aaudio: improve accuracy of timestamps
Account for latency added by the AAudio service.
Fix input timestamps.
Bug: 37080396
Test: test_timestamps.cpp input_monitor.cpp
Change-Id: I1053cd21af722bb9b9371df4e5731bf4a0a57b0b
diff --git a/media/libaaudio/examples/input_monitor/src/input_monitor.cpp b/media/libaaudio/examples/input_monitor/src/input_monitor.cpp
index edf644a..2dfd0a7 100644
--- a/media/libaaudio/examples/input_monitor/src/input_monitor.cpp
+++ b/media/libaaudio/examples/input_monitor/src/input_monitor.cpp
@@ -27,9 +27,11 @@
#include "AAudioSimpleRecorder.h"
// TODO support FLOAT
-#define REQUIRED_FORMAT AAUDIO_FORMAT_PCM_I16
+#define REQUIRED_FORMAT AAUDIO_FORMAT_PCM_I16
#define MIN_FRAMES_TO_READ 48 /* arbitrary, 1 msec at 48000 Hz */
+static const int FRAMES_PER_LINE = 20000;
+
int main(int argc, const char **argv)
{
AAudioArgsParser argParser;
@@ -46,7 +48,10 @@
int32_t framesPerRead = 0;
int32_t framesToRecord = 0;
int32_t framesLeft = 0;
+ int32_t nextFrameCount = 0;
+ int32_t frameCount = 0;
int32_t xRunCount = 0;
+ int64_t previousFramePosition = -1;
int16_t *data = nullptr;
float peakLevel = 0.0;
int loopCounter = 0;
@@ -56,7 +61,7 @@
// in a buffer if we hang or crash.
setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
- printf("%s - Monitor input level using AAudio\n", argv[0]);
+ printf("%s - Monitor input level using AAudio V0.1.1\n", argv[0]);
argParser.setFormat(REQUIRED_FORMAT);
if (argParser.parseArgs(argc, argv)) {
@@ -133,6 +138,7 @@
goto finish;
}
framesLeft -= actual;
+ frameCount += actual;
// Peak finder.
for (int frameIndex = 0; frameIndex < actual; frameIndex++) {
@@ -143,9 +149,36 @@
}
// Display level as stars, eg. "******".
- if ((loopCounter++ % 10) == 0) {
+ if (frameCount > nextFrameCount) {
displayPeakLevel(peakLevel);
peakLevel = 0.0;
+ nextFrameCount += FRAMES_PER_LINE;
+ }
+
+ // Print timestamps.
+ int64_t framePosition = 0;
+ int64_t frameTime = 0;
+ aaudio_result_t timeResult;
+ timeResult = AAudioStream_getTimestamp(aaudioStream, CLOCK_MONOTONIC,
+ &framePosition, &frameTime);
+
+ if (timeResult == AAUDIO_OK) {
+ if (framePosition > (previousFramePosition + FRAMES_PER_LINE)) {
+ int64_t realTime = getNanoseconds();
+ int64_t framesRead = AAudioStream_getFramesRead(aaudioStream);
+
+ double latencyMillis = calculateLatencyMillis(framesRead, realTime,
+ framePosition, frameTime,
+ actualSampleRate);
+
+ printf("--- timestamp: result = %4d, position = %lld, at %lld nanos"
+ ", latency = %7.2f msec\n",
+ timeResult,
+ (long long) framePosition,
+ (long long) frameTime,
+ latencyMillis);
+ previousFramePosition = framePosition;
+ }
}
}
diff --git a/media/libaaudio/examples/utils/AAudioExampleUtils.h b/media/libaaudio/examples/utils/AAudioExampleUtils.h
index 66de25f..6cbcc58 100644
--- a/media/libaaudio/examples/utils/AAudioExampleUtils.h
+++ b/media/libaaudio/examples/utils/AAudioExampleUtils.h
@@ -25,7 +25,7 @@
#define NANOS_PER_MILLISECOND (NANOS_PER_MICROSECOND * 1000)
#define NANOS_PER_SECOND (NANOS_PER_MILLISECOND * 1000)
-static const char *getSharingModeText(aaudio_sharing_mode_t mode) {
+const char *getSharingModeText(aaudio_sharing_mode_t mode) {
const char *modeText = "unknown";
switch (mode) {
case AAUDIO_SHARING_MODE_EXCLUSIVE:
@@ -49,7 +49,7 @@
return (time.tv_sec * NANOS_PER_SECOND) + time.tv_nsec;
}
-void displayPeakLevel(float peakLevel) {
+static void displayPeakLevel(float peakLevel) {
printf("%5.3f ", peakLevel);
const int maxStars = 50; // arbitrary, fits on one line
int numStars = (int) (peakLevel * maxStars);
@@ -59,4 +59,24 @@
printf("\n");
}
+/**
+ * @param position1 position of hardware frame
+ * @param nanoseconds1
+ * @param position2 position of client read/write
+ * @param nanoseconds2
+ * @param sampleRate
+ * @return latency in milliseconds
+ */
+static double calculateLatencyMillis(int64_t position1, int64_t nanoseconds1,
+ int64_t position2, int64_t nanoseconds2,
+ int64_t sampleRate) {
+ int64_t deltaFrames = position2 - position1;
+ int64_t deltaTime =
+ (NANOS_PER_SECOND * deltaFrames / sampleRate);
+ int64_t timeCurrentFramePlayed = nanoseconds1 + deltaTime;
+ int64_t latencyNanos = timeCurrentFramePlayed - nanoseconds2;
+ double latencyMillis = latencyNanos / 1000000.0;
+ return latencyMillis;
+}
+
#endif // AAUDIO_EXAMPLE_UTILS_H
diff --git a/media/libaaudio/src/binding/AAudioServiceMessage.h b/media/libaaudio/src/binding/AAudioServiceMessage.h
index b4377fb..54e8001 100644
--- a/media/libaaudio/src/binding/AAudioServiceMessage.h
+++ b/media/libaaudio/src/binding/AAudioServiceMessage.h
@@ -28,7 +28,6 @@
// Used to send information about the HAL to the client.
