Merge changes from topic 'fix-omx-dependencies' into oc-dr1-dev
* changes:
Fix OMX dependencies.
Use vendor variant of libstagefright_foundation.
diff --git a/media/libaaudio/examples/input_monitor/Android.mk b/media/libaaudio/examples/input_monitor/Android.mk
index b56328b..5053e7d 100644
--- a/media/libaaudio/examples/input_monitor/Android.mk
+++ b/media/libaaudio/examples/input_monitor/Android.mk
@@ -1,6 +1 @@
-# include $(call all-subdir-makefiles)
-
-# Just include static/ for now.
-LOCAL_PATH := $(call my-dir)
-#include $(LOCAL_PATH)/jni/Android.mk
-include $(LOCAL_PATH)/static/Android.mk
+include $(call all-subdir-makefiles)
diff --git a/media/libaaudio/examples/input_monitor/jni/Android.mk b/media/libaaudio/examples/input_monitor/jni/Android.mk
index 3e24f9f..9b1ce2c 100644
--- a/media/libaaudio/examples/input_monitor/jni/Android.mk
+++ b/media/libaaudio/examples/input_monitor/jni/Android.mk
@@ -11,7 +11,7 @@
# NDK recommends using this kind of relative path instead of an absolute path.
LOCAL_SRC_FILES:= ../src/input_monitor.cpp
LOCAL_SHARED_LIBRARIES := libaaudio
-LOCAL_MODULE := input_monitor_ndk
+LOCAL_MODULE := input_monitor
include $(BUILD_EXECUTABLE)
include $(CLEAR_VARS)
@@ -23,11 +23,11 @@
LOCAL_SRC_FILES:= ../src/input_monitor_callback.cpp
LOCAL_SHARED_LIBRARIES := libaaudio
-LOCAL_MODULE := input_monitor_callback_ndk
+LOCAL_MODULE := input_monitor_callback
include $(BUILD_EXECUTABLE)
include $(CLEAR_VARS)
LOCAL_MODULE := libaaudio_prebuilt
LOCAL_SRC_FILES := libaaudio.so
LOCAL_EXPORT_C_INCLUDES := $(LOCAL_PATH)/include
-include $(PREBUILT_SHARED_LIBRARY)
\ No newline at end of file
+include $(PREBUILT_SHARED_LIBRARY)
diff --git a/media/libaaudio/examples/input_monitor/static/Android.mk b/media/libaaudio/examples/input_monitor/static/Android.mk
deleted file mode 100644
index 80a3906..0000000
--- a/media/libaaudio/examples/input_monitor/static/Android.mk
+++ /dev/null
@@ -1,37 +0,0 @@
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE_TAGS := samples
-LOCAL_C_INCLUDES := \
- $(call include-path-for, audio-utils) \
- frameworks/av/media/libaaudio/include \
- frameworks/av/media/libaaudio/examples/utils
-
-# TODO reorganize folders to avoid using ../
-LOCAL_SRC_FILES:= ../src/input_monitor.cpp
-
-LOCAL_SHARED_LIBRARIES := libaudioutils libmedia \
- libbinder libcutils libutils \
- libaudioclient liblog libtinyalsa libaudiomanager
-LOCAL_STATIC_LIBRARIES := libaaudio
-
-LOCAL_MODULE := input_monitor
-include $(BUILD_EXECUTABLE)
-
-
-include $(CLEAR_VARS)
-LOCAL_MODULE_TAGS := tests
-LOCAL_C_INCLUDES := \
- $(call include-path-for, audio-utils) \
- frameworks/av/media/libaaudio/include \
- frameworks/av/media/libaaudio/examples/utils
-
-LOCAL_SRC_FILES:= ../src/input_monitor_callback.cpp
-
-LOCAL_SHARED_LIBRARIES := libaudioutils libmedia \
- libbinder libcutils libutils \
- libaudioclient liblog libaudiomanager
-LOCAL_STATIC_LIBRARIES := libaaudio
-
-LOCAL_MODULE := input_monitor_callback
-include $(BUILD_EXECUTABLE)
diff --git a/media/libaaudio/examples/input_monitor/static/README.md b/media/libaaudio/examples/input_monitor/static/README.md
deleted file mode 100644
index 6e26d7b..0000000
--- a/media/libaaudio/examples/input_monitor/static/README.md
+++ /dev/null
@@ -1,2 +0,0 @@
-Makefile for building simple command line examples.
-They link with AAudio as a static library.
diff --git a/media/libaaudio/examples/write_sine/jni/Android.mk b/media/libaaudio/examples/write_sine/jni/Android.mk
index c306ed3..d630e76 100644
--- a/media/libaaudio/examples/write_sine/jni/Android.mk
+++ b/media/libaaudio/examples/write_sine/jni/Android.mk
@@ -11,7 +11,7 @@
# NDK recommends using this kind of relative path instead of an absolute path.
LOCAL_SRC_FILES:= ../src/write_sine.cpp
LOCAL_SHARED_LIBRARIES := libaaudio
-LOCAL_MODULE := write_sine_ndk
+LOCAL_MODULE := write_sine
include $(BUILD_EXECUTABLE)
include $(CLEAR_VARS)
@@ -23,7 +23,7 @@
LOCAL_SRC_FILES:= ../src/write_sine_callback.cpp
LOCAL_SHARED_LIBRARIES := libaaudio
-LOCAL_MODULE := write_sine_callback_ndk
+LOCAL_MODULE := write_sine_callback
include $(BUILD_EXECUTABLE)
include $(CLEAR_VARS)
diff --git a/media/libaaudio/examples/write_sine/static/Android.mk b/media/libaaudio/examples/write_sine/static/Android.mk
deleted file mode 100644
index 1f8dcd9..0000000
--- a/media/libaaudio/examples/write_sine/static/Android.mk
+++ /dev/null
@@ -1,38 +0,0 @@
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE_TAGS := samples
-LOCAL_C_INCLUDES := \
- $(call include-path-for, audio-utils) \
- frameworks/av/media/libaaudio/src \
- frameworks/av/media/libaaudio/include \
- frameworks/av/media/libaaudio/examples/utils
-
-# NDK recommends using this kind of relative path instead of an absolute path.
-LOCAL_SRC_FILES:= ../src/write_sine.cpp
-
-LOCAL_SHARED_LIBRARIES := libaudioutils libmedia \
- libbinder libcutils libutils \
- libaudioclient liblog libtinyalsa libaudiomanager
-LOCAL_STATIC_LIBRARIES := libaaudio
-
-LOCAL_MODULE := write_sine
-include $(BUILD_EXECUTABLE)
-
-
-include $(CLEAR_VARS)
-LOCAL_MODULE_TAGS := tests
-LOCAL_C_INCLUDES := \
- $(call include-path-for, audio-utils) \
- frameworks/av/media/libaaudio/include \
- frameworks/av/media/libaaudio/examples/utils
-
-LOCAL_SRC_FILES:= ../src/write_sine_callback.cpp
-
-LOCAL_SHARED_LIBRARIES := libaudioutils libmedia \
- libbinder libcutils libutils \
- libaudioclient liblog libaudiomanager
-LOCAL_STATIC_LIBRARIES := libaaudio
-
-LOCAL_MODULE := write_sine_callback
-include $(BUILD_EXECUTABLE)
diff --git a/media/libaaudio/examples/write_sine/static/README.md b/media/libaaudio/examples/write_sine/static/README.md
deleted file mode 100644
index 6e26d7b..0000000
--- a/media/libaaudio/examples/write_sine/static/README.md
+++ /dev/null
@@ -1,2 +0,0 @@
-Makefile for building simple command line examples.
-They link with AAudio as a static library.
diff --git a/media/libaaudio/src/binding/IAAudioService.cpp b/media/libaaudio/src/binding/IAAudioService.cpp
index b8ef611..ef4a51f 100644
--- a/media/libaaudio/src/binding/IAAudioService.cpp
+++ b/media/libaaudio/src/binding/IAAudioService.cpp
@@ -45,11 +45,9 @@
Parcel data, reply;
// send command
data.writeInterfaceToken(IAAudioService::getInterfaceDescriptor());
- ALOGV("BpAAudioService::client openStream --------------------");
// request.dump();
request.writeToParcel(&data);
status_t err = remote()->transact(OPEN_STREAM, data, &reply);
- ALOGV("BpAAudioService::client openStream returned %d", err);
if (err != NO_ERROR) {
ALOGE("BpAAudioService::client openStream transact failed %d", err);
return AAudioConvert_androidToAAudioResult(err);
@@ -57,6 +55,7 @@
// parse reply
aaudio_handle_t stream;
err = reply.readInt32(&stream);
+ ALOGD("BpAAudioService::client openStream returned stream = 0x%08x", stream);
if (err != NO_ERROR) {
ALOGE("BpAAudioService::client transact(OPEN_STREAM) readInt %d", err);
return AAudioConvert_androidToAAudioResult(err);
diff --git a/media/libaaudio/src/client/AudioEndpoint.cpp b/media/libaaudio/src/client/AudioEndpoint.cpp
index 5cb642b..0684ed6 100644
--- a/media/libaaudio/src/client/AudioEndpoint.cpp
+++ b/media/libaaudio/src/client/AudioEndpoint.cpp
@@ -246,3 +246,7 @@
return (int32_t)mDataQueue->getBufferCapacityInFrames();
}
+void AudioEndpoint::dump() const {
+ ALOGD("AudioEndpoint: data readCounter = %lld", (long long) mDataQueue->getReadCounter());
+ ALOGD("AudioEndpoint: data writeCounter = %lld", (long long) mDataQueue->getWriteCounter());
+}
diff --git a/media/libaaudio/src/client/AudioEndpoint.h b/media/libaaudio/src/client/AudioEndpoint.h
index 53ba033..e7c6916 100644
--- a/media/libaaudio/src/client/AudioEndpoint.h
+++ b/media/libaaudio/src/client/AudioEndpoint.h
@@ -91,6 +91,8 @@
int32_t getBufferCapacityInFrames() const;
+ void dump() const;
+
private:
android::FifoBuffer *mUpCommandQueue;
android::FifoBuffer *mDataQueue;
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 3a827f0..b59c445 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -62,8 +62,9 @@
, mAudioEndpoint()
, mServiceStreamHandle(AAUDIO_HANDLE_INVALID)
, mFramesPerBurst(16)
- , mServiceInterface(serviceInterface)
- , mInService(inService) {
+ , mStreamVolume(1.0f)
+ , mInService(inService)
+ , mServiceInterface(serviceInterface) {
}
AudioStreamInternal::~AudioStreamInternal() {
@@ -153,13 +154,13 @@
if (getDataCallbackProc()) {
mCallbackFrames = builder.getFramesPerDataCallback();
if (mCallbackFrames > getBufferCapacity() / 2) {
- ALOGE("AudioStreamInternal.open(): framesPerCallback too large = %d, capacity = %d",
+ ALOGE("AudioStreamInternal::open(): framesPerCallback too big = %d, capacity = %d",
mCallbackFrames, getBufferCapacity());
mServiceInterface.closeStream(mServiceStreamHandle);
return AAUDIO_ERROR_OUT_OF_RANGE;
} else if (mCallbackFrames < 0) {
- ALOGE("AudioStreamInternal.open(): framesPerCallback negative");
+ ALOGE("AudioStreamInternal::open(): framesPerCallback negative");
mServiceInterface.closeStream(mServiceStreamHandle);
return AAUDIO_ERROR_OUT_OF_RANGE;
@@ -175,12 +176,16 @@
}
setState(AAUDIO_STREAM_STATE_OPEN);
+ // only connect to AudioManager if this is a playback stream running in client process
+ if (!mInService && getDirection() == AAUDIO_DIRECTION_OUTPUT) {
+ init(android::PLAYER_TYPE_AAUDIO, AUDIO_USAGE_MEDIA);
+ }
}
return result;
}
aaudio_result_t AudioStreamInternal::close() {
- ALOGD("AudioStreamInternal.close(): mServiceStreamHandle = 0x%08X",
+ ALOGD("AudioStreamInternal::close(): mServiceStreamHandle = 0x%08X",
mServiceStreamHandle);
if (mServiceStreamHandle != AAUDIO_HANDLE_INVALID) {
// Don't close a stream while it is running.
