo_frames_t fullFramesAvailable =
+ mAudioEndpoint->read(buffer, getBufferCapacity());
+ mEndPointParcelable.closeDataFileDescriptor();
+ aaudio_result_t result = mServiceInterface.exitStandby(
+ mServiceStreamHandle, endpointParcelable);
+ if (result != AAUDIO_OK) {
+ ALOGE("Failed to exit standby, error=%d", result);
+ goto exit;
+ }
+ // Reconstruct data queue descriptor using new shared file descriptor.
+ mEndPointParcelable.updateDataFileDescriptor(&endpointParcelable);
+ result = mEndPointParcelable.resolveDataQueue(&mEndpointDescriptor.dataQueueDescriptor);
+ if (result != AAUDIO_OK) {
+ ALOGE("Failed to resolve data queue after exiting standby, error=%d", result);
+ goto exit;
+ }
+ // Reconfigure audio endpoint with new data queue descriptor.
+ mAudioEndpoint->configureDataQueue(
+ mEndpointDescriptor.dataQueueDescriptor, getDirection());
+ // Set read and write counters with previous read counter, the later write action
+ // will make the counter at the correct place.
+ mAudioEndpoint->setDataReadCounter(readCounter);
+ mAudioEndpoint->setDataWriteCounter(readCounter);
+ result = configureDataInformation(mCallbackFrames);
+ if (result != AAUDIO_OK) {
+ ALOGE("Failed to configure data information after exiting standby, error=%d", result);
+ goto exit;
+ }
+ // Write data from previous data buffer to new endpoint.
+ if (android::fifo_frames_t framesWritten =
+ mAudioEndpoint->write(buffer, fullFramesAvailable);
+ framesWritten != fullFramesAvailable) {
+ ALOGW("Some data lost after exiting standby, frames written: %d, "
+ "frames to write: %d", framesWritten, fullFramesAvailable);
+ }
+ // Reset previous buffer size as it may be requested by the client.
+ setBufferSize(previousBufferSize);
+
+exit:
+ return result;
+}
+
/*
* It normally takes about 20-30 msec to start a stream on the server.
* But the first time can take as much as 200-300 msec. The HW
@@ -393,8 +445,15 @@
prepareBuffersForStart(); // tell subclasses to get ready
aaudio_result_t result = mServiceInterface.startStream(mServiceStreamHandle);
- if (result == AAUDIO_ERROR_INVALID_HANDLE) {
- ALOGD("%s() INVALID_HANDLE, stream was probably stolen", __func__);
+ if (result == AAUDIO_ERROR_STANDBY) {
+ // The stream is at standby mode. Need to exit standby before starting the stream.
+ result = exitStandby_l();
+ if (result == AAUDIO_OK) {
+ result = mServiceInterface.startStream(mServiceStreamHandle);
+ }
+ }
+ if (result != AAUDIO_OK) {
+ ALOGD("%s() error = %d, stream was probably stolen", __func__, result);
// Stealing was added in R. Coerce result to improve backward compatibility.
result = AAUDIO_ERROR_DISCONNECTED;
setState(AAUDIO_STREAM_STATE_DISCONNECTED);
@@ -414,6 +473,7 @@
result = createThread_l(periodNanos, aaudio_callback_thread_proc, this);
}
if (result != AAUDIO_OK) {
+ // TODO(b/214607638): Do we want to roll back to original state or keep as disconnected?
setState(originalState);
}
return result;
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index eab1382..2367572 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -184,9 +184,14 @@
aaudio_result_t writeNowWithConversion(const void *buffer,
int32_t numFrames);
+ // Exit the stream from standby, will reconstruct data path.
+ aaudio_result_t exitStandby_l() REQUIRES(mStreamLock);
+
// Adjust timing model based on timestamp from service.
void processTimestamp(uint64_t position, int64_t time);
+ aaudio_result_t configureDataInformation(int32_t callbackFrames);
+
// Thread on other side of FIFO will have wakeup jitter.