struct AAudioMessageTimestamp {
int64_t position; // number of frames transferred so far
- int64_t deviceOffset; // add to client position to get device position
int64_t timestamp; // time when that position was reached
};
@@ -51,7 +50,8 @@
typedef struct AAudioServiceMessage_s {
enum class code : uint32_t {
NOTHING,
- TIMESTAMP,
+ TIMESTAMP_SERVICE, // when frame is read or written by the service to the client
+ TIMESTAMP_HARDWARE, // when frame is at DAC or ADC
EVENT,
};
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 4c7d0f7..259c9b9 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -68,6 +68,7 @@
, mServiceInterface(serviceInterface)
, mWakeupDelayNanos(AAudioProperty_getWakeupDelayMicros() * AAUDIO_NANOS_PER_MICROSECOND)
, mMinimumSleepNanos(AAudioProperty_getMinimumSleepMicros() * AAUDIO_NANOS_PER_MICROSECOND)
+ , mAtomicTimestamp()
{
ALOGD("AudioStreamInternal(): mWakeupDelayNanos = %d, mMinimumSleepNanos = %d",
mWakeupDelayNanos, mMinimumSleepNanos);
@@ -351,12 +352,15 @@
aaudio_result_t AudioStreamInternal::getTimestamp(clockid_t clockId,
int64_t *framePosition,
int64_t *timeNanoseconds) {
- // TODO Generate in server and pass to client. Return latest.
- int64_t time = AudioClock::getNanoseconds();
- *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;
+ // Generated in server and passed to client. Return latest.
+ if (mAtomicTimestamp.isValid()) {
+ Timestamp timestamp = mAtomicTimestamp.read();
+ *framePosition = timestamp.getPosition();
+ *timeNanoseconds = timestamp.getNanoseconds();
+ return AAUDIO_OK;
+ } else {
+ return AAUDIO_ERROR_UNAVAILABLE;
+ }
}
aaudio_result_t AudioStreamInternal::updateStateWhileWaiting() {
@@ -385,7 +389,7 @@
oldTime = nanoTime;
}
-aaudio_result_t AudioStreamInternal::onTimestampFromServer(AAudioServiceMessage *message) {
+aaudio_result_t AudioStreamInternal::onTimestampService(AAudioServiceMessage *message) {
#if LOG_TIMESTAMPS
logTimestamp(*message);
#endif
@@ -393,6 +397,12 @@
return AAUDIO_OK;
}
+aaudio_result_t AudioStreamInternal::onTimestampHardware(AAudioServiceMessage *message) {
+ Timestamp timestamp(message->timestamp.position, message->timestamp.timestamp);
+ mAtomicTimestamp.write(timestamp);
+ return AAUDIO_OK;
+}
+
aaudio_result_t AudioStreamInternal::onEventFromServer(AAudioServiceMessage *message) {
aaudio_result_t result = AAUDIO_OK;
switch (message->event.event) {
@@ -456,8 +466,12 @@
break; // no command this time, no problem
}
switch (message.what) {
- case AAudioServiceMessage::code::TIMESTAMP:
- result = onTimestampFromServer(&message);
+ case AAudioServiceMessage::code::TIMESTAMP_SERVICE:
+ result = onTimestampService(&message);
+ break;
+
+ case AAudioServiceMessage::code::TIMESTAMP_HARDWARE:
+ result = onTimestampHardware(&message);
break;
case AAudioServiceMessage::code::EVENT:
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 1b991de..607e734 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -122,7 +122,9 @@
aaudio_result_t onEventFromServer(AAudioServiceMessage *message);
- aaudio_result_t onTimestampFromServer(AAudioServiceMessage *message);
+ aaudio_result_t onTimestampService(AAudioServiceMessage *message);
+
+ aaudio_result_t onTimestampHardware(AAudioServiceMessage *message);
void logTimestamp(AAudioServiceMessage &message);
@@ -181,6 +183,11 @@
AudioEndpointParcelable mEndPointParcelable; // description of the buffers filled by service
EndpointDescriptor mEndpointDescriptor; // buffer description with resolved addresses
+
+ SimpleDoubleBuffer<Timestamp> mAtomicTimestamp;
+
+ int64_t mServiceLatencyNanos = 0;
+
};
} /* namespace aaudio */
diff --git a/media/libaaudio/src/utility/AAudioUtilities.h b/media/libaaudio/src/utility/AAudioUtilities.h
index acd319b..b0c6c94 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.h
+++ b/media/libaaudio/src/utility/AAudioUtilities.h
@@ -258,4 +258,74 @@
}
}
+
+/**
+ * Simple double buffer for a structure that can be written occasionally and read occasionally.