@@ -196,12 +201,14 @@
result, AAudio_convertResultToText(result));
}
}
+ setState(AAUDIO_STREAM_STATE_CLOSING);
aaudio_handle_t serviceStreamHandle = mServiceStreamHandle;
mServiceStreamHandle = AAUDIO_HANDLE_INVALID;
mServiceInterface.closeStream(serviceStreamHandle);
delete[] mCallbackBuffer;
mCallbackBuffer = nullptr;
+ setState(AAUDIO_STREAM_STATE_CLOSED);
return mEndPointParcelable.close();
} else {
return AAUDIO_ERROR_INVALID_HANDLE;
@@ -223,15 +230,20 @@
aaudio_result_t AudioStreamInternal::requestStart()
{
int64_t startTime;
- ALOGD("AudioStreamInternal(): start()");
+ ALOGD("AudioStreamInternal()::requestStart()");
if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
return AAUDIO_ERROR_INVALID_STATE;
}
+ if (isActive()) {
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+ aaudio_stream_state_t originalState = getState();
+
+ setState(AAUDIO_STREAM_STATE_STARTING);
+ aaudio_result_t result = AAudioConvert_androidToAAudioResult(startWithStatus());
startTime = AudioClock::getNanoseconds();
mClockModel.start(startTime);
- setState(AAUDIO_STREAM_STATE_STARTING);
- aaudio_result_t result = mServiceInterface.startStream(mServiceStreamHandle);;
if (result == AAUDIO_OK && getDataCallbackProc() != nullptr) {
// Launch the callback loop thread.
@@ -241,6 +253,9 @@
mCallbackEnabled.store(true);
result = createThread(periodNanos, aaudio_callback_thread_proc, this);
}
+ if (result != AAUDIO_OK) {
+ setState(originalState);
+ }
return result;
}
@@ -274,14 +289,14 @@
aaudio_result_t AudioStreamInternal::requestPauseInternal()
{
if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
- ALOGE("AudioStreamInternal(): requestPauseInternal() mServiceStreamHandle invalid = 0x%08X",
+ ALOGE("AudioStreamInternal::requestPauseInternal() mServiceStreamHandle invalid = 0x%08X",
mServiceStreamHandle);
return AAUDIO_ERROR_INVALID_STATE;
}
mClockModel.stop(AudioClock::getNanoseconds());
setState(AAUDIO_STREAM_STATE_PAUSING);
- return mServiceInterface.pauseStream(mServiceStreamHandle);
+ return AAudioConvert_androidToAAudioResult(pauseWithStatus());
}
aaudio_result_t AudioStreamInternal::requestPause()
@@ -296,7 +311,7 @@
aaudio_result_t AudioStreamInternal::requestFlush() {
if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
- ALOGE("AudioStreamInternal(): requestFlush() mServiceStreamHandle invalid = 0x%08X",
+ ALOGE("AudioStreamInternal::requestFlush() mServiceStreamHandle invalid = 0x%08X",
mServiceStreamHandle);
return AAUDIO_ERROR_INVALID_STATE;
}
@@ -307,13 +322,14 @@
// TODO for Play only
void AudioStreamInternal::onFlushFromServer() {
- ALOGD("AudioStreamInternal(): onFlushFromServer()");
int64_t readCounter = mAudioEndpoint.getDataReadCounter();
int64_t writeCounter = mAudioEndpoint.getDataWriteCounter();
// Bump offset so caller does not see the retrograde motion in getFramesRead().
int64_t framesFlushed = writeCounter - readCounter;
mFramesOffsetFromService += framesFlushed;
+ ALOGD("AudioStreamInternal::onFlushFromServer() readN = %lld, writeN = %lld, offset = %lld",
+ (long long)readCounter, (long long)writeCounter, (long long)mFramesOffsetFromService);
// Flush written frames by forcing writeCounter to readCounter.
// This is because we cannot move the read counter in the hardware.
@@ -323,14 +339,14 @@
aaudio_result_t AudioStreamInternal::requestStopInternal()
{
if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
- ALOGE("AudioStreamInternal(): requestStopInternal() mServiceStreamHandle invalid = 0x%08X",
+ ALOGE("AudioStreamInternal::requestStopInternal() mServiceStreamHandle invalid = 0x%08X",
mServiceStreamHandle);
return AAUDIO_ERROR_INVALID_STATE;
}
mClockModel.stop(AudioClock::getNanoseconds());
setState(AAUDIO_STREAM_STATE_STOPPING);
- return mServiceInterface.stopStream(mServiceStreamHandle);
+ return AAudioConvert_androidToAAudioResult(stopWithStatus());
}
aaudio_result_t AudioStreamInternal::requestStop()
@@ -365,7 +381,7 @@
int64_t *timeNanoseconds) {
// TODO Generate in server and pass to client. Return latest.
int64_t time = AudioClock::getNanoseconds();
- *framePosition = mClockModel.convertTimeToPosition(time);
+ *framePosition = mClockModel.convertTimeToPosition(time) + mFramesOffsetFromService;
// TODO Get a more accurate timestamp from the service. This code just adds a fudge factor.
*timeNanoseconds = time + (6 * AAUDIO_NANOS_PER_MILLISECOND);
return AAUDIO_OK;
@@ -378,31 +394,28 @@
return processCommands();
}
-#if LOG_TIMESTAMPS
-static void AudioStreamInternal_logTimestamp(AAudioServiceMessage &command) {
+void AudioStreamInternal::logTimestamp(AAudioServiceMessage &command) {
static int64_t oldPosition = 0;
static int64_t oldTime = 0;
int64_t framePosition = command.timestamp.position;
int64_t nanoTime = command.timestamp.timestamp;
- ALOGD("AudioStreamInternal() timestamp says framePosition = %08lld at nanoTime %lld",
+ ALOGD("AudioStreamInternal: timestamp says framePosition = %08lld at nanoTime %lld",
(long long) framePosition,
(long long) nanoTime);
int64_t nanosDelta = nanoTime - oldTime;
if (nanosDelta > 0 && oldTime > 0) {
int64_t framesDelta = framePosition - oldPosition;
int64_t rate = (framesDelta * AAUDIO_NANOS_PER_SECOND) / nanosDelta;
- ALOGD("AudioStreamInternal() - framesDelta = %08lld", (long long) framesDelta);
- ALOGD("AudioStreamInternal() - nanosDelta = %08lld", (long long) nanosDelta);
- ALOGD("AudioStreamInternal() - measured rate = %lld", (long long) rate);
+ ALOGD("AudioStreamInternal: framesDelta = %08lld, nanosDelta = %08lld, rate = %lld",
+ (long long) framesDelta, (long long) nanosDelta, (long long) rate);
}
oldPosition = framePosition;
oldTime = nanoTime;
}
-#endif
aaudio_result_t AudioStreamInternal::onTimestampFromServer(AAudioServiceMessage *message) {
#if LOG_TIMESTAMPS
- AudioStreamInternal_logTimestamp(*message);
+ logTimestamp(*message);
#endif
processTimestamp(message->timestamp.position, message->timestamp.timestamp);
return AAUDIO_OK;
@@ -412,46 +425,48 @@
aaudio_result_t result = AAUDIO_OK;
switch (message->event.event) {
case AAUDIO_SERVICE_EVENT_STARTED:
- ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_STARTED");
+ ALOGD("AudioStreamInternal::onEventFromServergot() AAUDIO_SERVICE_EVENT_STARTED");
if (getState() == AAUDIO_STREAM_STATE_STARTING) {
setState(AAUDIO_STREAM_STATE_STARTED);
}
break;
case AAUDIO_SERVICE_EVENT_PAUSED:
- ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_PAUSED");
+ ALOGD("AudioStreamInternal::onEventFromServergot() AAUDIO_SERVICE_EVENT_PAUSED");
if (getState() == AAUDIO_STREAM_STATE_PAUSING) {
setState(AAUDIO_STREAM_STATE_PAUSED);
}
break;
case AAUDIO_SERVICE_EVENT_STOPPED:
- ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_STOPPED");
+ ALOGD("AudioStreamInternal::onEventFromServergot() AAUDIO_SERVICE_EVENT_STOPPED");
if (getState() == AAUDIO_STREAM_STATE_STOPPING) {
setState(AAUDIO_STREAM_STATE_STOPPED);
}
break;
case AAUDIO_SERVICE_EVENT_FLUSHED:
- ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_FLUSHED");
+ ALOGD("AudioStreamInternal::onEventFromServer() got AAUDIO_SERVICE_EVENT_FLUSHED");
if (getState() == AAUDIO_STREAM_STATE_FLUSHING) {
setState(AAUDIO_STREAM_STATE_FLUSHED);
onFlushFromServer();
}
break;
case AAUDIO_SERVICE_EVENT_CLOSED:
- ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_CLOSED");
+ ALOGD("AudioStreamInternal::onEventFromServer() got AAUDIO_SERVICE_EVENT_CLOSED");
setState(AAUDIO_STREAM_STATE_CLOSED);
break;
case AAUDIO_SERVICE_EVENT_DISCONNECTED:
result = AAUDIO_ERROR_DISCONNECTED;
setState(AAUDIO_STREAM_STATE_DISCONNECTED);
- ALOGW("WARNING - processCommands() AAUDIO_SERVICE_EVENT_DISCONNECTED");
+ ALOGW("WARNING - AudioStreamInternal::onEventFromServer()"
+ " AAUDIO_SERVICE_EVENT_DISCONNECTED");
break;
case AAUDIO_SERVICE_EVENT_VOLUME:
- mVolumeRamp.setTarget((float) message->event.dataDouble);
- ALOGD("processCommands() AAUDIO_SERVICE_EVENT_VOLUME %lf",
+ mStreamVolume = (float)message->event.dataDouble;
+ doSetVolume();
+ ALOGD("AudioStreamInternal::onEventFromServer() AAUDIO_SERVICE_EVENT_VOLUME %lf",
message->event.dataDouble);
break;
default:
- ALOGW("WARNING - processCommands() Unrecognized event = %d",
+ ALOGW("WARNING - AudioStreamInternal::onEventFromServer() Unrecognized event = %d",
(int) message->event.event);
break;
}
@@ -493,27 +508,27 @@
{
const char * traceName = (mInService) ? "aaWrtS" : "aaWrtC";
ATRACE_BEGIN(traceName);
- aaudio_result_t result = AAUDIO_OK;
- int32_t loopCount = 0;
- uint8_t* audioData = (uint8_t*)buffer;
- int64_t currentTimeNanos = AudioClock::getNanoseconds();
- int64_t deadlineNanos = currentTimeNanos + timeoutNanoseconds;
- int32_t framesLeft = numFrames;
-
int32_t fullFrames = mAudioEndpoint.getFullFramesAvailable();
if (ATRACE_ENABLED()) {
const char * traceName = (mInService) ? "aaFullS" : "aaFullC";
ATRACE_INT(traceName, fullFrames);
}
+ aaudio_result_t result = AAUDIO_OK;
+ int32_t loopCount = 0;
+ uint8_t* audioData = (uint8_t*)buffer;
+ int64_t currentTimeNanos = AudioClock::getNanoseconds();
+ const int64_t entryTimeNanos = currentTimeNanos;
+ const int64_t deadlineNanos = currentTimeNanos + timeoutNanoseconds;
+ int32_t framesLeft = numFrames;
+
// Loop until all the data has been processed or until a timeout occurs.
while (framesLeft > 0) {
- // The call to processDataNow() will not block. It will just read as much as it can.
+ // The call to processDataNow() will not block. It will just process as much as it can.
int64_t wakeTimeNanos = 0;
aaudio_result_t framesProcessed = processDataNow(audioData, framesLeft,
currentTimeNanos, &wakeTimeNanos);
if (framesProcessed < 0) {
- ALOGE("AudioStreamInternal::processData() loop: framesProcessed = %d", framesProcessed);
result = framesProcessed;
break;
}
@@ -531,12 +546,16 @@
if (wakeTimeNanos > deadlineNanos) {
// If we time out, just return the framesWritten so far.