// By delaying slightly we can avoid waking up before other side is ready.
const int32_t mWakeupDelayNanos; // delay past typical wakeup jitter
diff --git a/media/libaaudio/tests/test_steal_exclusive.cpp b/media/libaaudio/tests/test_steal_exclusive.cpp
index 5cb005c..ca4f3d6 100644
--- a/media/libaaudio/tests/test_steal_exclusive.cpp
+++ b/media/libaaudio/tests/test_steal_exclusive.cpp
@@ -114,7 +114,7 @@
mCloseEnabled = enabled;
}
- void restartStream() {
+ aaudio_result_t restartStream() {
int retriesLeft = mMaxRetries;
aaudio_result_t result;
do {
@@ -130,6 +130,7 @@
mName.c_str(),
AAudio_convertResultToText(result));
} while (retriesLeft-- > 0 && result != AAUDIO_OK);
+ return result;
}
aaudio_data_callback_result_t onAudioReady(
@@ -238,6 +239,12 @@
return AAudioStream_requestStart(mStream);
}
+ aaudio_result_t pause() {
+ std::lock_guard<std::mutex> lock(mLock);
+ if (mStream == nullptr) return 0;
+ return AAudioStream_requestPause(mStream);
+ }
+
aaudio_result_t stop() {
std::lock_guard<std::mutex> lock(mLock);
if (mStream == nullptr) return 0;
@@ -326,9 +333,10 @@
}
static void s_usage() {
- printf("test_steal_exclusive [-i] [-r{maxRetries}] [-d{delay}] -s -c{flag}\n");
+ printf("test_steal_exclusive [-i] [-r{maxRetries}] [-d{delay}] [-p{pausedTime}]-s -c{flag}\n");
printf(" -i direction INPUT, otherwise OUTPUT\n");
printf(" -d Delay open by milliseconds, default = 0\n");
+ printf(" -p Pause first stream then sleep for msec before opening second streams, default = 0\n");
printf(" -r max Retries in the error callback, default = 1\n");
printf(" -s try to open in SHARED mode\n");
printf(" -c enable or disabling Closing of the stream with 0/1, default = 1\n");
@@ -344,6 +352,7 @@
int openDelayMillis = 0;
bool closeEnabled = true;
aaudio_sharing_mode_t requestedSharingMode = AAUDIO_SHARING_MODE_EXCLUSIVE;
+ int pausedMillis = 0;
// Make printf print immediately so that debug info is not stuck
// in a buffer if we hang or crash.
@@ -366,6 +375,9 @@
case 'i':
direction = AAUDIO_DIRECTION_INPUT;
break;
+ case 'p':
+ pausedMillis = atoi(&arg[2]);
+ break;
case 'r':
maxRetries = atoi(&arg[2]);
break;
@@ -428,6 +440,12 @@
}
}
+ if (pausedMillis > 0) {
+ printf("Pausing the VICTIM for %d millis before starting THIEF -----\n", pausedMillis);
+ victim.pause();
+ usleep(pausedMillis * 1000);
+ }
+
printf("Trying to start the THIEF stream, which may steal the VICTIM MMAP resource -----\n");
result = thief.openAudioStream(direction, requestedSharingMode);
if (result != AAUDIO_OK) {
@@ -443,6 +461,25 @@
errorCount++;
}
+ if (pausedMillis > 0) {
+ result = victim.start();
+ printf("Restarting VICTIM, AAudioStream_requestStart(VICTIM) returned %d "
+ ">>>>>>>>>>>>>>>>>>>>>>\n", result);
+ if (result == AAUDIO_ERROR_DISCONNECTED) {
+ // The stream is disconnected due to thief steal the resource
+ printf("VICTIM was disconnected while hanging as the THIEF "
+ "stole the resource >>>>>>>>>>>>>>>>>>>>>>\n");
+ result = victim.restartStream();
+ printf("Restarting VICTIM, AAudioStream_requestStart(VICTIM) returned %d "
+ ">>>>>>>>>>>>>>>>>>>>>>\n", result);
+ if (result != AAUDIO_OK) {
+ errorCount++;
+ }
+ } else {
+ errorCount++;
+ }
+ }
+
// Give stream time to advance.