+ * This allows a SINGLE writer with multiple readers.
+ *
+ * It is OK if the FIFO overflows and we lose old values.
+ * It is also OK if we read an old value.
+ * Thread may return a non-atomic result if the other thread is rapidly writing
+ * new values on another core.
+ */
+template <class T>
+class SimpleDoubleBuffer {
+public:
+ SimpleDoubleBuffer()
+ : mValues()
+ , mCounter(0) {}
+
+ __attribute__((no_sanitize("integer")))
+ void write(T value) {
+ int index = mCounter.load() & 1;
+ mValues[index] = value;
+ mCounter++; // Increment AFTER updating storage, OK if it wraps.
+ }
+
+ T read() const {
+ T result;
+ int before;
+ int after;
+ int timeout = 3;
+ do {
+ // Check to see if a write occurred while were reading.
+ before = mCounter.load();
+ int index = (before & 1) ^ 1;
+ result = mValues[index];
+ after = mCounter.load();
+ } while ((after != before) && --timeout > 0);
+ return result;
+ }
+
+ /**
+ * @return true if at least one value has been written
+ */
+ bool isValid() const {
+ return mCounter.load() > 0;
+ }
+
+private:
+ T mValues[2];
+ std::atomic<int> mCounter;
+};
+
+class Timestamp {
+public:
+ Timestamp()
+ : mPosition(0)
+ , mNanoseconds(0) {}
+ Timestamp(int64_t position, int64_t nanoseconds)
+ : mPosition(position)
+ , mNanoseconds(nanoseconds) {}
+
+ int64_t getPosition() const { return mPosition; }
+
+ int64_t getNanoseconds() const { return mNanoseconds; }
+
+private:
+ // These cannot be const because we need to implement the copy assignment operator.
+ int64_t mPosition;
+ int64_t mNanoseconds;
+};
+
#endif //UTILITY_AAUDIO_UTILITIES_H
diff --git a/media/libaaudio/tests/Android.mk b/media/libaaudio/tests/Android.mk
index e4eef06..4402919 100644
--- a/media/libaaudio/tests/Android.mk
+++ b/media/libaaudio/tests/Android.mk
@@ -34,6 +34,17 @@
LOCAL_C_INCLUDES := \
$(call include-path-for, audio-utils) \
frameworks/av/media/libaaudio/include \
+ frameworks/av/media/libaaudio/src \
+ frameworks/av/media/libaaudio/examples
+LOCAL_SRC_FILES:= test_timestamps.cpp
+LOCAL_SHARED_LIBRARIES := libaaudio
+LOCAL_MODULE := test_timestamps
+include $(BUILD_NATIVE_TEST)
+
+include $(CLEAR_VARS)
+LOCAL_C_INCLUDES := \
+ $(call include-path-for, audio-utils) \
+ frameworks/av/media/libaaudio/include \
frameworks/av/media/libaaudio/src
LOCAL_SRC_FILES:= test_linear_ramp.cpp
LOCAL_SHARED_LIBRARIES := libaaudio
diff --git a/media/libaaudio/tests/test_timestamps.cpp b/media/libaaudio/tests/test_timestamps.cpp
new file mode 100644
index 0000000..d9ca391
--- /dev/null
+++ b/media/libaaudio/tests/test_timestamps.cpp
@@ -0,0 +1,188 @@
+/*
+ * 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.
+ */
+
+// Play silence and recover from dead servers or disconnected devices.