// TODO remove after we fix the deadline bug
- ALOGE("AudioStreamInternal::processData(): timed out after %lld nanos",
+ ALOGW("AudioStreamInternal::processData(): entered at %lld nanos, currently %lld",
+ (long long) entryTimeNanos, (long long) currentTimeNanos);
+ ALOGW("AudioStreamInternal::processData(): timed out after %lld nanos",
(long long) timeoutNanoseconds);
- ALOGE("AudioStreamInternal::processData(): wakeTime = %lld, deadline = %lld nanos",
+ ALOGW("AudioStreamInternal::processData(): wakeTime = %lld, deadline = %lld nanos",
(long long) wakeTimeNanos, (long long) deadlineNanos);
- ALOGE("AudioStreamInternal::processData(): past deadline by %d micros",
+ ALOGW("AudioStreamInternal::processData(): past deadline by %d micros",
(int)((wakeTimeNanos - deadlineNanos) / AAUDIO_NANOS_PER_MICROSECOND));
+ mClockModel.dump();
+ mAudioEndpoint.dump();
break;
}
@@ -588,3 +607,32 @@
aaudio_result_t AudioStreamInternal::joinThread(void** returnArg) {
return AudioStream::joinThread(returnArg, calculateReasonableTimeout(getFramesPerBurst()));
}
+
+void AudioStreamInternal::doSetVolume() {
+ // No pan and only left volume is taken into account from IPLayer interface
+ mVolumeRamp.setTarget(mStreamVolume * mVolumeMultiplierL /* * mPanMultiplierL */);
+}
+
+
+//------------------------------------------------------------------------------
+// Implementation of PlayerBase
+status_t AudioStreamInternal::playerStart() {
+ return AAudioConvert_aaudioToAndroidStatus(mServiceInterface.startStream(mServiceStreamHandle));
+}
+
+status_t AudioStreamInternal::playerPause() {
+ return AAudioConvert_aaudioToAndroidStatus(mServiceInterface.pauseStream(mServiceStreamHandle));
+}
+
+status_t AudioStreamInternal::playerStop() {
+ return AAudioConvert_aaudioToAndroidStatus(mServiceInterface.stopStream(mServiceStreamHandle));
+}
+
+status_t AudioStreamInternal::playerSetVolume() {
+ doSetVolume();
+ return NO_ERROR;
+}
+
+void AudioStreamInternal::destroy() {
+ baseDestroy();
+}
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index a11f309..c2c6419 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -18,6 +18,7 @@
#define ANDROID_AAUDIO_AUDIO_STREAM_INTERNAL_H
#include <stdint.h>
+#include <media/PlayerBase.h>
#include <aaudio/AAudio.h>
#include "binding/IAAudioService.h"
@@ -34,7 +35,7 @@
namespace aaudio {
// A stream that talks to the AAudioService or directly to a HAL.
-class AudioStreamInternal : public AudioStream {
+class AudioStreamInternal : public AudioStream, public android::PlayerBase {
public:
AudioStreamInternal(AAudioServiceInterface &serviceInterface, bool inService);
@@ -89,6 +90,9 @@
// Calculate timeout based on framesPerBurst
int64_t calculateReasonableTimeout();
+ //PlayerBase virtuals
+ virtual void destroy();
+
protected:
aaudio_result_t processData(void *buffer,
@@ -121,9 +125,19 @@
aaudio_result_t onTimestampFromServer(AAudioServiceMessage *message);
+ void logTimestamp(AAudioServiceMessage &message);
+
// Calculate timeout for an operation involving framesPerOperation.
int64_t calculateReasonableTimeout(int32_t framesPerOperation);
+ void doSetVolume();
+
+ //PlayerBase virtuals
+ virtual status_t playerStart();
+ virtual status_t playerPause();
+ virtual status_t playerStop();
+ virtual status_t playerSetVolume();
+
aaudio_format_t mDeviceFormat = AAUDIO_FORMAT_UNSPECIFIED;
IsochronousClockModel mClockModel; // timing model for chasing the HAL
@@ -135,6 +149,7 @@
int32_t mXRunCount = 0; // how many underrun events?
LinearRamp mVolumeRamp;
+ float mStreamVolume;
// Offset from underlying frame position.
int64_t mFramesOffsetFromService = 0; // offset for timestamps
@@ -142,6 +157,9 @@
uint8_t *mCallbackBuffer = nullptr;
int32_t mCallbackFrames = 0;
+ // The service uses this for SHARED mode.
+ bool mInService = false; // Is this running in the client or the service?
+
private:
/*
* Asynchronous write with data conversion.
@@ -159,8 +177,6 @@
EndpointDescriptor mEndpointDescriptor; // buffer description with resolved addresses
AAudioServiceInterface &mServiceInterface; // abstract interface to the service
- // The service uses this for SHARED mode.
- bool mInService = false; // Is this running in the client or the service?
};
} /* namespace aaudio */
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index 93693bd..22f8bd1 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -14,10 +14,11 @@
* limitations under the License.
*/
-#define LOG_TAG "AAudio"
+#define LOG_TAG (mInService ? "AAudioService" : "AAudio")
//#define LOG_NDEBUG 0
#include <utils/Log.h>
+#include <algorithm>
#include <aaudio/AAudio.h>
#include "client/AudioStreamInternalCapture.h"
@@ -155,29 +156,27 @@
int32_t framesProcessed = numFrames - framesLeft;
mAudioEndpoint.advanceReadIndex(framesProcessed);
- incrementFramesRead(framesProcessed);
//ALOGD("AudioStreamInternalCapture::readNowWithConversion() returns %d", framesProcessed);
return framesProcessed;
}
-int64_t AudioStreamInternalCapture::getFramesWritten()
-{
- int64_t frames =
- mClockModel.convertTimeToPosition(AudioClock::getNanoseconds())
- + mFramesOffsetFromService;
- // Prevent retrograde motion.
- if (frames < mLastFramesWritten) {
- frames = mLastFramesWritten;
+int64_t AudioStreamInternalCapture::getFramesWritten() {
+ int64_t framesWrittenHardware;
+ if (isActive()) {
+ framesWrittenHardware = mClockModel.convertTimeToPosition(AudioClock::getNanoseconds());
} else {
- mLastFramesWritten = frames;
+ framesWrittenHardware = mAudioEndpoint.getDataWriteCounter();
}
- //ALOGD("AudioStreamInternalCapture::getFramesWritten() returns %lld", (long long)frames);
- return frames;
+ // Prevent retrograde motion.
+ mLastFramesWritten = std::max(mLastFramesWritten,
+ framesWrittenHardware + mFramesOffsetFromService);
+ //ALOGD("AudioStreamInternalCapture::getFramesWritten() returns %lld",
+ // (long long)mLastFramesWritten);
+ return mLastFramesWritten;
}
-int64_t AudioStreamInternalCapture::getFramesRead()
-{
+int64_t AudioStreamInternalCapture::getFramesRead() {
int64_t frames = mAudioEndpoint.getDataWriteCounter()
+ mFramesOffsetFromService;
//ALOGD("AudioStreamInternalCapture::getFramesRead() returns %lld", (long long)frames);
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index fc9766f..76ecbf9 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -14,7 +14,7 @@
* limitations under the License.
*/
-#define LOG_TAG "AAudio"
+#define LOG_TAG (mInService ? "AAudioService" : "AAudio")
//#define LOG_NDEBUG 0
#include <utils/Log.h>
@@ -51,15 +51,17 @@
}
if (mAudioEndpoint.isFreeRunning()) {
- //ALOGD("AudioStreamInternal::processDataNow() - update read counter");
// Update data queue based on the timing model.
int64_t estimatedReadCounter = mClockModel.convertTimeToPosition(currentNanoTime);
+ // ALOGD("AudioStreamInternal::processDataNow() - estimatedReadCounter = %d", (int)estimatedReadCounter);
mAudioEndpoint.setDataReadCounter(estimatedReadCounter);
}
- // TODO else query from endpoint cuz set by actual reader, maybe
// If the read index passed the write index then consider it an underrun.
if (mAudioEndpoint.getFullFramesAvailable() < 0) {
+ ALOGV("AudioStreamInternal::processDataNow() - XRun! write = %d, read = %d",
+ (int)mAudioEndpoint.getDataWriteCounter(),
+ (int)mAudioEndpoint.getDataReadCounter());
mXRunCount++;
}
@@ -201,9 +203,6 @@
int32_t framesWritten = numFrames - framesLeft;
mAudioEndpoint.advanceWriteIndex(framesWritten);
- if (framesWritten > 0) {
- incrementFramesWritten(framesWritten);
- }
// ALOGD("AudioStreamInternal::writeNowWithConversion() returns %d", framesWritten);
return framesWritten;
}
@@ -211,25 +210,29 @@
int64_t AudioStreamInternalPlay::getFramesRead()
{
- int64_t framesRead =
- mClockModel.convertTimeToPosition(AudioClock::getNanoseconds())
- + mFramesOffsetFromService;
+ int64_t framesReadHardware;
+ if (isActive()) {
+ framesReadHardware = mClockModel.convertTimeToPosition(AudioClock::getNanoseconds());
+ } else {
+ framesReadHardware = mAudioEndpoint.getDataReadCounter();
+ }
+ int64_t framesRead = framesReadHardware + mFramesOffsetFromService;
// Prevent retrograde motion.
if (framesRead < mLastFramesRead) {
framesRead = mLastFramesRead;
} else {
mLastFramesRead = framesRead;
}
- ALOGD("AudioStreamInternal::getFramesRead() returns %lld", (long long)framesRead);
+ //ALOGD("AudioStreamInternalPlay::getFramesRead() returns %lld", (long long)framesRead);
return framesRead;
}
int64_t AudioStreamInternalPlay::getFramesWritten()
{
- int64_t getFramesWritten = mAudioEndpoint.getDataWriteCounter()
+ int64_t framesWritten = mAudioEndpoint.getDataWriteCounter()
+ mFramesOffsetFromService;
- ALOGD("AudioStreamInternal::getFramesWritten() returns %lld", (long long)getFramesWritten);
- return getFramesWritten;
+ //ALOGD("AudioStreamInternalPlay::getFramesWritten() returns %lld", (long long)framesWritten);
+ return framesWritten;
}
diff --git a/media/libaaudio/src/client/IsochronousClockModel.cpp b/media/libaaudio/src/client/IsochronousClockModel.cpp
index 1de33bb..73f4c1d 100644
--- a/media/libaaudio/src/client/IsochronousClockModel.cpp
+++ b/media/libaaudio/src/client/IsochronousClockModel.cpp
@@ -41,6 +41,13 @@
IsochronousClockModel::~IsochronousClockModel() {
}
+void IsochronousClockModel::setPositionAndTime(int64_t framePosition, int64_t nanoTime) {
+ ALOGV("IsochronousClockModel::setPositionAndTime(%lld, %lld)",
+ (long long) framePosition, (long long) nanoTime);
+ mMarkerFramePosition = framePosition;
+ mMarkerNanoTime = nanoTime;
+}
+
void IsochronousClockModel::start(int64_t nanoTime) {
ALOGD("IsochronousClockModel::start(nanos = %lld)\n", (long long) nanoTime);
mMarkerNanoTime = nanoTime;
@@ -49,8 +56,8 @@
void IsochronousClockModel::stop(int64_t nanoTime) {
ALOGD("IsochronousClockModel::stop(nanos = %lld)\n", (long long) nanoTime);
- mMarkerNanoTime = nanoTime;
- mMarkerFramePosition = convertTimeToPosition(nanoTime); // TODO should we do this?
+ setPositionAndTime(convertTimeToPosition(nanoTime), nanoTime);
+ // TODO should we set position?
mState = STATE_STOPPED;
}
@@ -79,15 +86,13 @@
case STATE_STOPPED:
break;
case STATE_STARTING:
- mMarkerFramePosition = framePosition;
- mMarkerNanoTime = nanoTime;
+ setPositionAndTime(framePosition, nanoTime);
mState = STATE_SYNCING;
break;
case STATE_SYNCING:
// This will handle a burst of rapid transfer at the beginning.
if (nanosDelta < expectedNanosDelta) {
- mMarkerFramePosition = framePosition;
- mMarkerNanoTime = nanoTime;
+ setPositionAndTime(framePosition, nanoTime);
} else {
// ALOGD("processTimestamp() - advance to STATE_RUNNING");
mState = STATE_RUNNING;
@@ -98,17 +103,15 @@
// Earlier than expected timestamp.
// This data is probably more accurate so use it.
// or we may be drifting due to a slow HW clock.
- mMarkerFramePosition = framePosition;
- mMarkerNanoTime = nanoTime;
// ALOGD("processTimestamp() - STATE_RUNNING - %d < %d micros - EARLY",
// (int) (nanosDelta / 1000), (int)(expectedNanosDelta / 1000));
+ setPositionAndTime(framePosition, nanoTime);
} else if (nanosDelta > (expectedNanosDelta + mMaxLatenessInNanos)) {
// Later than expected timestamp.
