aaudio: cache framesPerBurst and capacity
These do not change after opening the stream so we can
simply return a stored value and avoid querying the device.
Test: atest AAudioTests
Change-Id: I7a66c12bd695fd732194ff0a14ac9c8d9cacf923
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 2688597..94f10e5 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -210,10 +210,10 @@
result = AAUDIO_ERROR_OUT_OF_RANGE;
goto error;
}
- mFramesPerBurst = framesPerBurst; // only save good value
+ setFramesPerBurst(framesPerBurst); // only save good value
mBufferCapacityInFrames = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames;
- if (mBufferCapacityInFrames < mFramesPerBurst
+ if (mBufferCapacityInFrames < getFramesPerBurst()
|| mBufferCapacityInFrames > MAX_BUFFER_CAPACITY_IN_FRAMES) {
ALOGE("%s - bufferCapacity out of range = %d", __func__, mBufferCapacityInFrames);
result = AAUDIO_ERROR_OUT_OF_RANGE;
@@ -238,7 +238,7 @@
}
if (mCallbackFrames == AAUDIO_UNSPECIFIED) {
- mCallbackFrames = mFramesPerBurst;
+ mCallbackFrames = getFramesPerBurst();
}
const int32_t callbackBufferSize = mCallbackFrames * getBytesPerFrame();
@@ -756,9 +756,9 @@
aaudio_result_t AudioStreamInternal::setBufferSize(int32_t requestedFrames) {
int32_t adjustedFrames = requestedFrames;
- const int32_t maximumSize = getBufferCapacity() - mFramesPerBurst;
+ const int32_t maximumSize = getBufferCapacity() - getFramesPerBurst();
// Minimum size should be a multiple number of bursts.
- const int32_t minimumSize = 1 * mFramesPerBurst;
+ const int32_t minimumSize = 1 * getFramesPerBurst();
// Clip to minimum size so that rounding up will work better.
adjustedFrames = std::max(minimumSize, adjustedFrames);
@@ -768,9 +768,9 @@
adjustedFrames = maximumSize;
} else {
// Round to the next highest burst size.
- int32_t numBursts = (adjustedFrames + mFramesPerBurst - 1) / mFramesPerBurst;
- adjustedFrames = numBursts * mFramesPerBurst;
- // Clip just in case maximumSize is not a multiple of mFramesPerBurst.
+ int32_t numBursts = (adjustedFrames + getFramesPerBurst() - 1) / getFramesPerBurst();
+ adjustedFrames = numBursts * getFramesPerBurst();
+ // Clip just in case maximumSize is not a multiple of getFramesPerBurst().
adjustedFrames = std::min(maximumSize, adjustedFrames);
}
@@ -805,10 +805,6 @@
return mBufferCapacityInFrames;
}
-int32_t AudioStreamInternal::getFramesPerBurst() const {
- return mFramesPerBurst;
-}
-
// This must be called under mStreamLock.
aaudio_result_t AudioStreamInternal::joinThread(void** returnArg) {
return AudioStream::joinThread(returnArg, calculateReasonableTimeout(getFramesPerBurst()));
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 162f098..d7024cf 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -64,8 +64,6 @@
int32_t getBufferCapacity() const override;
- int32_t getFramesPerBurst() const override;
-
int32_t getXRunCount() const override {
return mXRunCount;
}
@@ -159,7 +157,6 @@
aaudio_handle_t mServiceStreamHandle; // opaque handle returned from service
- int32_t mFramesPerBurst = MIN_FRAMES_PER_BURST; // frames per HAL transfer
int32_t mXRunCount = 0; // how many underrun events?
// Offset from underlying frame position.
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index 55fc986..5d311fc 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -149,7 +149,7 @@
// Calculate frame position based off of the readCounter because
// the writeCounter might have just advanced in the background,
// causing us to sleep until a later burst.
- int64_t nextPosition = mAudioEndpoint->getDataReadCounter() + mFramesPerBurst;
+ int64_t nextPosition = mAudioEndpoint->getDataReadCounter() + getFramesPerBurst();
wakeTime = mClockModel.convertPositionToLatestTime(nextPosition);
}
break;
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index e0bd9d8..e438477 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -202,11 +202,11 @@
}
virtual int32_t getBufferCapacity() const {
- return AAUDIO_ERROR_UNIMPLEMENTED;
+ return mBufferCapacity;
}
virtual int32_t getFramesPerBurst() const {
- return AAUDIO_ERROR_UNIMPLEMENTED;
+ return mFramesPerBurst;
}
virtual int32_t getXRunCount() const {
@@ -498,30 +498,32 @@
mSampleRate = sampleRate;
}
- /**
- * This should not be called after the open() call.