+
+#include <stdio.h>
+#include <unistd.h>
+
+#include <aaudio/AAudio.h>
+#include <aaudio/AAudioTesting.h>
+
+#include "utils/AAudioExampleUtils.h"
+
+#define DEFAULT_TIMEOUT_NANOS ((int64_t)1000000000)
+
+int main(int argc, char **argv) {
+ (void) argc;
+ (void *)argv;
+
+ aaudio_result_t result = AAUDIO_OK;
+
+ int32_t triesLeft = 3;
+ int32_t bufferCapacity;
+ int32_t framesPerBurst = 0;
+ float *buffer = nullptr;
+
+ int32_t actualChannelCount = 0;
+ int32_t actualSampleRate = 0;
+ int32_t originalBufferSize = 0;
+ int32_t requestedBufferSize = 0;
+ int32_t finalBufferSize = 0;
+ aaudio_format_t actualDataFormat = AAUDIO_FORMAT_PCM_FLOAT;
+ aaudio_sharing_mode_t actualSharingMode = AAUDIO_SHARING_MODE_SHARED;
+ int32_t framesMax;
+ int64_t framesTotal;
+ int64_t printAt;
+ int samplesPerBurst;
+ int64_t previousFramePosition = -1;
+
+ AAudioStreamBuilder *aaudioBuilder = nullptr;
+ AAudioStream *aaudioStream = nullptr;
+
+ // Make printf print immediately so that debug info is not stuck
+ // in a buffer if we hang or crash.
+ setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
+
+ printf("Test Timestamps V0.1.1\n");
+
+ AAudio_setMMapPolicy(AAUDIO_POLICY_AUTO);
+
+ // Use an AAudioStreamBuilder to contain requested parameters.
+ result = AAudio_createStreamBuilder(&aaudioBuilder);
+ if (result != AAUDIO_OK) {
+ printf("AAudio_createStreamBuilder returned %s",
+ AAudio_convertResultToText(result));
+ goto finish;
+ }
+
+ // Request stream properties.
+ AAudioStreamBuilder_setFormat(aaudioBuilder, AAUDIO_FORMAT_PCM_FLOAT);
+ //AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, AAUDIO_PERFORMANCE_MODE_NONE);
+ AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+
+ // Create an AAudioStream using the Builder.
+ result = AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream);
+ if (result != AAUDIO_OK) {
+ printf("AAudioStreamBuilder_openStream returned %s",
+ AAudio_convertResultToText(result));
+ goto finish;
+ }
+
+ // Check to see what kind of stream we actually got.
+ actualSampleRate = AAudioStream_getSampleRate(aaudioStream);
+ actualChannelCount = AAudioStream_getChannelCount(aaudioStream);
+ actualDataFormat = AAudioStream_getFormat(aaudioStream);
+
+ printf("-------- chans = %3d, rate = %6d format = %d\n",
+ actualChannelCount, actualSampleRate, actualDataFormat);
+ printf(" Is MMAP used? %s\n", AAudioStream_isMMapUsed(aaudioStream)
+ ? "yes" : "no");
+
+ // This is the number of frames that are read in one chunk by a DMA controller
+ // or a DSP or a mixer.
+ framesPerBurst = AAudioStream_getFramesPerBurst(aaudioStream);
+ printf(" framesPerBurst = %3d\n", framesPerBurst);
+
+ originalBufferSize = AAudioStream_getBufferSizeInFrames(aaudioStream);
+ requestedBufferSize = 2 * framesPerBurst;
+ finalBufferSize = AAudioStream_setBufferSizeInFrames(aaudioStream, requestedBufferSize);
+
+ printf(" BufferSize: original = %4d, requested = %4d, final = %4d\n",
+ originalBufferSize, requestedBufferSize, finalBufferSize);
+
+ samplesPerBurst = framesPerBurst * actualChannelCount;
+ buffer = new float[samplesPerBurst];
+
+ result = AAudioStream_requestStart(aaudioStream);
+ if (result != AAUDIO_OK) {
+ printf("AAudioStream_requestStart returned %s",
+ AAudio_convertResultToText(result));
+ goto finish;
+ }
+
+ // Play silence very briefly.
+ framesMax = actualSampleRate * 4;
+ framesTotal = 0;
+ printAt = actualSampleRate;
+ while (result == AAUDIO_OK && framesTotal < framesMax) {
+ int32_t framesWritten = AAudioStream_write(aaudioStream,
+ buffer, framesPerBurst,
+ DEFAULT_TIMEOUT_NANOS);
+ if (framesWritten < 0) {
+ result = framesWritten;
+ printf("write() returned %s, frames = %d\n",
+ AAudio_convertResultToText(result), (int)framesTotal);
+ printf(" frames = %d\n", (int)framesTotal);
+ } else if (framesWritten != framesPerBurst) {
+ printf("write() returned %d, frames = %d\n", framesWritten, (int)framesTotal);
+ result = AAUDIO_ERROR_TIMEOUT;
+ } else {
+ framesTotal += framesWritten;
+ if (framesTotal >= printAt) {
+ printf("frames = %d\n", (int)framesTotal);
+ printAt += actualSampleRate;
+ }
+ }
+
+ // Print timestamps.