- mMarkerFramePosition = framePosition;
- mMarkerNanoTime = nanoTime - mMaxLatenessInNanos;
// ALOGD("processTimestamp() - STATE_RUNNING - %d > %d + %d micros - LATE",
// (int) (nanosDelta / 1000), (int)(expectedNanosDelta / 1000),
// (int) (mMaxLatenessInNanos / 1000));
+ setPositionAndTime(framePosition - mFramesPerBurst, nanoTime - mMaxLatenessInNanos);
}
break;
default:
@@ -131,8 +134,7 @@
mMaxLatenessInNanos = (nanosLate > MIN_LATENESS_NANOS) ? nanosLate : MIN_LATENESS_NANOS;
}
-int64_t IsochronousClockModel::convertDeltaPositionToTime(
- int64_t framesDelta) const {
+int64_t IsochronousClockModel::convertDeltaPositionToTime(int64_t framesDelta) const {
return (AAUDIO_NANOS_PER_SECOND * framesDelta) / mSampleRate;
}
@@ -171,3 +173,12 @@
// (long long) framesDelta, mFramesPerBurst);
return position;
}
+
+void IsochronousClockModel::dump() const {
+ ALOGD("IsochronousClockModel::mMarkerFramePosition = %lld", (long long) mMarkerFramePosition);
+ ALOGD("IsochronousClockModel::mMarkerNanoTime = %lld", (long long) mMarkerNanoTime);
+ ALOGD("IsochronousClockModel::mSampleRate = %6d", mSampleRate);
+ ALOGD("IsochronousClockModel::mFramesPerBurst = %6d", mFramesPerBurst);
+ ALOGD("IsochronousClockModel::mMaxLatenessInNanos = %6d", mMaxLatenessInNanos);
+ ALOGD("IsochronousClockModel::mState = %6d", mState);
+}
diff --git a/media/libaaudio/src/client/IsochronousClockModel.h b/media/libaaudio/src/client/IsochronousClockModel.h
index 0314f55..585f53a 100644
--- a/media/libaaudio/src/client/IsochronousClockModel.h
+++ b/media/libaaudio/src/client/IsochronousClockModel.h
@@ -43,6 +43,8 @@
*/
void setSampleRate(int32_t sampleRate);
+ void setPositionAndTime(int64_t framePosition, int64_t nanoTime);
+
int32_t getSampleRate() const {
return mSampleRate;
}
@@ -86,6 +88,8 @@
*/
int64_t convertDeltaTimeToPosition(int64_t nanosDelta) const;
+ void dump() const;
+
private:
enum clock_model_state_t {
STATE_STOPPED,
diff --git a/media/libaaudio/src/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index 76f98fa..3f5de77 100644
--- a/media/libaaudio/src/core/AAudioAudio.cpp
+++ b/media/libaaudio/src/core/AAudioAudio.cpp
@@ -253,8 +253,10 @@
AAUDIO_API aaudio_result_t AAudioStream_requestStart(AAudioStream* stream)
{
AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
- ALOGD("AAudioStream_requestStart(%p)", stream);
- return audioStream->requestStart();
+ ALOGD("AAudioStream_requestStart(%p) called --------------", stream);
+ aaudio_result_t result = audioStream->requestStart();
+ ALOGD("AAudioStream_requestStart(%p) returned ------------", stream);
+ return result;
}
AAUDIO_API aaudio_result_t AAudioStream_requestPause(AAudioStream* stream)
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index 5a05c0e..19b08c4 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -36,6 +36,22 @@
setPeriodNanoseconds(0);
}
+static const char *AudioStream_convertSharingModeToShortText(aaudio_sharing_mode_t sharingMode) {
+ const char *result;
+ switch (sharingMode) {
+ case AAUDIO_SHARING_MODE_EXCLUSIVE:
+ result = "EX";
+ break;
+ case AAUDIO_SHARING_MODE_SHARED:
+ result = "SH";
+ break;
+ default:
+ result = "?!";
+ break;
+ }
+ return result;
+}
+
aaudio_result_t AudioStream::open(const AudioStreamBuilder& builder)
{
// Call here as well because the AAudioService will call this without calling build().
@@ -62,8 +78,9 @@
mErrorCallbackUserData = builder.getErrorCallbackUserData();
// This is very helpful for debugging in the future. Please leave it in.
- ALOGI("AudioStream::open() rate = %d, channels = %d, format = %d, sharing = %d, dir = %s",
- mSampleRate, mSamplesPerFrame, mFormat, mSharingMode,
+ ALOGI("AudioStream::open() rate = %d, channels = %d, format = %d, sharing = %s, dir = %s",
+ mSampleRate, mSamplesPerFrame, mFormat,
+ AudioStream_convertSharingModeToShortText(mSharingMode),
(getDirection() == AAUDIO_DIRECTION_OUTPUT) ? "OUTPUT" : "INPUT");
ALOGI("AudioStream::open() device = %d, perfMode = %d, callbackFrames = %d",
mDeviceId, mPerformanceMode, mFramesPerDataCallback);
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index 39c9f9c..7c7fcc5 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -186,13 +186,9 @@
return AAudioConvert_formatToSizeInBytes(mFormat);
}
- virtual int64_t getFramesWritten() {
- return mFramesWritten.get();
- }
+ virtual int64_t getFramesWritten() = 0;
- virtual int64_t getFramesRead() {
- return mFramesRead.get();
- }
+ virtual int64_t getFramesRead() = 0;
AAudioStream_dataCallback getDataCallbackProc() const {
return mDataCallbackProc;
@@ -232,13 +228,6 @@
protected:
- virtual int64_t incrementFramesWritten(int32_t frames) {
- return mFramesWritten.increment(frames);
- }
-
- virtual int64_t incrementFramesRead(int32_t frames) {
- return mFramesRead.increment(frames);
- }
/**
* This should not be called after the open() call.
@@ -281,8 +270,6 @@
std::atomic<bool> mCallbackEnabled;
protected:
- MonotonicCounter mFramesWritten;
- MonotonicCounter mFramesRead;
void setPeriodNanoseconds(int64_t periodNanoseconds) {
mPeriodNanoseconds.store(periodNanoseconds, std::memory_order_release);
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.h b/media/libaaudio/src/legacy/AudioStreamLegacy.h
index 0ded8e1..d2ef3c7 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.h
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.h
@@ -74,6 +74,15 @@
virtual int64_t incrementClientFrameCounter(int32_t frames) = 0;
+
+ virtual int64_t getFramesWritten() override {
+ return mFramesWritten.get();
+ }
+
+ virtual int64_t getFramesRead() override {
+ return mFramesRead.get();
+ }
+
protected:
class StreamDeviceCallback : public android::AudioSystem::AudioDeviceCallback
@@ -103,6 +112,17 @@
void onStart() { mCallbackEnabled.store(true); }
void onStop() { mCallbackEnabled.store(false); }
+ int64_t incrementFramesWritten(int32_t frames) {
+ return mFramesWritten.increment(frames);
+ }
+
+ int64_t incrementFramesRead(int32_t frames) {
+ return mFramesRead.increment(frames);
+ }
+
+ MonotonicCounter mFramesWritten;
+ MonotonicCounter mFramesRead;
+
FixedBlockAdapter *mBlockAdapter = nullptr;
aaudio_wrapping_frames_t mPositionWhenStarting = 0;
int32_t mCallbackBufferSize = 0;
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 7e39908..77f31e2 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -423,7 +423,7 @@
default:
break;
}
- return AudioStream::getFramesRead();
+ return AudioStreamLegacy::getFramesRead();
}
aaudio_result_t AudioStreamTrack::getTimestamp(clockid_t clockId,
diff --git a/media/libaaudio/src/utility/AAudioUtilities.cpp b/media/libaaudio/src/utility/AAudioUtilities.cpp
index 164784d..2d8ac6e 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.cpp
+++ b/media/libaaudio/src/utility/AAudioUtilities.cpp
@@ -208,9 +208,12 @@
status_t status;
switch (result) {
case AAUDIO_ERROR_DISCONNECTED:
- case AAUDIO_ERROR_INVALID_HANDLE:
+ case AAUDIO_ERROR_NO_SERVICE:
status = DEAD_OBJECT;
break;
+ case AAUDIO_ERROR_INVALID_HANDLE:
+ status = BAD_TYPE;
+ break;
case AAUDIO_ERROR_INVALID_STATE:
status = INVALID_OPERATION;
break;
@@ -233,7 +236,6 @@
case AAUDIO_ERROR_NO_FREE_HANDLES:
case AAUDIO_ERROR_NO_MEMORY:
case AAUDIO_ERROR_TIMEOUT:
- case AAUDIO_ERROR_NO_SERVICE:
default:
status = UNKNOWN_ERROR;
break;
@@ -257,12 +259,12 @@
case INVALID_OPERATION:
result = AAUDIO_ERROR_INVALID_STATE;
break;
- case UNEXPECTED_NULL:
- result = AAUDIO_ERROR_NULL;
- break;
- case BAD_VALUE:
- result = AAUDIO_ERROR_ILLEGAL_ARGUMENT;
- break;
+ case UNEXPECTED_NULL:
+ result = AAUDIO_ERROR_NULL;
+ break;
+ case BAD_VALUE:
+ result = AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+ break;
case WOULD_BLOCK:
result = AAUDIO_ERROR_WOULD_BLOCK;
break;
diff --git a/media/libaaudio/tests/Android.mk b/media/libaaudio/tests/Android.mk
index afcdebf..e6c779b 100644
--- a/media/libaaudio/tests/Android.mk
+++ b/media/libaaudio/tests/Android.mk
@@ -6,9 +6,7 @@
frameworks/av/media/libaaudio/include \
frameworks/av/media/libaaudio/src
LOCAL_SRC_FILES:= test_handle_tracker.cpp
-LOCAL_SHARED_LIBRARIES := libaudioclient libaudioutils libbinder \
- libcutils liblog libmedia libutils libaudiomanager
-LOCAL_STATIC_LIBRARIES := libaaudio
+LOCAL_SHARED_LIBRARIES := libaaudio
LOCAL_MODULE := test_handle_tracker
include $(BUILD_NATIVE_TEST)
@@ -18,9 +16,7 @@
frameworks/av/media/libaaudio/include \
frameworks/av/media/libaaudio/src
LOCAL_SRC_FILES:= test_marshalling.cpp
-LOCAL_SHARED_LIBRARIES := libaudioclient libaudioutils libbinder \
- libcutils liblog libmedia libutils libaudiomanager
-LOCAL_STATIC_LIBRARIES := libaaudio
+LOCAL_SHARED_LIBRARIES := libaaudio libbinder libcutils libutils
LOCAL_MODULE := test_aaudio_marshalling
include $(BUILD_NATIVE_TEST)
@@ -30,9 +26,7 @@
frameworks/av/media/libaaudio/include \
frameworks/av/media/libaaudio/src
LOCAL_SRC_FILES:= test_block_adapter.cpp
-LOCAL_SHARED_LIBRARIES := libaudioclient libaudioutils libbinder \
- libcutils liblog libmedia libutils libaudiomanager
-LOCAL_STATIC_LIBRARIES := libaaudio
+LOCAL_SHARED_LIBRARIES := libaaudio
LOCAL_MODULE := test_block_adapter
include $(BUILD_NATIVE_TEST)
@@ -42,9 +36,7 @@
frameworks/av/media/libaaudio/include \
frameworks/av/media/libaaudio/src
LOCAL_SRC_FILES:= test_linear_ramp.cpp
-LOCAL_SHARED_LIBRARIES := libaudioclient libaudioutils libbinder \
- libcutils liblog libmedia libutils libaudiomanager
-LOCAL_STATIC_LIBRARIES := libaaudio
+LOCAL_SHARED_LIBRARIES := libaaudio
LOCAL_MODULE := test_linear_ramp
include $(BUILD_NATIVE_TEST)
@@ -54,8 +46,6 @@
frameworks/av/media/libaaudio/include \
frameworks/av/media/libaaudio/src
LOCAL_SRC_FILES:= test_open_params.cpp
-LOCAL_SHARED_LIBRARIES := libaudioclient libaudioutils libbinder \
- libcutils liblog libmedia libutils libaudiomanager
-LOCAL_STATIC_LIBRARIES := libaaudio
+LOCAL_SHARED_LIBRARIES := libaaudio libbinder libcutils libutils
LOCAL_MODULE := test_open_params
include $(BUILD_NATIVE_TEST)
diff --git a/media/libaudioclient/AudioRecord.cpp b/media/libaudioclient/AudioRecord.cpp
index d29aa74..13198e8 100644
--- a/media/libaudioclient/AudioRecord.cpp
+++ b/media/libaudioclient/AudioRecord.cpp
@@ -507,6 +507,18 @@
}
// -------------------------------------------------------------------------
+// TODO Move this macro to a common header file for enum to string conversion in audio framework.