usleep(SLEEP_DURATION_MSEC * 1000);
diff --git a/services/oboeservice/AAudioService.cpp b/services/oboeservice/AAudioService.cpp
index 40a664e..2679b2e 100644
--- a/services/oboeservice/AAudioService.cpp
+++ b/services/oboeservice/AAudioService.cpp
@@ -280,6 +280,22 @@
AIDL_RETURN(serviceStream->unregisterAudioThread(clientThreadId));
}
+Status AAudioService::exitStandby(int32_t streamHandle, Endpoint* endpoint, int32_t *_aidl_return) {
+ static_assert(std::is_same_v<aaudio_result_t, std::decay_t<typeof(*_aidl_return)>>);
+
+ sp<AAudioServiceStreamBase> serviceStream = convertHandleToServiceStream(streamHandle);
+ if (serviceStream.get() == nullptr) {
+ ALOGE("getStreamDescription(), illegal stream handle = 0x%0x", streamHandle);
+ AIDL_RETURN(AAUDIO_ERROR_INVALID_HANDLE);
+ }
+ AudioEndpointParcelable endpointParcelable;
+ aaudio_result_t result = serviceStream->exitStandby(&endpointParcelable);
+ if (result == AAUDIO_OK) {
+ *endpoint = std::move(endpointParcelable).parcelable();
+ }
+ AIDL_RETURN(result);
+}
+
bool AAudioService::isCallerInService() {
pid_t clientPid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mAudioClient.attributionSource.pid));
uid_t clientUid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(mAudioClient.attributionSource.uid));
diff --git a/services/oboeservice/AAudioService.h b/services/oboeservice/AAudioService.h
index 7c1b796..0a111fb 100644
--- a/services/oboeservice/AAudioService.h
+++ b/services/oboeservice/AAudioService.h
@@ -82,6 +82,9 @@
binder::Status unregisterAudioThread(int32_t streamHandle, int32_t clientThreadId,
int32_t* _aidl_return) override;
+ binder::Status exitStandby(int32_t streamHandle, ::aaudio::Endpoint* endpoint,
+ int32_t* _aidl_return) override;
+
aaudio_result_t startClient(aaudio::aaudio_handle_t streamHandle,
const android::AudioClient& client,
const audio_attributes_t *attr,
diff --git a/services/oboeservice/AAudioServiceEndpoint.h b/services/oboeservice/AAudioServiceEndpoint.h
index a7f63d3..92004c5 100644
--- a/services/oboeservice/AAudioServiceEndpoint.h
+++ b/services/oboeservice/AAudioServiceEndpoint.h
@@ -77,6 +77,16 @@
return AAUDIO_ERROR_UNAVAILABLE;
}
+ virtual aaudio_result_t standby() {
+ ALOGD("AAudioServiceEndpoint::standby() AAUDIO_ERROR_UNAVAILABLE");
+ return AAUDIO_ERROR_UNAVAILABLE;
+ }
+
+ virtual aaudio_result_t exitStandby(AudioEndpointParcelable* parcelable) {
+ ALOGD("AAudioServiceEndpoint::exitStandby() AAUDIO_ERROR_UNAVAILABLE");
+ return AAUDIO_ERROR_UNAVAILABLE;
+ }
+
/**
* @param positionFrames
* @param timeNanos
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.cpp b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
index 046b84b..a266d5b 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.cpp
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
@@ -186,37 +186,8 @@
ALOGD("%s() deviceId = %d, sessionId = %d", __func__, getDeviceId(), getSessionId());
// Create MMAP/NOIRQ buffer.
- int32_t minSizeFrames = getBufferCapacity();
- if (minSizeFrames <= 0) { // zero will get rejected
- minSizeFrames = AAUDIO_BUFFER_CAPACITY_MIN;
- }
- status = mMmapStream->createMmapBuffer(minSizeFrames, &mMmapBufferinfo);
- bool isBufferShareable = mMmapBufferinfo.flags & AUDIO_MMAP_APPLICATION_SHAREABLE;
- if (status != OK) {
- ALOGE("%s() - createMmapBuffer() failed with status %d %s",
- __func__, status, strerror(-status));
- result = AAUDIO_ERROR_UNAVAILABLE;
+ if (createMmapBuffer(&mAudioDataFileDescriptor) != AAUDIO_OK) {
goto error;
- } else {
- ALOGD("%s() createMmapBuffer() buffer_size = %d fr, burst_size %d fr"
- ", Sharable FD: %s",
- __func__,
- mMmapBufferinfo.buffer_size_frames,
- mMmapBufferinfo.burst_size_frames,
- isBufferShareable ? "Yes" : "No");
- }
-
- setBufferCapacity(mMmapBufferinfo.buffer_size_frames);
- if (!isBufferShareable) {
- // Exclusive mode can only be used by the service because the FD cannot be shared.