+ int64_t framePosition = 0;
+ int64_t frameTime = 0;
+ aaudio_result_t timeResult;
+ timeResult = AAudioStream_getTimestamp(aaudioStream, CLOCK_MONOTONIC,
+ &framePosition, &frameTime);
+
+ if (timeResult == AAUDIO_OK) {
+ if (framePosition > (previousFramePosition + 5000)) {
+ int64_t realTime = getNanoseconds();
+ int64_t framesWritten = AAudioStream_getFramesWritten(aaudioStream);
+
+ double latencyMillis = calculateLatencyMillis(framePosition, frameTime,
+ framesWritten, realTime,
+ actualSampleRate);
+
+ printf("--- timestamp: result = %4d, position = %lld, at %lld nanos"
+ ", latency = %7.2f msec\n",
+ timeResult,
+ (long long) framePosition,
+ (long long) frameTime,
+ latencyMillis);
+ previousFramePosition = framePosition;
+ }
+ }
+ }
+
+ result = AAudioStream_requestStop(aaudioStream);
+ if (result != AAUDIO_OK) {
+ printf("AAudioStream_requestStop returned %s\n",
+ AAudio_convertResultToText(result));
+ }
+ result = AAudioStream_close(aaudioStream);
+ if (result != AAUDIO_OK) {
+ printf("AAudioStream_close returned %s\n",
+ AAudio_convertResultToText(result));
+ }
+ aaudioStream = nullptr;
+
+
+finish:
+ if (aaudioStream != nullptr) {
+ AAudioStream_close(aaudioStream);
+ }
+ AAudioStreamBuilder_delete(aaudioBuilder);
+ delete[] buffer;
+ printf("result = %d = %s\n", result, AAudio_convertResultToText(result));
+}
diff --git a/services/oboeservice/AAudioServiceEndpoint.cpp b/services/oboeservice/AAudioServiceEndpoint.cpp
index 0f863fe..6c345cd 100644
--- a/services/oboeservice/AAudioServiceEndpoint.cpp
+++ b/services/oboeservice/AAudioServiceEndpoint.cpp
@@ -177,7 +177,10 @@
configuration.getSamplesPerFrame() != mStreamInternal->getSamplesPerFrame()) {
return false;
}
-
return true;
}
+
+aaudio_result_t AAudioServiceEndpoint::getTimestamp(int64_t *positionFrames, int64_t *timeNanos) {
+ return mStreamInternal->getTimestamp(CLOCK_MONOTONIC, positionFrames, timeNanos);
+}
diff --git a/services/oboeservice/AAudioServiceEndpoint.h b/services/oboeservice/AAudioServiceEndpoint.h
index e40a670..603d497 100644
--- a/services/oboeservice/AAudioServiceEndpoint.h
+++ b/services/oboeservice/AAudioServiceEndpoint.h
@@ -69,6 +69,8 @@
mReferenceCount = count;
}
+ aaudio_result_t getTimestamp(int64_t *positionFrames, int64_t *timeNanos);
+
bool matches(const AAudioStreamConfiguration& configuration);
virtual AudioStreamInternal *getStreamInternal() = 0;
diff --git a/services/oboeservice/AAudioServiceEndpointCapture.cpp b/services/oboeservice/AAudioServiceEndpointCapture.cpp
index 6a37330..6504cc1 100644
--- a/services/oboeservice/AAudioServiceEndpointCapture.cpp
+++ b/services/oboeservice/AAudioServiceEndpointCapture.cpp
@@ -62,6 +62,9 @@
// result might be a frame count
while (mCallbackEnabled.load() && getStreamInternal()->isActive() && (result >= 0)) {
+
+ int64_t mmapFramesRead = getStreamInternal()->getFramesRead();
+
// Read audio data from stream using a blocking read.
result = getStreamInternal()->read(mDistributionBuffer, getFramesPerBurst(), timeoutNanos);
if (result == AAUDIO_ERROR_DISCONNECTED) {
@@ -74,18 +77,32 @@
}
// Distribute data to each active stream.
- { // use lock guard
+ { // brackets are for lock_guard
+
std::lock_guard <std::mutex> lock(mLockStreams);
- for (sp<AAudioServiceStreamShared> sharedStream : mRegisteredStreams) {
- if (sharedStream->isRunning()) {
- FifoBuffer *fifo = sharedStream->getDataFifoBuffer();
+ for (sp<AAudioServiceStreamShared> clientStream : mRegisteredStreams) {
+ if (clientStream->isRunning()) {
+ FifoBuffer *fifo = clientStream->getDataFifoBuffer();
+
+ // Determine offset between framePosition in client's stream vs the underlying
+ // MMAP stream.
+ int64_t clientFramesWritten = fifo->getWriteCounter();
+ // There are two indices that refer to the same frame.
+ int64_t positionOffset = mmapFramesRead - clientFramesWritten;
+ clientStream->setTimestampPositionOffset(positionOffset);
+
if (fifo->getFifoControllerBase()->getEmptyFramesAvailable() <
getFramesPerBurst()) {
underflowCount++;
} else {
fifo->write(mDistributionBuffer, getFramesPerBurst());
}
- sharedStream->markTransferTime(AudioClock::getNanoseconds());
+
+ // This timestamp represents the completion of data being written into the
+ // client buffer. It is sent to the client and used in the timing model
+ // to decide when data will be available to read.