+#define MEDIA_CASE_ENUM(name) case name: return #name
+const char * AudioRecord::convertTransferToText(transfer_type transferType) {
+ switch (transferType) {
+ MEDIA_CASE_ENUM(TRANSFER_DEFAULT);
+ MEDIA_CASE_ENUM(TRANSFER_CALLBACK);
+ MEDIA_CASE_ENUM(TRANSFER_OBTAIN);
+ MEDIA_CASE_ENUM(TRANSFER_SYNC);
+ default:
+ return "UNRECOGNIZED";
+ }
+}
// must be called with mLock held
status_t AudioRecord::openRecord_l(const Modulo<uint32_t> &epoch, const String16& opPackageName)
@@ -590,12 +602,20 @@
(mTransfer == TRANSFER_SYNC) ||
// use case 3: obtain/release mode
(mTransfer == TRANSFER_OBTAIN);
+ if (!useCaseAllowed) {
+ ALOGW("AUDIO_INPUT_FLAG_FAST denied, incompatible transfer = %s",
+ convertTransferToText(mTransfer));
+ }
+
// sample rates must also match
- bool fastAllowed = useCaseAllowed && (mSampleRate == afSampleRate);
+ bool sampleRateAllowed = mSampleRate == afSampleRate;
+ if (!sampleRateAllowed) {
+ ALOGW("AUDIO_INPUT_FLAG_FAST denied, rates do not match %u Hz, require %u Hz",
+ mSampleRate, afSampleRate);
+ }
+
+ bool fastAllowed = useCaseAllowed && sampleRateAllowed;
if (!fastAllowed) {
- ALOGW("AUDIO_INPUT_FLAG_FAST denied by client; transfer %d, "
- "track %u Hz, input %u Hz",
- mTransfer, mSampleRate, afSampleRate);
mFlags = (audio_input_flags_t) (mFlags & ~(AUDIO_INPUT_FLAG_FAST |
AUDIO_INPUT_FLAG_RAW));
AudioSystem::releaseInput(input, mSessionId);
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 98ec7d7..e1805a7 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -1268,6 +1268,20 @@
}
}
+// TODO Move this macro to a common header file for enum to string conversion in audio framework.
+#define MEDIA_CASE_ENUM(name) case name: return #name
+const char * AudioTrack::convertTransferToText(transfer_type transferType) {
+ switch (transferType) {
+ MEDIA_CASE_ENUM(TRANSFER_DEFAULT);
+ MEDIA_CASE_ENUM(TRANSFER_CALLBACK);
+ MEDIA_CASE_ENUM(TRANSFER_OBTAIN);
+ MEDIA_CASE_ENUM(TRANSFER_SYNC);
+ MEDIA_CASE_ENUM(TRANSFER_SHARED);
+ default:
+ return "UNRECOGNIZED";
+ }
+}
+
status_t AudioTrack::createTrack_l()
{
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
@@ -1343,22 +1357,32 @@
// Client can only express a preference for FAST. Server will perform additional tests.
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
- bool useCaseAllowed =
- // either of these use cases:
- // use case 1: shared buffer
- (mSharedBuffer != 0) ||
+ // either of these use cases:
+ // use case 1: shared buffer
+ bool sharedBuffer = mSharedBuffer != 0;
+ bool transferAllowed =
// use case 2: callback transfer mode
(mTransfer == TRANSFER_CALLBACK) ||
// use case 3: obtain/release mode
(mTransfer == TRANSFER_OBTAIN) ||
// use case 4: synchronous write
((mTransfer == TRANSFER_SYNC) && mThreadCanCallJava);
+
+ bool useCaseAllowed = sharedBuffer || transferAllowed;
+ if (!useCaseAllowed) {
+ ALOGW("AUDIO_OUTPUT_FLAG_FAST denied, not shared buffer and transfer = %s",
+ convertTransferToText(mTransfer));
+ }
+
// sample rates must also match
- bool fastAllowed = useCaseAllowed && (mSampleRate == mAfSampleRate);
+ bool sampleRateAllowed = mSampleRate == mAfSampleRate;
+ if (!sampleRateAllowed) {
+ ALOGW("AUDIO_OUTPUT_FLAG_FAST denied, rates do not match %u Hz, require %u Hz",
+ mSampleRate, mAfSampleRate);
+ }
+
+ bool fastAllowed = useCaseAllowed && sampleRateAllowed;
if (!fastAllowed) {
- ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client; transfer %d, "
- "track %u Hz, output %u Hz",
- mTransfer, mSampleRate, mAfSampleRate);
mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
}
}
diff --git a/media/libaudioclient/PlayerBase.cpp b/media/libaudioclient/PlayerBase.cpp
index cbef1b3..7868318 100644
--- a/media/libaudioclient/PlayerBase.cpp
+++ b/media/libaudioclient/PlayerBase.cpp
@@ -79,7 +79,7 @@
}
}
-//FIXME temporary method while some AudioTrack state is outside of this class
+//FIXME temporary method while some player state is outside of this class
void PlayerBase::reportEvent(player_state_t event) {
servicePlayerEvent(event);
}
@@ -87,10 +87,30 @@
status_t PlayerBase::startWithStatus() {
status_t status = playerStart();
if (status == NO_ERROR) {
- ALOGD("PlayerBase::start() from IPlayer");
servicePlayerEvent(PLAYER_STATE_STARTED);
} else {
- ALOGD("PlayerBase::start() no AudioTrack to start from IPlayer");
+ ALOGW("PlayerBase::start() error %d", status);
+ }
+ return status;
+}
+
+status_t PlayerBase::pauseWithStatus() {
+ status_t status = playerPause();
+ if (status == NO_ERROR) {
+ servicePlayerEvent(PLAYER_STATE_PAUSED);
+ } else {
+ ALOGW("PlayerBase::pause() error %d", status);
+ }
+ return status;
+}
+
+
+status_t PlayerBase::stopWithStatus() {
+ status_t status = playerStop();
+ if (status == NO_ERROR) {
+ servicePlayerEvent(PLAYER_STATE_STOPPED);
+ } else {
+ ALOGW("PlayerBase::stop() error %d", status);
}
return status;
}
@@ -98,42 +118,36 @@
//------------------------------------------------------------------------------
// Implementation of IPlayer
void PlayerBase::start() {
+ ALOGD("PlayerBase::start() from IPlayer");
(void)startWithStatus();
}
void PlayerBase::pause() {
- if (playerPause() == NO_ERROR) {
- ALOGD("PlayerBase::pause() from IPlayer");
- servicePlayerEvent(PLAYER_STATE_PAUSED);
- } else {
- ALOGD("PlayerBase::pause() no AudioTrack to pause from IPlayer");
- }
+ ALOGD("PlayerBase::pause() from IPlayer");
+ (void)pauseWithStatus();
}
void PlayerBase::stop() {
- if (playerStop() == NO_ERROR) {
- ALOGD("PlayerBase::stop() from IPlayer");
- servicePlayerEvent(PLAYER_STATE_STOPPED);
- } else {
- ALOGD("PlayerBase::stop() no AudioTrack to stop from IPlayer");
- }
+ ALOGD("PlayerBase::stop() from IPlayer");
+ (void)stopWithStatus();
}
void PlayerBase::setVolume(float vol) {
+ ALOGD("PlayerBase::setVolume() from IPlayer");
{
Mutex::Autolock _l(mSettingsLock);
mVolumeMultiplierL = vol;
mVolumeMultiplierR = vol;
}
- if (playerSetVolume() == NO_ERROR) {
- ALOGD("PlayerBase::setVolume() from IPlayer");
- } else {
- ALOGD("PlayerBase::setVolume() no AudioTrack for volume control from IPlayer");
+ status_t status = playerSetVolume();
+ if (status != NO_ERROR) {
+ ALOGW("PlayerBase::setVolume() error %d", status);
}
}
void PlayerBase::setPan(float pan) {
+ ALOGD("PlayerBase::setPan() from IPlayer");
{
Mutex::Autolock _l(mSettingsLock);
pan = min(max(-1.0f, pan), 1.0f);
@@ -145,10 +159,9 @@
mPanMultiplierR = 1.0f + pan;
}
}
- if (playerSetVolume() == NO_ERROR) {
- ALOGD("PlayerBase::setPan() from IPlayer");
- } else {
- ALOGD("PlayerBase::setPan() no AudioTrack for volume control from IPlayer");
+ status_t status = playerSetVolume();
+ if (status != NO_ERROR) {
+ ALOGW("PlayerBase::setPan() error %d", status);
}
}
diff --git a/media/libaudioclient/include/media/AudioRecord.h b/media/libaudioclient/include/media/AudioRecord.h
index 1b034b5..825a0a0 100644
--- a/media/libaudioclient/include/media/AudioRecord.h
+++ b/media/libaudioclient/include/media/AudioRecord.h
@@ -334,6 +334,12 @@
*/
status_t getTimestamp(ExtendedTimestamp *timestamp);
+ /**
+ * @param transferType
+ * @return text string that matches the enum name
+ */
+ static const char * convertTransferToText(transfer_type transferType);
+
/* Returns a handle on the audio input used by this AudioRecord.
*
* Parameters:
diff --git a/media/libaudioclient/include/media/AudioTrack.h b/media/libaudioclient/include/media/AudioTrack.h
index 5d73df3..c5dfedc 100644
--- a/media/libaudioclient/include/media/AudioTrack.h
+++ b/media/libaudioclient/include/media/AudioTrack.h
@@ -564,6 +564,12 @@
*/
status_t reload();
+ /**
+ * @param transferType
+ * @return text string that matches the enum name
+ */
+ static const char * convertTransferToText(transfer_type transferType);
+
/* Returns a handle on the audio output used by this AudioTrack.
*
* Parameters:
diff --git a/media/libaudioclient/include/media/PlayerBase.h b/media/libaudioclient/include/media/PlayerBase.h
index fe1db7b..e63090b 100644
--- a/media/libaudioclient/include/media/PlayerBase.h
+++ b/media/libaudioclient/include/media/PlayerBase.h
@@ -48,6 +48,8 @@
status_t startWithStatus();
+ status_t pauseWithStatus();
+ status_t stopWithStatus();
//FIXME temporary method while some player state is outside of this class
void reportEvent(player_state_t event);
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index 6a5a229..f2b1f10 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -66,7 +66,8 @@
mNumFramesReceived(0),
mNumFramesSkipped(0),
mNumFramesLost(0),
- mNumClientOwnedBuffers(0) {
+ mNumClientOwnedBuffers(0),
+ mNoMoreFramesToRead(false) {
ALOGV("sampleRate: %u, outSampleRate: %u, channelCount: %u",
sampleRate, outSampleRate, channelCount);
CHECK(channelCount == 1 || channelCount == 2);
@@ -178,6 +179,7 @@
mStarted = false;
mStopSystemTimeUs = -1;
+ mNoMoreFramesToRead = false;
mFrameAvailableCondition.signal();
mRecord->stop();
@@ -246,6 +248,9 @@
while (mStarted && mBuffersReceived.empty()) {
mFrameAvailableCondition.wait(mLock);
+ if (mNoMoreFramesToRead) {
+ return OK;
+ }
}
if (!mStarted) {
return OK;
@@ -359,6 +364,8 @@
if (mStopSystemTimeUs != -1 && timeUs >= mStopSystemTimeUs) {
ALOGV("Drop Audio frame at %lld stop time: %lld us",
(long long)timeUs, (long long)mStopSystemTimeUs);
+ mNoMoreFramesToRead = true;
+ mFrameAvailableCondition.signal();
return OK;
}
diff --git a/media/libstagefright/include/media/stagefright/AudioSource.h b/media/libstagefright/include/media/stagefright/AudioSource.h
index 07a51bf..1595be4 100644
--- a/media/libstagefright/include/media/stagefright/AudioSource.h
+++ b/media/libstagefright/include/media/stagefright/AudioSource.h
@@ -95,6 +95,7 @@
int64_t mNumFramesSkipped;
int64_t mNumFramesLost;
int64_t mNumClientOwnedBuffers;
+ bool mNoMoreFramesToRead;
List<MediaBuffer * > mBuffersReceived;
diff --git a/services/camera/libcameraservice/Android.mk b/services/camera/libcameraservice/Android.mk
index 16c7f2d..f1cdea3 100644
--- a/services/camera/libcameraservice/Android.mk
+++ b/services/camera/libcameraservice/Android.mk
@@ -54,7 +54,8 @@
gui/RingBufferConsumer.cpp \
utils/CameraTraces.cpp \
utils/AutoConditionLock.cpp \
- utils/TagMonitor.cpp
+ utils/TagMonitor.cpp \
+ utils/LatencyHistogram.cpp
LOCAL_SHARED_LIBRARIES:= \
libui \
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index d6ed3ff..a11f4e2 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -613,6 +613,11 @@
}
write(fd, lines.string(), lines.size());
+ if (mRequestThread != NULL) {
+ mRequestThread->dumpCaptureRequestLatency(fd,
+ " ProcessCaptureRequest latency histogram:");
+ }
+
{
lines = String8(" Last request sent:\n");
write(fd, lines.string(), lines.size());
@@ -3426,7 +3431,8 @@
mCurrentPreCaptureTriggerId(0),
mRepeatingLastFrameNumber(
hardware::camera2::ICameraDeviceUser::NO_IN_FLIGHT_REPEATING_FRAMES),
- mPrepareVideoStream(false) {
+ mPrepareVideoStream(false),
+ mRequestLatency(kRequestLatencyBinSize) {
mStatusId = statusTracker->addComponent();
}
@@ -3651,6 +3657,9 @@
// The exit from any possible waits
mDoPauseSignal.signal();
mRequestSignal.signal();
+
+ mRequestLatency.log("ProcessCaptureRequest latency histogram");
+ mRequestLatency.reset();
}
void Camera3Device::RequestThread::checkAndStopRepeatingRequest() {
@@ -3867,11 +3876,14 @@
mNextRequests.size());
bool submitRequestSuccess = false;
+ nsecs_t tRequestStart = systemTime(SYSTEM_TIME_MONOTONIC);
if (mInterface->supportBatchRequest()) {
submitRequestSuccess = sendRequestsBatch();
} else {
submitRequestSuccess = sendRequestsOneByOne();
}
+ nsecs_t tRequestEnd = systemTime(SYSTEM_TIME_MONOTONIC);
+ mRequestLatency.add(tRequestStart, tRequestEnd);
if (useFlushLock) {
mFlushLock.unlock();
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index 1ca6811..bfb58c6 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -40,6 +40,7 @@
#include "device3/StatusTracker.h"
#include "device3/Camera3BufferManager.h"
#include "utils/TagMonitor.h"
+#include "utils/LatencyHistogram.h"
#include <camera_metadata_hidden.h>
/**
@@ -699,6 +700,11 @@
*/
bool isStreamPending(sp<camera3::Camera3StreamInterface>& stream);
+ // dump processCaptureRequest latency
+ void dumpCaptureRequestLatency(int fd, const char* name) {
+ mRequestLatency.dump(fd, name);
+ }
+
protected:
virtual bool threadLoop();
@@ -820,6 +826,9 @@
// Flag indicating if we should prepare video stream for video requests.