- int32_t audioServiceUid =
- VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
- if ((mMmapClient.attributionSource.uid != audioServiceUid) &&
- getSharingMode() == AAUDIO_SHARING_MODE_EXCLUSIVE) {
- ALOGW("%s() - exclusive FD cannot be used by client", __func__);
- result = AAUDIO_ERROR_UNAVAILABLE;
- goto error;
- }
}
// Get information about the stream and pass it back to the caller.
@@ -224,21 +195,6 @@
config.channel_mask, getDirection() == AAUDIO_DIRECTION_INPUT,
AAudio_isChannelIndexMask(config.channel_mask)));
- // AAudio creates a copy of this FD and retains ownership of the copy.
- // Assume that AudioFlinger will close the original shared_memory_fd.
- mAudioDataFileDescriptor.reset(dup(mMmapBufferinfo.shared_memory_fd));
- if (mAudioDataFileDescriptor.get() == -1) {
- ALOGE("%s() - could not dup shared_memory_fd", __func__);
- result = AAUDIO_ERROR_INTERNAL;
- goto error;
- }
- // Call to HAL to make sure the transport FD was able to be closed by binder.
- // This is a tricky workaround for a problem in Binder.
- // TODO:[b/192048842] When that problem is fixed we may be able to remove or change this code.
- struct audio_mmap_position position;
- mMmapStream->getMmapPosition(&position);
-
- mFramesPerBurst = mMmapBufferinfo.burst_size_frames;
setFormat(config.format);
setSampleRate(config.sample_rate);
@@ -323,6 +279,32 @@
return result;
}
+aaudio_result_t AAudioServiceEndpointMMAP::standby() {
+ if (mMmapStream == nullptr) {
+ return AAUDIO_ERROR_NULL;
+ }
+ aaudio_result_t result = AAudioConvert_androidToAAudioResult(mMmapStream->standby());
+ return result;
+}
+
+aaudio_result_t AAudioServiceEndpointMMAP::exitStandby(AudioEndpointParcelable* parcelable) {
+ if (mMmapStream == nullptr) {
+ return AAUDIO_ERROR_NULL;
+ }
+ mAudioDataFileDescriptor.reset();
+ aaudio_result_t result = createMmapBuffer(&mAudioDataFileDescriptor);
+ if (result == AAUDIO_OK) {
+ int32_t bytesPerFrame = calculateBytesPerFrame();
+ int32_t capacityInBytes = getBufferCapacity() * bytesPerFrame;
+ int fdIndex = parcelable->addFileDescriptor(mAudioDataFileDescriptor, capacityInBytes);
+ parcelable->mDownDataQueueParcelable.setupMemory(fdIndex, 0, capacityInBytes);
+ parcelable->mDownDataQueueParcelable.setBytesPerFrame(bytesPerFrame);
+ parcelable->mDownDataQueueParcelable.setFramesPerBurst(mFramesPerBurst);
+ parcelable->mDownDataQueueParcelable.setCapacityInFrames(getBufferCapacity());
+ }
+ return result;
+}
+
// Get free-running DSP or DMA hardware position from the HAL.
aaudio_result_t AAudioServiceEndpointMMAP::getFreeRunningPosition(int64_t *positionFrames,
int64_t *timeNanos) {
@@ -502,3 +484,57 @@
*timeNanos = tempTimeNanos;
return mHalExternalPositionStatus;
}
+
+aaudio_result_t AAudioServiceEndpointMMAP::createMmapBuffer(
+ android::base::unique_fd* fileDescriptor)
+{
+ memset(&mMmapBufferinfo, 0, sizeof(struct audio_mmap_buffer_info));
+ int32_t minSizeFrames = getBufferCapacity();
+ if (minSizeFrames <= 0) { // zero will get rejected
+ minSizeFrames = AAUDIO_BUFFER_CAPACITY_MIN;
+ }
+ status_t status = mMmapStream->createMmapBuffer(minSizeFrames, &mMmapBufferinfo);
+ bool isBufferShareable = mMmapBufferinfo.flags & AUDIO_MMAP_APPLICATION_SHAREABLE;
+ if (status != OK) {
+ ALOGE("%s() - createMmapBuffer() failed with status %d %s",
+ __func__, status, strerror(-status));
+ return AAUDIO_ERROR_UNAVAILABLE;
+ } else {
+ ALOGD("%s() createMmapBuffer() buffer_size = %d fr, burst_size %d fr"
+ ", Sharable FD: %s",
+ __func__,
+ mMmapBufferinfo.buffer_size_frames,
+ mMmapBufferinfo.burst_size_frames,
+ isBufferShareable ? "Yes" : "No");
+ }
+
+ setBufferCapacity(mMmapBufferinfo.buffer_size_frames);
+ if (!isBufferShareable) {
+ // Exclusive mode can only be used by the service because the FD cannot be shared.