+ Timestamp timestamp(fifo->getWriteCounter(), AudioClock::getNanoseconds());
+ clientStream->markTransferTime(timestamp);
}
}
}
diff --git a/services/oboeservice/AAudioServiceEndpointPlay.cpp b/services/oboeservice/AAudioServiceEndpointPlay.cpp
index b83b918..20cc5b8 100644
--- a/services/oboeservice/AAudioServiceEndpointPlay.cpp
+++ b/services/oboeservice/AAudioServiceEndpointPlay.cpp
@@ -73,17 +73,31 @@
while (mCallbackEnabled.load() && getStreamInternal()->isActive() && (result >= 0)) {
// Mix data from each active stream.
mMixer.clear();
- { // use lock guard
+ { // brackets are for lock_guard
int index = 0;
+ int64_t mmapFramesWritten = getStreamInternal()->getFramesWritten();
+
std::lock_guard <std::mutex> lock(mLockStreams);
- for (sp<AAudioServiceStreamShared> sharedStream : mRegisteredStreams) {
- if (sharedStream->isRunning()) {
- FifoBuffer *fifo = sharedStream->getDataFifoBuffer();
+ for (sp<AAudioServiceStreamShared> clientStream : mRegisteredStreams) {
+ if (clientStream->isRunning()) {
+ FifoBuffer *fifo = clientStream->getDataFifoBuffer();
+ // Determine offset between framePosition in client's stream vs the underlying
+ // MMAP stream.
+ int64_t clientFramesRead = fifo->getReadCounter();
+ // These two indices refer to the same frame.
+ int64_t positionOffset = mmapFramesWritten - clientFramesRead;
+ clientStream->setTimestampPositionOffset(positionOffset);
+
float volume = 1.0; // to match legacy volume
bool underflowed = mMixer.mix(index, fifo, volume);
underflowCount += underflowed ? 1 : 0;
// TODO log underflows in each stream
- sharedStream->markTransferTime(AudioClock::getNanoseconds());
+
+ // This timestamp represents the completion of data being read out of the
+ // client buffer. It is sent to the client and used in the timing model
+ // to decide when the client has room to write more data.
+ Timestamp timestamp(fifo->getReadCounter(), AudioClock::getNanoseconds());
+ clientStream->markTransferTime(timestamp);
}
index++;
}
diff --git a/services/oboeservice/AAudioServiceStreamBase.cpp b/services/oboeservice/AAudioServiceStreamBase.cpp
index 5f7d179..3fc5957 100644
--- a/services/oboeservice/AAudioServiceStreamBase.cpp
+++ b/services/oboeservice/AAudioServiceStreamBase.cpp
@@ -37,7 +37,8 @@
AAudioServiceStreamBase::AAudioServiceStreamBase()
: mUpMessageQueue(nullptr)
- , mAAudioThread() {
+ , mAAudioThread()
+ , mAtomicTimestamp() {
mMmapClient.clientUid = -1;
mMmapClient.clientPid = -1;
mMmapClient.packageName = String16("");
@@ -211,15 +212,25 @@
aaudio_result_t AAudioServiceStreamBase::sendCurrentTimestamp() {
AAudioServiceMessage command;
+ // Send a timestamp for the clock model.
aaudio_result_t result = getFreeRunningPosition(&command.timestamp.position,
&command.timestamp.timestamp);
if (result == AAUDIO_OK) {
- // ALOGD("sendCurrentTimestamp(): position = %lld, nanos = %lld",
- // (long long) command.timestamp.position,
- // (long long) command.timestamp.timestamp);
- command.what = AAudioServiceMessage::code::TIMESTAMP;
+ command.what = AAudioServiceMessage::code::TIMESTAMP_SERVICE;
result = writeUpMessageQueue(&command);
- } else if (result == AAUDIO_ERROR_UNAVAILABLE) {
+
+ if (result == AAUDIO_OK) {
+ // Send a hardware timestamp for presentation time.