bool mPrepareVideoStream;
+
+ static const int32_t kRequestLatencyBinSize = 40; // in ms
+ CameraLatencyHistogram mRequestLatency;
};
sp<RequestThread> mRequestThread;
diff --git a/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp b/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp
index cb39244..7ad2300 100644
--- a/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp
+++ b/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp
@@ -85,6 +85,8 @@
lines.appendFormat(" Total buffers: %zu, currently dequeued: %zu\n",
mTotalBufferCount, mHandoutTotalBufferCount);
write(fd, lines.string(), lines.size());
+
+ Camera3Stream::dump(fd, args);
}
status_t Camera3IOStreamBase::configureQueueLocked() {
diff --git a/services/camera/libcameraservice/device3/Camera3OutputStream.cpp b/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
index ec0f508..e15aa43 100644
--- a/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
@@ -43,7 +43,8 @@
mTraceFirstBuffer(true),
mUseBufferManager(false),
mTimestampOffset(timestampOffset),
- mConsumerUsage(0) {
+ mConsumerUsage(0),
+ mDequeueBufferLatency(kDequeueLatencyBinSize) {
if (mConsumer == NULL) {
ALOGE("%s: Consumer is NULL!", __FUNCTION__);
@@ -68,7 +69,8 @@
mUseMonoTimestamp(false),
mUseBufferManager(false),
mTimestampOffset(timestampOffset),
- mConsumerUsage(0) {
+ mConsumerUsage(0),
+ mDequeueBufferLatency(kDequeueLatencyBinSize) {
if (format != HAL_PIXEL_FORMAT_BLOB && format != HAL_PIXEL_FORMAT_RAW_OPAQUE) {
ALOGE("%s: Bad format for size-only stream: %d", __FUNCTION__,
@@ -97,7 +99,8 @@
mTraceFirstBuffer(true),
mUseBufferManager(false),
mTimestampOffset(timestampOffset),
- mConsumerUsage(consumerUsage) {
+ mConsumerUsage(consumerUsage),
+ mDequeueBufferLatency(kDequeueLatencyBinSize) {
// Deferred consumer only support preview surface format now.
if (format != HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED) {
ALOGE("%s: Deferred consumer only supports IMPLEMENTATION_DEFINED format now!",
@@ -134,7 +137,8 @@
mUseMonoTimestamp(false),
mUseBufferManager(false),
mTimestampOffset(timestampOffset),
- mConsumerUsage(consumerUsage) {
+ mConsumerUsage(consumerUsage),
+ mDequeueBufferLatency(kDequeueLatencyBinSize) {
if (setId > CAMERA3_STREAM_SET_ID_INVALID) {
mBufferReleasedListener = new BufferReleasedListener(this);
@@ -290,6 +294,9 @@
write(fd, lines.string(), lines.size());
Camera3IOStreamBase::dump(fd, args);
+
+ mDequeueBufferLatency.dump(fd,
+ " DequeueBuffer latency histogram:");
}
status_t Camera3OutputStream::setTransform(int transform) {
@@ -529,7 +536,11 @@
sp<ANativeWindow> currentConsumer = mConsumer;
mLock.unlock();
+ nsecs_t dequeueStart = systemTime(SYSTEM_TIME_MONOTONIC);
res = currentConsumer->dequeueBuffer(currentConsumer.get(), anb, fenceFd);
+ nsecs_t dequeueEnd = systemTime(SYSTEM_TIME_MONOTONIC);
+ mDequeueBufferLatency.add(dequeueStart, dequeueEnd);
+
mLock.lock();
if (res != OK) {
ALOGE("%s: Stream %d: Can't dequeue next output buffer: %s (%d)",
@@ -611,6 +622,9 @@
mState = (mState == STATE_IN_RECONFIG) ? STATE_IN_CONFIG
: STATE_CONSTRUCTED;
+
+ mDequeueBufferLatency.log("Stream %d dequeueBuffer latency histogram", mId);
+ mDequeueBufferLatency.reset();
return OK;
}
diff --git a/services/camera/libcameraservice/device3/Camera3OutputStream.h b/services/camera/libcameraservice/device3/Camera3OutputStream.h
index 98ffb73..97aa7d4 100644
--- a/services/camera/libcameraservice/device3/Camera3OutputStream.h
+++ b/services/camera/libcameraservice/device3/Camera3OutputStream.h
@@ -21,6 +21,7 @@
#include <gui/IProducerListener.h>
#include <gui/Surface.h>
+#include "utils/LatencyHistogram.h"
#include "Camera3Stream.h"
#include "Camera3IOStreamBase.h"
#include "Camera3OutputStreamInterface.h"
@@ -269,6 +270,9 @@
void onBuffersRemovedLocked(const std::vector<sp<GraphicBuffer>>&);
status_t detachBufferLocked(sp<GraphicBuffer>* buffer, int* fenceFd);
+ static const int32_t kDequeueLatencyBinSize = 5; // in ms
+ CameraLatencyHistogram mDequeueBufferLatency;
+
}; // class Camera3OutputStream
} // namespace camera3
diff --git a/services/camera/libcameraservice/device3/Camera3Stream.cpp b/services/camera/libcameraservice/device3/Camera3Stream.cpp
index b45ef77..ba352c4 100644
--- a/services/camera/libcameraservice/device3/Camera3Stream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Stream.cpp
@@ -60,7 +60,8 @@
mOldMaxBuffers(0),
mPrepared(false),
mPreparedBufferIdx(0),
- mLastMaxCount(Camera3StreamInterface::ALLOCATE_PIPELINE_MAX) {
+ mLastMaxCount(Camera3StreamInterface::ALLOCATE_PIPELINE_MAX),
+ mBufferLimitLatency(kBufferLimitLatencyBinSize) {
camera3_stream::stream_type = type;
camera3_stream::width = width;
@@ -459,8 +460,11 @@
// Wait for new buffer returned back if we are running into the limit.
if (getHandoutOutputBufferCountLocked() == camera3_stream::max_buffers) {
ALOGV("%s: Already dequeued max output buffers (%d), wait for next returned one.",
- __FUNCTION__, camera3_stream::max_buffers);
+ __FUNCTION__, camera3_stream::max_buffers);
+ nsecs_t waitStart = systemTime(SYSTEM_TIME_MONOTONIC);
res = mOutputBufferReturnedSignal.waitRelative(mLock, kWaitForBufferDuration);
+ nsecs_t waitEnd = systemTime(SYSTEM_TIME_MONOTONIC);
+ mBufferLimitLatency.add(waitStart, waitEnd);
if (res != OK) {
if (res == TIMED_OUT) {
ALOGE("%s: wait for output buffer return timed out after %lldms (max_buffers %d)",
@@ -655,6 +659,9 @@
ALOGV("%s: Stream %d: Disconnecting...", __FUNCTION__, mId);
status_t res = disconnectLocked();
+ mBufferLimitLatency.log("Stream %d latency histogram for wait on max_buffers", mId);
+ mBufferLimitLatency.reset();
+
if (res == -ENOTCONN) {
// "Already disconnected" -- not an error
return OK;
@@ -663,6 +670,13 @@
}
}
+void Camera3Stream::dump(int fd, const Vector<String16> &args) const
+{
+ (void)args;
+ mBufferLimitLatency.dump(fd,
+ " Latency histogram for wait on max_buffers");
+}
+
status_t Camera3Stream::getBufferLocked(camera3_stream_buffer *,
const std::vector<size_t>&) {
ALOGE("%s: This type of stream does not support output", __FUNCTION__);
diff --git a/services/camera/libcameraservice/device3/Camera3Stream.h b/services/camera/libcameraservice/device3/Camera3Stream.h
index 9cdc1b3..b5a9c5d 100644
--- a/services/camera/libcameraservice/device3/Camera3Stream.h
+++ b/services/camera/libcameraservice/device3/Camera3Stream.h
@@ -25,6 +25,7 @@
#include "hardware/camera3.h"
+#include "utils/LatencyHistogram.h"
#include "Camera3StreamBufferListener.h"
#include "Camera3StreamInterface.h"
@@ -349,7 +350,7 @@
/**
* Debug dump of the stream's state.
*/
- virtual void dump(int fd, const Vector<String16> &args) const = 0;
+ virtual void dump(int fd, const Vector<String16> &args) const;
/**
* Add a camera3 buffer listener. Adding the same listener twice has
@@ -502,6 +503,10 @@
// Outstanding buffers dequeued from the stream's buffer queue.
List<buffer_handle_t> mOutstandingBuffers;
+ // Latency histogram of the wait time for handout buffer count to drop below
+ // max_buffers.
+ static const int32_t kBufferLimitLatencyBinSize = 33; //in ms
+ CameraLatencyHistogram mBufferLimitLatency;
}; // class Camera3Stream
}; // namespace camera3
diff --git a/services/camera/libcameraservice/utils/LatencyHistogram.cpp b/services/camera/libcameraservice/utils/LatencyHistogram.cpp
new file mode 100644
index 0000000..538bb6e
--- /dev/null
+++ b/services/camera/libcameraservice/utils/LatencyHistogram.cpp
@@ -0,0 +1,106 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "CameraLatencyHistogram"
+#include <inttypes.h>
+#include <utils/Log.h>
+#include <utils/String8.h>
+
+#include "LatencyHistogram.h"
+
+namespace android {
+
+CameraLatencyHistogram::CameraLatencyHistogram(int32_t binSizeMs, int32_t binCount) :
+ mBinSizeMs(binSizeMs),
+ mBinCount(binCount),
+ mBins(binCount),
+ mTotalCount(0) {
+}
+
+void CameraLatencyHistogram::add(nsecs_t start, nsecs_t end) {
+ nsecs_t duration = end - start;
+ int32_t durationMs = static_cast<int32_t>(duration / 1000000LL);
+ int32_t binIndex = durationMs / mBinSizeMs;
+
+ if (binIndex < 0) {
+ binIndex = 0;
+ } else if (binIndex >= mBinCount) {
+ binIndex = mBinCount-1;
+ }
+
+ mBins[binIndex]++;
+ mTotalCount++;
+}
+
+void CameraLatencyHistogram::reset() {
+ mBins.clear();
+ mTotalCount = 0;
+}
+
+void CameraLatencyHistogram::dump(int fd, const char* name) const {
+ if (mTotalCount == 0) {
+ return;
+ }
+
+ String8 lines;
+ lines.appendFormat("%s (%" PRId64 ") samples\n", name, mTotalCount);
+
+ String8 lineBins, lineBinCounts;
+ formatHistogramText(lineBins, lineBinCounts);
+
+ lineBins.append("\n");
+ lineBinCounts.append("\n");
+ lines.append(lineBins);
+ lines.append(lineBinCounts);
+
+ write(fd, lines.string(), lines.size());
+}
+
+void CameraLatencyHistogram::log(const char* fmt, ...) {
+ if (mTotalCount == 0) {
+ return;
+ }
+
+ va_list args;
+ va_start(args, fmt);
+ String8 histogramName = String8::formatV(fmt, args);
+ ALOGI("%s (%" PRId64 ") samples:", histogramName.string(), mTotalCount);
+ va_end(args);
+
+ String8 lineBins, lineBinCounts;
+ formatHistogramText(lineBins, lineBinCounts);
+
+ ALOGI("%s", lineBins.c_str());
+ ALOGI("%s", lineBinCounts.c_str());
+}
+
+void CameraLatencyHistogram::formatHistogramText(
+ String8& lineBins, String8& lineBinCounts) const {
+ lineBins = " ";
+ lineBinCounts = " ";
+
+ for (int32_t i = 0; i < mBinCount; i++) {
+ if (i == mBinCount - 1) {
+ lineBins.append(" inf (max ms)");
+ } else {
+ lineBins.appendFormat("%7d", mBinSizeMs*(i+1));
+ }
+ lineBinCounts.appendFormat(" %02.2f", 100.0*mBins[i]/mTotalCount);
+ }
+ lineBinCounts.append(" (%)");
+}
+
+}; //namespace android
diff --git a/services/camera/libcameraservice/utils/LatencyHistogram.h b/services/camera/libcameraservice/utils/LatencyHistogram.h
new file mode 100644
index 0000000..bfd9b1b
--- /dev/null
+++ b/services/camera/libcameraservice/utils/LatencyHistogram.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_SERVERS_CAMERA_LATENCY_HISTOGRAM_H_
+#define ANDROID_SERVERS_CAMERA_LATENCY_HISTOGRAM_H_
+
+#include <vector>
+
+#include <utils/Timers.h>
+#include <utils/Mutex.h>
+
+namespace android {
+
+// Histogram for camera latency characteristic
+class CameraLatencyHistogram {
+public:
+ CameraLatencyHistogram() = delete;
+ CameraLatencyHistogram(int32_t binSizeMs, int32_t binCount=10);
+ void add(nsecs_t start, nsecs_t end);
+ void reset();
+
+ void dump(int fd, const char* name) const;
+ void log(const char* format, ...);
+private:
+ int32_t mBinSizeMs;
+ int32_t mBinCount;
+ std::vector<int64_t> mBins;
+ uint64_t mTotalCount;
+
+ void formatHistogramText(String8& lineBins, String8& lineBinCounts) const;
+}; // class CameraLatencyHistogram
+
+}; // namespace android
+
+#endif // ANDROID_SERVERS_CAMERA_LATENCY_HISTOGRAM_H_
diff --git a/services/oboeservice/AAudioEndpointManager.cpp b/services/oboeservice/AAudioEndpointManager.cpp
index 3dc1feb..5c6825d 100644
--- a/services/oboeservice/AAudioEndpointManager.cpp
+++ b/services/oboeservice/AAudioEndpointManager.cpp
@@ -52,15 +52,17 @@
assert(false); // There are only two possible directions.