- */
+ // This should not be called after the open() call.
void setSamplesPerFrame(int32_t samplesPerFrame) {
mSamplesPerFrame = samplesPerFrame;
}
- /**
- * This should not be called after the open() call.
- */
+ // This should not be called after the open() call.
+ void setFramesPerBurst(int32_t framesPerBurst) {
+ mFramesPerBurst = framesPerBurst;
+ }
+
+ // This should not be called after the open() call.
+ void setBufferCapacity(int32_t bufferCapacity) {
+ mBufferCapacity = bufferCapacity;
+ }
+
+ // This should not be called after the open() call.
void setSharingMode(aaudio_sharing_mode_t sharingMode) {
mSharingMode = sharingMode;
}
- /**
- * This should not be called after the open() call.
- */
+ // This should not be called after the open() call.
void setFormat(audio_format_t format) {
mFormat = format;
}
- /**
- * This should not be called after the open() call.
- */
+ // This should not be called after the open() call.
void setDeviceFormat(audio_format_t format) {
mDeviceFormat = format;
}
@@ -536,6 +538,7 @@
mDeviceId = deviceId;
}
+ // This should not be called after the open() call.
void setSessionId(int32_t sessionId) {
mSessionId = sessionId;
}
@@ -623,6 +626,8 @@
audio_format_t mFormat = AUDIO_FORMAT_DEFAULT;
aaudio_stream_state_t mState = AAUDIO_STREAM_STATE_UNINITIALIZED;
aaudio_performance_mode_t mPerformanceMode = AAUDIO_PERFORMANCE_MODE_NONE;
+ int32_t mFramesPerBurst = 0;
+ int32_t mBufferCapacity = 0;
aaudio_usage_t mUsage = AAUDIO_UNSPECIFIED;
aaudio_content_type_t mContentType = AAUDIO_UNSPECIFIED;
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.h b/media/libaaudio/src/legacy/AudioStreamLegacy.h
index fefe6e0..88ef270 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.h
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.h
@@ -112,6 +112,18 @@
return mFramesRead.increment(frames);
}
+ /**
+ * Get the framesPerBurst from the underlying API.
+ * @return framesPerBurst
+ */
+ virtual int32_t getFramesPerBurstFromDevice() const = 0;
+
+ /**
+ * Get the bufferCapacity from the underlying API.
+ * @return bufferCapacity in frames
+ */
+ virtual int32_t getBufferCapacityFromDevice() const = 0;
+
// This is used for exact matching by MediaMetrics. So do not change it.
// MediaMetricsConstants.h: AMEDIAMETRICS_PROP_CALLERNAME_VALUE_AAUDIO
static constexpr char kCallerName[] = "aaudio";
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index a8ae0fb..d46ef56 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
@@ -210,9 +210,9 @@
// Get the actual values from the AudioRecord.
setSamplesPerFrame(mAudioRecord->channelCount());
-
- int32_t actualSampleRate = mAudioRecord->getSampleRate();
- setSampleRate(actualSampleRate);
+ setSampleRate(mAudioRecord->getSampleRate());
+ setBufferCapacity(getBufferCapacityFromDevice());
+ setFramesPerBurst(getFramesPerBurstFromDevice());
// We may need to pass the data through a block size adapter to guarantee constant size.
if (mCallbackBufferSize != AAUDIO_UNSPECIFIED) {
@@ -488,7 +488,7 @@
return getBufferCapacity(); // TODO implement in AudioRecord?