+ result = getHardwareTimestamp(&command.timestamp.position,
+ &command.timestamp.timestamp);
+ if (result == AAUDIO_OK) {
+ command.what = AAudioServiceMessage::code::TIMESTAMP_HARDWARE;
+ result = writeUpMessageQueue(&command);
+ }
+ }
+ }
+
+ if (result == AAUDIO_ERROR_UNAVAILABLE) {
result = AAUDIO_OK; // just not available yet, try again later
}
return result;
diff --git a/services/oboeservice/AAudioServiceStreamBase.h b/services/oboeservice/AAudioServiceStreamBase.h
index cebefec..e91ea82 100644
--- a/services/oboeservice/AAudioServiceStreamBase.h
+++ b/services/oboeservice/AAudioServiceStreamBase.h
@@ -20,6 +20,7 @@
#include <assert.h>
#include <mutex>
+#include <media/AudioClient.h>
#include <utils/RefBase.h>
#include "fifo/FifoBuffer.h"
@@ -27,7 +28,7 @@
#include "binding/AudioEndpointParcelable.h"
#include "binding/AAudioServiceMessage.h"
#include "utility/AAudioUtilities.h"
-#include <media/AudioClient.h>
+#include "utility/AudioClock.h"
#include "SharedRingBuffer.h"
#include "AAudioThread.h"
@@ -170,6 +171,8 @@
*/
virtual aaudio_result_t getFreeRunningPosition(int64_t *positionFrames, int64_t *timeNanos) = 0;
+ virtual aaudio_result_t getHardwareTimestamp(int64_t *positionFrames, int64_t *timeNanos) = 0;
+
virtual aaudio_result_t getDownDataDescription(AudioEndpointParcelable &parcelable) = 0;
aaudio_stream_state_t mState = AAUDIO_STREAM_STATE_UNINITIALIZED;
@@ -191,6 +194,8 @@
android::AudioClient mMmapClient;
audio_port_handle_t mClientHandle = AUDIO_PORT_HANDLE_NONE;
+ SimpleDoubleBuffer<Timestamp> mAtomicTimestamp;
+
private:
aaudio_handle_t mHandle = -1;
};
diff --git a/services/oboeservice/AAudioServiceStreamMMAP.cpp b/services/oboeservice/AAudioServiceStreamMMAP.cpp
index ff02c0f..970d734 100644
--- a/services/oboeservice/AAudioServiceStreamMMAP.cpp
+++ b/services/oboeservice/AAudioServiceStreamMMAP.cpp
@@ -37,6 +37,11 @@
#define AAUDIO_BUFFER_CAPACITY_MIN 4 * 512
#define AAUDIO_SAMPLE_RATE_DEFAULT 48000
+// This is an estimate of the time difference between the HW and the MMAP time.
+// TODO Get presentation timestamps from the HAL instead of using these estimates.
+#define OUTPUT_ESTIMATED_HARDWARE_OFFSET_NANOS (3 * AAUDIO_NANOS_PER_MILLISECOND)
+#define INPUT_ESTIMATED_HARDWARE_OFFSET_NANOS (-1 * AAUDIO_NANOS_PER_MILLISECOND)
+
/**
* Service Stream that uses an MMAP buffer.
*/
@@ -113,10 +118,14 @@
config.channel_mask = (aaudioSamplesPerFrame == AAUDIO_UNSPECIFIED)
? AUDIO_CHANNEL_OUT_STEREO
: audio_channel_out_mask_from_count(aaudioSamplesPerFrame);
+ mHardwareTimeOffsetNanos = OUTPUT_ESTIMATED_HARDWARE_OFFSET_NANOS; // frames at DAC later
+
} else if (direction == AAUDIO_DIRECTION_INPUT) {
config.channel_mask = (aaudioSamplesPerFrame == AAUDIO_UNSPECIFIED)
? AUDIO_CHANNEL_IN_STEREO
: audio_channel_in_mask_from_count(aaudioSamplesPerFrame);
+ mHardwareTimeOffsetNanos = INPUT_ESTIMATED_HARDWARE_OFFSET_NANOS; // frames at ADC earlier
+
} else {
ALOGE("openMmapStream - invalid direction = %d", direction);
return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
@@ -289,6 +298,7 @@
return AAudioConvert_androidToAAudioResult(mMmapStream->stop(clientHandle));
}
+// Get free-running DSP or DMA hardware position from the HAL.
aaudio_result_t AAudioServiceStreamMMAP::getFreeRunningPosition(int64_t *positionFrames,
int64_t *timeNanos) {
struct audio_mmap_position position;
@@ -305,12 +315,29 @@
disconnect();
} else {
mFramesRead.update32(position.position_frames);
- *positionFrames = mFramesRead.get();
- *timeNanos = position.time_nanoseconds;
+
+ Timestamp timestamp(mFramesRead.get(), position.time_nanoseconds);
+ mAtomicTimestamp.write(timestamp);
+ *positionFrames = timestamp.getPosition();
+ *timeNanos = timestamp.getNanoseconds();
}
return result;
}
+// Get timestamp that was written by getFreeRunningPosition()
+aaudio_result_t AAudioServiceStreamMMAP::getHardwareTimestamp(int64_t *positionFrames,
+ int64_t *timeNanos) {
+ // TODO Get presentation timestamp from the HAL
+ if (mAtomicTimestamp.isValid()) {
+ Timestamp timestamp = mAtomicTimestamp.read();
+ *positionFrames = timestamp.getPosition();
+ *timeNanos = timestamp.getNanoseconds() + mHardwareTimeOffsetNanos;
+ return AAUDIO_OK;
+ } else {
+ return AAUDIO_ERROR_UNAVAILABLE;
+ }
+}
+
void AAudioServiceStreamMMAP::onTearDown() {
ALOGD("AAudioServiceStreamMMAP::onTearDown() called");
disconnect();
diff --git a/services/oboeservice/AAudioServiceStreamMMAP.h b/services/oboeservice/AAudioServiceStreamMMAP.h
index 533e5a8..e6f8fad 100644
--- a/services/oboeservice/AAudioServiceStreamMMAP.h
+++ b/services/oboeservice/AAudioServiceStreamMMAP.h
@@ -100,6 +100,8 @@
aaudio_result_t getDownDataDescription(AudioEndpointParcelable &parcelable) override;
aaudio_result_t getFreeRunningPosition(int64_t *positionFrames, int64_t *timeNanos) override;
+ virtual aaudio_result_t getHardwareTimestamp(int64_t *positionFrames,
+ int64_t *timeNanos) override;
private:
// This proxy class was needed to prevent a crash in AudioFlinger
@@ -132,6 +134,7 @@
MonotonicCounter mFramesRead;
int32_t mPreviousFrameCounter = 0; // from HAL
int mAudioDataFileDescriptor = -1;
+ int64_t mHardwareTimeOffsetNanos = 0; // TODO get from HAL
// Interface to the AudioFlinger MMAP support.