break;
}
- ALOGD("AAudioEndpointManager::openEndpoint(), found %p for device = %d, dir = %d",
- endpoint, deviceId, (int)direction);
// If we can't find an existing one then open a new one.
- if (endpoint == nullptr) {
+ if (endpoint != nullptr) {
+ ALOGD("AAudioEndpointManager::openEndpoint(), found %p for device = %d, dir = %d",
+ endpoint, deviceId, (int)direction);
+
+ } else {
if (direction == AAUDIO_DIRECTION_INPUT) {
AAudioServiceEndpointCapture *capture = new AAudioServiceEndpointCapture(audioService);
if (capture->open(deviceId) != AAUDIO_OK) {
- ALOGE("AAudioEndpointManager::openEndpoint(), open failed");
+ ALOGE("AAudioEndpointManager::openEndpoint(), open input failed");
delete capture;
} else {
mInputs[deviceId] = capture;
@@ -69,17 +71,20 @@
} else if (direction == AAUDIO_DIRECTION_OUTPUT) {
AAudioServiceEndpointPlay *player = new AAudioServiceEndpointPlay(audioService);
if (player->open(deviceId) != AAUDIO_OK) {
- ALOGE("AAudioEndpointManager::openEndpoint(), open failed");
+ ALOGE("AAudioEndpointManager::openEndpoint(), open output failed");
delete player;
} else {
mOutputs[deviceId] = player;
endpoint = player;
}
}
-
+ ALOGD("AAudioEndpointManager::openEndpoint(), created %p for device = %d, dir = %d",
+ endpoint, deviceId, (int)direction);
}
if (endpoint != nullptr) {
+ ALOGD("AAudioEndpointManager::openEndpoint(), sampleRate = %d, framesPerBurst = %d",
+ endpoint->getSampleRate(), endpoint->getFramesPerBurst());
// Increment the reference count under this lock.
endpoint->setReferenceCount(endpoint->getReferenceCount() + 1);
}
@@ -95,9 +100,15 @@
// Decrement the reference count under this lock.
int32_t newRefCount = serviceEndpoint->getReferenceCount() - 1;
serviceEndpoint->setReferenceCount(newRefCount);
+ ALOGD("AAudioEndpointManager::closeEndpoint(%p) newRefCount = %d",
+ serviceEndpoint, newRefCount);
+
+ // If no longer in use then close and delete it.
if (newRefCount <= 0) {
aaudio_direction_t direction = serviceEndpoint->getDirection();
- int32_t deviceId = serviceEndpoint->getDeviceId();
+ // Track endpoints based on requested deviceId because UNSPECIFIED
+ // can change to a specific device after opening.
+ int32_t deviceId = serviceEndpoint->getRequestedDeviceId();
switch (direction) {
case AAUDIO_DIRECTION_INPUT:
@@ -109,6 +120,8 @@
}
serviceEndpoint->close();
+ ALOGD("AAudioEndpointManager::closeEndpoint() delete %p for device %d, dir = %d",
+ serviceEndpoint, deviceId, (int)direction);
delete serviceEndpoint;
}
}
diff --git a/services/oboeservice/AAudioEndpointManager.h b/services/oboeservice/AAudioEndpointManager.h
index db1103d..899ea35 100644
--- a/services/oboeservice/AAudioEndpointManager.h
+++ b/services/oboeservice/AAudioEndpointManager.h
@@ -35,7 +35,7 @@
/**
* Find a service endpoint for the given deviceId and direction.
- * If an endpoint does not already exist then it will try to create one.
+ * If an endpoint does not already exist then try to create one.
*
* @param deviceId
* @param direction
diff --git a/services/oboeservice/AAudioService.cpp b/services/oboeservice/AAudioService.cpp
index c9b9065..b0e0a74 100644
--- a/services/oboeservice/AAudioService.cpp
+++ b/services/oboeservice/AAudioService.cpp
@@ -89,7 +89,7 @@
return result;
} else {
aaudio_handle_t handle = mHandleTracker.put(AAUDIO_HANDLE_TYPE_STREAM, serviceStream);
- ALOGV("AAudioService::openStream(): handle = 0x%08X", handle);
+ ALOGD("AAudioService::openStream(): handle = 0x%08X", handle);
if (handle < 0) {
ALOGE("AAudioService::openStream(): handle table full");
delete serviceStream;
diff --git a/services/oboeservice/AAudioServiceEndpoint.cpp b/services/oboeservice/AAudioServiceEndpoint.cpp
index d8ae284..cc2cb44 100644
--- a/services/oboeservice/AAudioServiceEndpoint.cpp
+++ b/services/oboeservice/AAudioServiceEndpoint.cpp
@@ -46,6 +46,7 @@
// Set up an EXCLUSIVE MMAP stream that will be shared.
aaudio_result_t AAudioServiceEndpoint::open(int32_t deviceId) {
+ mRequestedDeviceId = deviceId;
mStreamInternal = getStreamInternal();
AudioStreamBuilder builder;
diff --git a/services/oboeservice/AAudioServiceEndpoint.h b/services/oboeservice/AAudioServiceEndpoint.h
index 50bf049..c271dbd 100644
--- a/services/oboeservice/AAudioServiceEndpoint.h
+++ b/services/oboeservice/AAudioServiceEndpoint.h
@@ -48,6 +48,7 @@
aaudio_result_t stopStream(AAudioServiceStreamShared *sharedStream);
aaudio_result_t close();
+ int32_t getRequestedDeviceId() const { return mRequestedDeviceId; }
int32_t getDeviceId() const { return mStreamInternal->getDeviceId(); }
aaudio_direction_t getDirection() const { return mStreamInternal->getDirection(); }
@@ -81,6 +82,7 @@
AudioStreamInternal *mStreamInternal = nullptr;
int32_t mReferenceCount = 0;
+ int32_t mRequestedDeviceId = 0;
};
} /* namespace aaudio */
diff --git a/services/oboeservice/AAudioServiceEndpointPlay.h b/services/oboeservice/AAudioServiceEndpointPlay.h
index b977960..89935ae 100644
--- a/services/oboeservice/AAudioServiceEndpointPlay.h
+++ b/services/oboeservice/AAudioServiceEndpointPlay.h
@@ -32,6 +32,9 @@
namespace aaudio {
+/**
+ * Contains a mixer and a stream for writing the result of the mix.
+ */
class AAudioServiceEndpointPlay : public AAudioServiceEndpoint {
public:
explicit AAudioServiceEndpointPlay(android::AAudioService &audioService);
diff --git a/services/oboeservice/AAudioServiceStreamBase.cpp b/services/oboeservice/AAudioServiceStreamBase.cpp
index 8f0abc2..ee0e7ed 100644
--- a/services/oboeservice/AAudioServiceStreamBase.cpp
+++ b/services/oboeservice/AAudioServiceStreamBase.cpp
@@ -71,12 +71,11 @@
}
aaudio_result_t AAudioServiceStreamBase::pause() {
-
sendCurrentTimestamp();
mThreadEnabled.store(false);
aaudio_result_t result = mAAudioThread.stop();
if (result != AAUDIO_OK) {
- processError();
+ processFatalError();
return result;
}
sendServiceEvent(AAUDIO_SERVICE_EVENT_PAUSED);
@@ -90,7 +89,7 @@
mThreadEnabled.store(false);
aaudio_result_t result = mAAudioThread.stop();
if (result != AAUDIO_OK) {
- processError();
+ processFatalError();
return result;
}
sendServiceEvent(AAUDIO_SERVICE_EVENT_STOPPED);
@@ -126,7 +125,7 @@
ALOGD("AAudioServiceStreamBase::run() exiting ----------------");
}
-void AAudioServiceStreamBase::processError() {
+void AAudioServiceStreamBase::processFatalError() {
sendServiceEvent(AAUDIO_SERVICE_EVENT_DISCONNECTED);
}
diff --git a/services/oboeservice/AAudioServiceStreamBase.h b/services/oboeservice/AAudioServiceStreamBase.h
index ee52c39..46ceeae 100644
--- a/services/oboeservice/AAudioServiceStreamBase.h
+++ b/services/oboeservice/AAudioServiceStreamBase.h
@@ -111,7 +111,7 @@
void run() override; // to implement Runnable
- void processError();
+ void processFatalError();
protected:
aaudio_result_t writeUpMessageQueue(AAudioServiceMessage *command);
@@ -122,16 +122,16 @@
virtual aaudio_result_t getDownDataDescription(AudioEndpointParcelable &parcelable) = 0;
- aaudio_stream_state_t mState = AAUDIO_STREAM_STATE_UNINITIALIZED;
+ aaudio_stream_state_t mState = AAUDIO_STREAM_STATE_UNINITIALIZED;
pid_t mRegisteredClientThread = ILLEGAL_THREAD_ID;
SharedRingBuffer* mUpMessageQueue;
std::mutex mLockUpMessageQueue;
- AAudioThread mAAudioThread;
+ AAudioThread mAAudioThread;
// This is used by one thread to tell another thread to exit. So it must be atomic.