+ int32_t audioServiceUid =
+ VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
+ if ((mMmapClient.attributionSource.uid != audioServiceUid) &&
+ getSharingMode() == AAUDIO_SHARING_MODE_EXCLUSIVE) {
+ ALOGW("%s() - exclusive FD cannot be used by client", __func__);
+ return AAUDIO_ERROR_UNAVAILABLE;
+ }
+ }
+
+ // AAudio creates a copy of this FD and retains ownership of the copy.
+ // Assume that AudioFlinger will close the original shared_memory_fd.
+ fileDescriptor->reset(dup(mMmapBufferinfo.shared_memory_fd));
+ if (fileDescriptor->get() == -1) {
+ ALOGE("%s() - could not dup shared_memory_fd", __func__);
+ return AAUDIO_ERROR_INTERNAL;
+ }
+
+ // Call to HAL to make sure the transport FD was able to be closed by binder.
+ // This is a tricky workaround for a problem in Binder.
+ // TODO:[b/192048842] When that problem is fixed we may be able to remove or change this code.
+ struct audio_mmap_position position;
+ mMmapStream->getMmapPosition(&position);
+
+ mFramesPerBurst = mMmapBufferinfo.burst_size_frames;
+
+ return AAUDIO_OK;
+}
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.h b/services/oboeservice/AAudioServiceEndpointMMAP.h
index 6314e5e..3e7f2c7 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.h
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.h
@@ -64,6 +64,10 @@
aaudio_result_t stopClient(audio_port_handle_t clientHandle) override;
+ aaudio_result_t standby() override;
+
+ aaudio_result_t exitStandby(AudioEndpointParcelable* parcelable) override;
+
aaudio_result_t getFreeRunningPosition(int64_t *positionFrames, int64_t *timeNanos) override;
aaudio_result_t getTimestamp(int64_t *positionFrames, int64_t *timeNanos) override;
@@ -91,6 +95,8 @@
aaudio_result_t openWithFormat(audio_format_t audioFormat);
+ aaudio_result_t createMmapBuffer(android::base::unique_fd* fileDescriptor);
+
MonotonicCounter mFramesTransferred;
// Interface to the AudioFlinger MMAP support.
diff --git a/services/oboeservice/AAudioServiceStreamBase.cpp b/services/oboeservice/AAudioServiceStreamBase.cpp
index 8b5ccaa..9f48f80 100644
--- a/services/oboeservice/AAudioServiceStreamBase.cpp
+++ b/services/oboeservice/AAudioServiceStreamBase.cpp
@@ -41,6 +41,8 @@
using content::AttributionSourceState;
static const int64_t TIMEOUT_NANOS = 3LL * 1000 * 1000 * 1000;
+// If the stream is idle for more than `IDLE_TIMEOUT_NANOS`, the stream will be put into standby.
+static const int64_t IDLE_TIMEOUT_NANOS = 3e9;
/**
* Base class for streams in the service.