android::sp<android::MmapStreamInterface> mMmapStream;
diff --git a/services/oboeservice/AAudioServiceStreamShared.cpp b/services/oboeservice/AAudioServiceStreamShared.cpp
index 5654113..07c4faf 100644
--- a/services/oboeservice/AAudioServiceStreamShared.cpp
+++ b/services/oboeservice/AAudioServiceStreamShared.cpp
@@ -41,6 +41,7 @@
AAudioServiceStreamShared::AAudioServiceStreamShared(AAudioService &audioService)
: mAudioService(audioService)
+ , mTimestampPositionOffset(0)
{
}
@@ -307,15 +308,30 @@
return AAUDIO_OK;
}
-void AAudioServiceStreamShared::markTransferTime(int64_t nanoseconds) {
- mMarkedPosition = mAudioDataQueue->getFifoBuffer()->getReadCounter();
- mMarkedTime = nanoseconds;
+void AAudioServiceStreamShared::markTransferTime(Timestamp ×tamp) {
+ mAtomicTimestamp.write(timestamp);
}
+// Get timestamp that was written by the real-time service thread, eg. mixer.
aaudio_result_t AAudioServiceStreamShared::getFreeRunningPosition(int64_t *positionFrames,
int64_t *timeNanos) {
- // TODO get these two numbers as an atomic pair
- *positionFrames = mMarkedPosition;
- *timeNanos = mMarkedTime;
- return AAUDIO_OK;
+ if (mAtomicTimestamp.isValid()) {
+ Timestamp timestamp = mAtomicTimestamp.read();
+ *positionFrames = timestamp.getPosition();
+ *timeNanos = timestamp.getNanoseconds();
+ return AAUDIO_OK;
+ } else {
+ return AAUDIO_ERROR_UNAVAILABLE;
+ }
+}
+
+// Get timestamp from lower level service.
+aaudio_result_t AAudioServiceStreamShared::getHardwareTimestamp(int64_t *positionFrames,
+ int64_t *timeNanos) {
+
+ aaudio_result_t result = mServiceEndpoint->getTimestamp(positionFrames, timeNanos);
+ if (result == AAUDIO_OK) {
+ *positionFrames -= mTimestampPositionOffset.load(); // Offset from shared MMAP stream
+ }
+ return result;
}
diff --git a/services/oboeservice/AAudioServiceStreamShared.h b/services/oboeservice/AAudioServiceStreamShared.h
index 6b67337..8caccda 100644
--- a/services/oboeservice/AAudioServiceStreamShared.h
+++ b/services/oboeservice/AAudioServiceStreamShared.h
@@ -85,7 +85,11 @@
/* Keep a record of when a buffer transfer completed.
* This allows for a more accurate timing model.
*/
- void markTransferTime(int64_t nanoseconds);
+ void markTransferTime(Timestamp ×tamp);
+
+ void setTimestampPositionOffset(int64_t deltaFrames) {
+ mTimestampPositionOffset.store(deltaFrames);
+ }
protected:
@@ -93,6 +97,9 @@
aaudio_result_t getFreeRunningPosition(int64_t *positionFrames, int64_t *timeNanos) override;
+ virtual aaudio_result_t getHardwareTimestamp(int64_t *positionFrames,
+ int64_t *timeNanos) override;
+
/**
* @param requestedCapacityFrames
* @param framesPerBurst
@@ -106,8 +113,7 @@
AAudioServiceEndpoint *mServiceEndpoint = nullptr;
SharedRingBuffer *mAudioDataQueue = nullptr;
- int64_t mMarkedPosition = 0;
- int64_t mMarkedTime = 0;
+ std::atomic<int64_t> mTimestampPositionOffset;
};
} /* namespace aaudio */