- std::atomic<bool> mThreadEnabled;
+ std::atomic<bool> mThreadEnabled;
aaudio_format_t mAudioFormat = AAUDIO_FORMAT_UNSPECIFIED;
int32_t mFramesPerBurst = 0;
diff --git a/services/oboeservice/AAudioServiceStreamMMAP.cpp b/services/oboeservice/AAudioServiceStreamMMAP.cpp
index 97b9937..2f3ec27 100644
--- a/services/oboeservice/AAudioServiceStreamMMAP.cpp
+++ b/services/oboeservice/AAudioServiceStreamMMAP.cpp
@@ -138,15 +138,19 @@
return AAUDIO_ERROR_UNAVAILABLE;
}
+ if (deviceId == AAUDIO_UNSPECIFIED) {
+ ALOGW("AAudioServiceStreamMMAP::open() - openMmapStream() failed to set deviceId");
+ }
+
// Create MMAP/NOIRQ buffer.
int32_t minSizeFrames = configurationInput.getBufferCapacity();
- if (minSizeFrames == 0) { // zero will get rejected
+ if (minSizeFrames <= 0) { // zero will get rejected
minSizeFrames = AAUDIO_BUFFER_CAPACITY_MIN;
}
status = mMmapStream->createMmapBuffer(minSizeFrames, &mMmapBufferinfo);
if (status != OK) {
- ALOGE("%s: createMmapBuffer() returned status %d, return AAUDIO_ERROR_UNAVAILABLE",
- __FILE__, status);
+ ALOGE("AAudioServiceStreamMMAP::open() - createMmapBuffer() returned status %d",
+ status);
return AAUDIO_ERROR_UNAVAILABLE;
} else {
ALOGD("createMmapBuffer status %d shared_address = %p buffer_size %d burst_size %d",
@@ -181,6 +185,9 @@
ALOGD("AAudioServiceStreamMMAP::open() original burst = %d, minMicros = %d, final burst = %d\n",
mMmapBufferinfo.burst_size_frames, burstMinMicros, mFramesPerBurst);
+ ALOGD("AAudioServiceStreamMMAP::open() actual rate = %d, channels = %d, deviceId = %d\n",
+ mSampleRate, mSamplesPerFrame, deviceId);
+
// Fill in AAudioStreamConfiguration
configurationOutput.setSampleRate(mSampleRate);
configurationOutput.setSamplesPerFrame(mSamplesPerFrame);
@@ -199,7 +206,7 @@
status_t status = mMmapStream->start(mMmapClient, &mPortHandle);
if (status != OK) {
ALOGE("AAudioServiceStreamMMAP::start() mMmapStream->start() returned %d", status);
- processError();
+ processFatalError();
result = AAudioConvert_androidToAAudioResult(status);
} else {
result = AAudioServiceStreamBase::start();
@@ -234,8 +241,6 @@
aaudio_result_t AAudioServiceStreamMMAP::flush() {
if (mMmapStream == nullptr) return AAUDIO_ERROR_NULL;
// TODO how do we flush an MMAP/NOIRQ buffer? sync pointers?
- sendServiceEvent(AAUDIO_SERVICE_EVENT_FLUSHED);
- mState = AAUDIO_STREAM_STATE_FLUSHED;
return AAudioServiceStreamBase::flush();;
}
@@ -244,13 +249,13 @@
int64_t *timeNanos) {
struct audio_mmap_position position;
if (mMmapStream == nullptr) {
- processError();
+ processFatalError();
return AAUDIO_ERROR_NULL;
}
status_t status = mMmapStream->getMmapPosition(&position);
if (status != OK) {
ALOGE("sendCurrentTimestamp(): getMmapPosition() returned %d", status);
- processError();
+ processFatalError();
return AAudioConvert_androidToAAudioResult(status);
} else {
mFramesRead.update32(position.position_frames);
@@ -295,4 +300,4 @@
parcelable.mDownDataQueueParcelable.setFramesPerBurst(mFramesPerBurst);
parcelable.mDownDataQueueParcelable.setCapacityInFrames(mCapacityInFrames);
return AAUDIO_OK;
-}
\ No newline at end of file
+}
diff --git a/services/oboeservice/AAudioServiceStreamShared.cpp b/services/oboeservice/AAudioServiceStreamShared.cpp
index 494b18e..f246fc02 100644
--- a/services/oboeservice/AAudioServiceStreamShared.cpp
+++ b/services/oboeservice/AAudioServiceStreamShared.cpp
@@ -34,8 +34,10 @@
using namespace android;
using namespace aaudio;
-#define MIN_BURSTS_PER_BUFFER 2
-#define MAX_BURSTS_PER_BUFFER 32
+#define MIN_BURSTS_PER_BUFFER 2
+#define DEFAULT_BURSTS_PER_BUFFER 16
+// This is an arbitrary range. TODO review.
+#define MAX_FRAMES_PER_BUFFER (32 * 1024)
AAudioServiceStreamShared::AAudioServiceStreamShared(AAudioService &audioService)
: mAudioService(audioService)
@@ -46,12 +48,58 @@
close();
}
+int32_t AAudioServiceStreamShared::calculateBufferCapacity(int32_t requestedCapacityFrames,
+ int32_t framesPerBurst) {
+
+ if (requestedCapacityFrames > MAX_FRAMES_PER_BUFFER) {
+ ALOGE("AAudioServiceStreamShared::open(), requested capacity %d > max %d",
+ requestedCapacityFrames, MAX_FRAMES_PER_BUFFER);
+ return AAUDIO_ERROR_OUT_OF_RANGE;
+ }
+
+ // Determine how many bursts will fit in the buffer.
+ int32_t numBursts;
+ if (requestedCapacityFrames == AAUDIO_UNSPECIFIED) {
+ // Use fewer bursts if default is too many.
+ if ((DEFAULT_BURSTS_PER_BUFFER * framesPerBurst) > MAX_FRAMES_PER_BUFFER) {
+ numBursts = MAX_FRAMES_PER_BUFFER / framesPerBurst;
+ } else {
+ numBursts = DEFAULT_BURSTS_PER_BUFFER;
+ }
+ } else {
+ // round up to nearest burst boundary
+ numBursts = (requestedCapacityFrames + framesPerBurst - 1) / framesPerBurst;
+ }
+
+ // Clip to bare minimum.
+ if (numBursts < MIN_BURSTS_PER_BUFFER) {
+ numBursts = MIN_BURSTS_PER_BUFFER;
+ }
+ // Check for numeric overflow.
+ if (numBursts > 0x8000 || framesPerBurst > 0x8000) {
+ ALOGE("AAudioServiceStreamShared::open(), numeric overflow, capacity = %d * %d",
+ numBursts, framesPerBurst);
+ return AAUDIO_ERROR_OUT_OF_RANGE;
+ }
+ int32_t capacityInFrames = numBursts * framesPerBurst;
+
+ // Final sanity check.
+ if (capacityInFrames > MAX_FRAMES_PER_BUFFER) {
+ ALOGE("AAudioServiceStreamShared::open(), calculated capacity %d > max %d",
+ capacityInFrames, MAX_FRAMES_PER_BUFFER);
+ return AAUDIO_ERROR_OUT_OF_RANGE;
+ }
+ ALOGD("AAudioServiceStreamShared::open(), requested capacity = %d frames, actual = %d",
+ requestedCapacityFrames, capacityInFrames);
+ return capacityInFrames;
+}
+
aaudio_result_t AAudioServiceStreamShared::open(const aaudio::AAudioStreamRequest &request,
aaudio::AAudioStreamConfiguration &configurationOutput) {
aaudio_result_t result = AAudioServiceStreamBase::open(request, configurationOutput);
if (result != AAUDIO_OK) {
- ALOGE("AAudioServiceStreamBase open returned %d", result);
+ ALOGE("AAudioServiceStreamBase open() returned %d", result);
return result;
}
@@ -72,16 +120,18 @@
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;
+ result = AAUDIO_ERROR_INVALID_FORMAT;
+ goto error;
}
mSampleRate = configurationInput.getSampleRate();
if (mSampleRate == AAUDIO_UNSPECIFIED) {
mSampleRate = mServiceEndpoint->getSampleRate();
} else if (mSampleRate != mServiceEndpoint->getSampleRate()) {
- ALOGE("AAudioServiceStreamShared::open(), mAudioFormat = %d, need %d",
+ ALOGE("AAudioServiceStreamShared::open(), mSampleRate = %d, need %d",
mSampleRate, mServiceEndpoint->getSampleRate());
- return AAUDIO_ERROR_INVALID_RATE;
+ result = AAUDIO_ERROR_INVALID_RATE;
+ goto error;
}
mSamplesPerFrame = configurationInput.getSamplesPerFrame();
@@ -90,37 +140,51 @@
} else if (mSamplesPerFrame != mServiceEndpoint->getSamplesPerFrame()) {
ALOGE("AAudioServiceStreamShared::open(), mSamplesPerFrame = %d, need %d",
mSamplesPerFrame, mServiceEndpoint->getSamplesPerFrame());
- return AAUDIO_ERROR_OUT_OF_RANGE;
+ result = AAUDIO_ERROR_OUT_OF_RANGE;
+ goto error;
}
- // Determine this stream's shared memory buffer capacity.
mFramesPerBurst = mServiceEndpoint->getFramesPerBurst();
- int32_t minCapacityFrames = configurationInput.getBufferCapacity();
- 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;
- }
+ ALOGD("AAudioServiceStreamShared::open(), mSampleRate = %d, mFramesPerBurst = %d",
+ mSampleRate, mFramesPerBurst);
+
+ mCapacityInFrames = calculateBufferCapacity(configurationInput.getBufferCapacity(),
+ mFramesPerBurst);
+ if (mCapacityInFrames < 0) {
+ result = mCapacityInFrames; // negative error code
+ mCapacityInFrames = 0;
+ goto error;
}
- mCapacityInFrames = numBursts * mFramesPerBurst;
- ALOGD("AAudioServiceStreamShared::open(), mCapacityInFrames = %d", mCapacityInFrames);
// Create audio data shared memory buffer for client.
mAudioDataQueue = new SharedRingBuffer();
- mAudioDataQueue->allocate(calculateBytesPerFrame(), mCapacityInFrames);
+ result = mAudioDataQueue->allocate(calculateBytesPerFrame(), mCapacityInFrames);
+ if (result != AAUDIO_OK) {
+ ALOGE("AAudioServiceStreamShared::open(), could not allocate FIFO with %d frames",
+ mCapacityInFrames);
+ result = AAUDIO_ERROR_NO_MEMORY;
+ goto error;
+ }
+
+ ALOGD("AAudioServiceStreamShared::open() actual rate = %d, channels = %d, deviceId = %d",
+ mSampleRate, mSamplesPerFrame, mServiceEndpoint->getDeviceId());
// Fill in configuration for client.
configurationOutput.setSampleRate(mSampleRate);
configurationOutput.setSamplesPerFrame(mSamplesPerFrame);
configurationOutput.setAudioFormat(mAudioFormat);
- configurationOutput.setDeviceId(deviceId);
+ configurationOutput.setDeviceId(mServiceEndpoint->getDeviceId());
- mServiceEndpoint->registerStream(this);
+ result = mServiceEndpoint->registerStream(this);
+ if (result != AAUDIO_OK) {
+ goto error;
+ }
return AAUDIO_OK;
+
+error:
+ close();
+ return result;
}
/**
@@ -137,11 +201,11 @@
aaudio_result_t result = endpoint->startStream(this);
if (result != AAUDIO_OK) {
ALOGE("AAudioServiceStreamShared::start() mServiceEndpoint returned %d", result);
- processError();
+ processFatalError();
} else {
result = AAudioServiceStreamBase::start();
}
- return AAUDIO_OK;
+ return result;
}
/**
@@ -154,11 +218,10 @@
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::pause() mServiceEndpoint returned %d", result);
- processError();
+ processFatalError();
}
return AAudioServiceStreamBase::pause();
}
@@ -168,11 +231,10 @@
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();
+ processFatalError();
}
return AAudioServiceStreamBase::stop();
}
@@ -183,9 +245,17 @@
* An AAUDIO_SERVICE_EVENT_FLUSHED will be sent to the client when complete.
*/
aaudio_result_t AAudioServiceStreamShared::flush() {
- // TODO make sure we are paused
- // TODO actually flush the data
- return AAudioServiceStreamBase::flush() ;
+ AAudioServiceEndpoint *endpoint = mServiceEndpoint;
+ if (endpoint == nullptr) {
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+ if (mState != AAUDIO_STREAM_STATE_PAUSED) {
+ ALOGE("AAudioServiceStreamShared::flush() stream not paused, state = %s",
+ AAudio_convertStreamStateToText(mState));
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+ // Data will get flushed when the client receives the FLUSHED event.
+ return AAudioServiceStreamBase::flush();
}
aaudio_result_t AAudioServiceStreamShared::close() {
diff --git a/services/oboeservice/AAudioServiceStreamShared.h b/services/oboeservice/AAudioServiceStreamShared.h
index dfdbbb3..35af434 100644
--- a/services/oboeservice/AAudioServiceStreamShared.h
+++ b/services/oboeservice/AAudioServiceStreamShared.h
@@ -97,6 +97,14 @@
aaudio_result_t getFreeRunningPosition(int64_t *positionFrames, int64_t *timeNanos) override;
+ /**
+ * @param requestedCapacityFrames
+ * @param framesPerBurst
+ * @return capacity or negative error
+ */
+ static int32_t calculateBufferCapacity(int32_t requestedCapacityFrames,
+ int32_t framesPerBurst);
+
private:
android::AAudioService &mAudioService;
AAudioServiceEndpoint *mServiceEndpoint = nullptr;
diff --git a/services/oboeservice/Android.mk b/services/oboeservice/Android.mk
index b447725..7f7d465 100644
--- a/services/oboeservice/Android.mk
+++ b/services/oboeservice/Android.mk
@@ -46,6 +46,7 @@
LOCAL_SHARED_LIBRARIES := \
libaaudio \
libaudioflinger \
+ libaudioclient \
libbinder \
libcutils \
libmediautils \