@@ -247,6 +249,12 @@
return AAUDIO_ERROR_INVALID_STATE;
}
+ if (mStandby) {
+ ALOGW("%s() the stream is standby, return ERROR_STANDBY, "
+ "expecting the client call exitStandby before start", __func__);
+ return AAUDIO_ERROR_STANDBY;
+ }
+
mediametrics::Defer defer([&] {
mediametrics::LogItem(mMetricsId)
.set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_START)
@@ -394,6 +402,7 @@
android::sp<AAudioServiceStreamBase> holdStream(this);
TimestampScheduler timestampScheduler;
int64_t nextTime;
+ int64_t standbyTime = AudioClock::getNanoseconds() + IDLE_TIMEOUT_NANOS;
// Balance the incStrong from when the thread was launched.
holdStream->decStrong(nullptr);
@@ -405,8 +414,8 @@
while (mThreadEnabled.load()) {
loopCount++;
int64_t timeoutNanos = -1;
- if (isRunning()) {
- timeoutNanos = nextTime - AudioClock::getNanoseconds();
+ if (isRunning() || (isIdle_l() && !isStandby_l())) {
+ timeoutNanos = (isRunning() ? nextTime : standbyTime) - AudioClock::getNanoseconds();
timeoutNanos = std::max<int64_t>(0, timeoutNanos);
}
@@ -425,6 +434,9 @@
nextTime = timestampScheduler.nextAbsoluteTime();
}
}
+ if (isIdle_l() && AudioClock::getNanoseconds() >= standbyTime) {
+ standby_l();
+ }
if (command != nullptr) {
std::scoped_lock<std::mutex> _commandLock(command->lock);
@@ -437,9 +449,11 @@
break;
case PAUSE:
command->result = pause_l();
+ standbyTime = AudioClock::getNanoseconds() + IDLE_TIMEOUT_NANOS;
break;
case STOP:
command->result = stop_l();
+ standbyTime = AudioClock::getNanoseconds() + IDLE_TIMEOUT_NANOS;
break;
case FLUSH:
command->result = flush_l();
@@ -474,6 +488,12 @@
: getDescription_l(param->mParcelable);
}
break;
+ case EXIT_STANDBY: {
+ ExitStandbyParam *param = (ExitStandbyParam *) command->parameter.get();
+ command->result = param == nullptr ? AAUDIO_ERROR_ILLEGAL_ARGUMENT
+ : exitStandby_l(param->mParcelable);
+ standbyTime = AudioClock::getNanoseconds() + IDLE_TIMEOUT_NANOS;
+ } break;
default:
ALOGE("Invalid command op code: %d", command->operationCode);
break;
@@ -673,6 +693,15 @@
return getAudioDataDescription_l(parcelable);
}
+aaudio_result_t AAudioServiceStreamBase::exitStandby(AudioEndpointParcelable *parcelable) {
+ auto command = std::make_shared<AAudioCommand>(
+ EXIT_STANDBY,
+ std::make_shared<ExitStandbyParam>(parcelable),
+ true /*waitForReply*/,
+ TIMEOUT_NANOS);
+ return mCommandQueue.sendCommand(command);
+}
+
void AAudioServiceStreamBase::onVolumeChanged(float volume) {
sendServiceEvent(AAUDIO_SERVICE_EVENT_VOLUME, volume);
}
diff --git a/services/oboeservice/AAudioServiceStreamBase.h b/services/oboeservice/AAudioServiceStreamBase.h
index dddd69f..b2ba725 100644
--- a/services/oboeservice/AAudioServiceStreamBase.h
+++ b/services/oboeservice/AAudioServiceStreamBase.h
@@ -116,6 +116,11 @@
*/
aaudio_result_t flush() EXCLUDES(mLock);
+ /**
+ * Exit standby mode. The MMAP buffer will be reallocated.