}
-int32_t AudioStreamRecord::getBufferCapacity() const
+int32_t AudioStreamRecord::getBufferCapacityFromDevice() const
{
return static_cast<int32_t>(mAudioRecord->frameCount());
}
@@ -498,8 +498,7 @@
return 0; // TODO implement when AudioRecord supports it
}
-int32_t AudioStreamRecord::getFramesPerBurst() const
-{
+int32_t AudioStreamRecord::getFramesPerBurstFromDevice() const {
return static_cast<int32_t>(mAudioRecord->getNotificationPeriodInFrames());
}
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.h b/media/libaaudio/src/legacy/AudioStreamRecord.h
index e4ef1c0..ad8dfe4 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.h
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.h
@@ -56,14 +56,10 @@
int32_t getBufferSize() const override;
- int32_t getBufferCapacity() const override;
-
int32_t getXRunCount() const override;
int64_t getFramesWritten() override;
- int32_t getFramesPerBurst() const override;
-
aaudio_result_t updateStateMachine() override;
aaudio_direction_t getDirection() const override {
@@ -79,6 +75,11 @@
const void * maybeConvertDeviceData(const void *audioData, int32_t numFrames) override;
+protected:
+
+ int32_t getFramesPerBurstFromDevice() const override;
+ int32_t getBufferCapacityFromDevice() const override;
+
private:
android::sp<android::AudioRecord> mAudioRecord;
// adapts between variable sized blocks and fixed size blocks
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 4ba08fd..307904e 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -192,9 +192,9 @@
setSamplesPerFrame(mAudioTrack->channelCount());
setFormat(mAudioTrack->format());
setDeviceFormat(mAudioTrack->format());
-
- int32_t actualSampleRate = mAudioTrack->getSampleRate();
- setSampleRate(actualSampleRate);
+ setSampleRate(mAudioTrack->getSampleRate());
+ setBufferCapacity(getBufferCapacityFromDevice());
+ setFramesPerBurst(getFramesPerBurstFromDevice());
// We may need to pass the data through a block size adapter to guarantee constant size.
if (mCallbackBufferSize != AAUDIO_UNSPECIFIED) {
@@ -217,9 +217,6 @@
: (aaudio_session_id_t) mAudioTrack->getSessionId();
setSessionId(actualSessionId);
- mInitialBufferCapacity = getBufferCapacity();
- mInitialFramesPerBurst = getFramesPerBurst();
-
mAudioTrack->addAudioDeviceCallback(this);
// Update performance mode based on the actual stream flags.
@@ -284,8 +281,8 @@
|| mAudioTrack->format() != getFormat()
|| mAudioTrack->getSampleRate() != getSampleRate()
|| mAudioTrack->getRoutedDeviceId() != getDeviceId()
- || getBufferCapacity() != mInitialBufferCapacity
- || getFramesPerBurst() != mInitialFramesPerBurst) {
+ || getBufferCapacityFromDevice() != getBufferCapacity()
+ || getFramesPerBurstFromDevice() != getFramesPerBurst()) {
processCallbackCommon(AAUDIO_CALLBACK_OPERATION_DISCONNECTED, info);
}
break;
@@ -474,7 +471,7 @@
return static_cast<int32_t>(mAudioTrack->getBufferSizeInFrames());
}
-int32_t AudioStreamTrack::getBufferCapacity() const
+int32_t AudioStreamTrack::getBufferCapacityFromDevice() const
{
return static_cast<int32_t>(mAudioTrack->frameCount());
}
@@ -484,8 +481,7 @@
return static_cast<int32_t>(mAudioTrack->getUnderrunCount());
}
-int32_t AudioStreamTrack::getFramesPerBurst() const
-{
+int32_t AudioStreamTrack::getFramesPerBurstFromDevice() const {
return static_cast<int32_t>(mAudioTrack->getNotificationPeriodInFrames());
}
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.h b/media/libaaudio/src/legacy/AudioStreamTrack.h
index 6334f66..5a8fb39 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.h
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.h
@@ -69,8 +69,6 @@
aaudio_result_t setBufferSize(int32_t requestedFrames) override;
int32_t getBufferSize() const override;
- int32_t getBufferCapacity() const override;
- int32_t getFramesPerBurst()const override;
int32_t getXRunCount() const override;
int64_t getFramesRead() override;
@@ -96,6 +94,11 @@
const android::media::VolumeShaper::Operation& operation) override;
#endif
+protected:
+
+ int32_t getFramesPerBurstFromDevice() const override;
+ int32_t getBufferCapacityFromDevice() const override;
+
private:
android::sp<android::AudioTrack> mAudioTrack;
@@ -105,10 +108,6 @@
// TODO add 64-bit position reporting to AudioTrack and use it.
aaudio_wrapping_frames_t mPositionWhenPausing = 0;
-
- // initial AudioTrack frame count and notification period
- int32_t mInitialBufferCapacity = 0;
- int32_t mInitialFramesPerBurst = 0;
};
} /* namespace aaudio */