+ */
+ aaudio_result_t exitStandby(AudioEndpointParcelable *parcelable) EXCLUDES(mLock);
+
virtual aaudio_result_t startClient(const android::AudioClient& client,
const audio_attributes_t *attr __unused,
audio_port_handle_t *clientHandle __unused) {
@@ -314,6 +319,33 @@
mDisconnected = flag;
}
+ virtual aaudio_result_t standby_l() REQUIRES(mLock) {
+ return AAUDIO_ERROR_UNAVAILABLE;
+ }
+ class ExitStandbyParam : public AAudioCommandParam {
+ public:
+ ExitStandbyParam(AudioEndpointParcelable* parcelable)
+ : AAudioCommandParam(), mParcelable(parcelable) { }
+ ~ExitStandbyParam() = default;
+
+ AudioEndpointParcelable* mParcelable;
+ };
+ virtual aaudio_result_t exitStandby_l(
+ AudioEndpointParcelable* parcelable __unused) REQUIRES(mLock) {
+ return AAUDIO_ERROR_UNAVAILABLE;
+ }
+ bool isStandby_l() const REQUIRES(mLock) {
+ return mStandby;
+ }
+ void setStandby_l(bool standby) REQUIRES(mLock) {
+ mStandby = standby;
+ }
+
+ bool isIdle_l() const REQUIRES(mLock) {
+ return mState == AAUDIO_STREAM_STATE_OPEN || mState == AAUDIO_STREAM_STATE_PAUSED
+ || mState == AAUDIO_STREAM_STATE_STOPPED;
+ }
+
pid_t mRegisteredClientThread = ILLEGAL_THREAD_ID;
std::mutex mUpMessageQueueLock;
@@ -329,6 +361,7 @@
REGISTER_AUDIO_THREAD,
UNREGISTER_AUDIO_THREAD,
GET_DESCRIPTION,
+ EXIT_STANDBY,
};
AAudioThread mCommandThread;
std::atomic<bool> mThreadEnabled{false};
@@ -391,6 +424,8 @@
bool mDisconnected GUARDED_BY(mLock) {false};
+ bool mStandby GUARDED_BY(mLock) = false;
+
protected:
// Locking order is important.
// Acquire mLock before acquiring AAudioServiceEndpoint::mLockStreams
diff --git a/services/oboeservice/AAudioServiceStreamMMAP.cpp b/services/oboeservice/AAudioServiceStreamMMAP.cpp
index ffc16ac..ec9b2e2 100644
--- a/services/oboeservice/AAudioServiceStreamMMAP.cpp
+++ b/services/oboeservice/AAudioServiceStreamMMAP.cpp
@@ -117,6 +117,35 @@
return result;
}
+aaudio_result_t AAudioServiceStreamMMAP::standby_l() {
+ sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+ if (endpoint == nullptr) {
+ ALOGE("%s() has no endpoint", __func__);
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+ aaudio_result_t result = endpoint->standby();
+ if (result == AAUDIO_OK) {
+ setStandby_l(true);
+ }
+ return result;
+}
+
+aaudio_result_t AAudioServiceStreamMMAP::exitStandby_l(AudioEndpointParcelable* parcelable) {
+ sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+ if (endpoint == nullptr) {
+ ALOGE("%s() has no endpoint", __func__);
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+ aaudio_result_t result = endpoint->exitStandby(parcelable);
+ if (result == AAUDIO_OK) {
+ setStandby_l(false);
+ } else {
+ ALOGE("%s failed, result %d, disconnecting stream.", __func__, result);
+ disconnect_l();
+ }
+ return result;
+}
+
aaudio_result_t AAudioServiceStreamMMAP::startClient(const android::AudioClient& client,
const audio_attributes_t *attr,
audio_port_handle_t *clientHandle) {
diff --git a/services/oboeservice/AAudioServiceStreamMMAP.h b/services/oboeservice/AAudioServiceStreamMMAP.h
index 28da120..cd8c91e 100644
--- a/services/oboeservice/AAudioServiceStreamMMAP.h
+++ b/services/oboeservice/AAudioServiceStreamMMAP.h
@@ -71,6 +71,10 @@
aaudio_result_t stop_l() REQUIRES(mLock) override;
+ aaudio_result_t standby_l() REQUIRES(mLock) override;
+
+ aaudio_result_t exitStandby_l(AudioEndpointParcelable* parcelable) REQUIRES(mLock) override;
+
aaudio_result_t getAudioDataDescription_l(
AudioEndpointParcelable* parcelable) REQUIRES(mLock) override;
diff --git a/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp b/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp
index 17e8d36..5e48955 100644
--- a/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp
+++ b/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp
@@ -180,6 +180,11 @@
return AAUDIO_ERROR_UNAVAILABLE;
}
+ aaudio_result_t exitStandby(aaudio_handle_t streamHandle UNUSED_PARAM,
+ AudioEndpointParcelable &parcelable UNUSED_PARAM) override {
+ return AAUDIO_ERROR_UNAVAILABLE;
+ }
+
void onStreamChange(aaudio_handle_t handle, int32_t opcode, int32_t value) {}
int getDeathCount() { return mDeathCount; }