libaaudio: implement callback
Use AudioTrack and AudioRecord TRANSFER_CALLBACK.
Add FixedBlockAdapter to provide fixed size callbacks.
Bug: 36489240
Test: CTS test_aaudio.cpp
Change-Id: Id2034dd640f878dd27fee6b43ad80a01c627dfd6
Signed-off-by: Phil Burk <philburk@google.com>
diff --git a/media/libaaudio/src/client/AudioEndpoint.cpp b/media/libaaudio/src/client/AudioEndpoint.cpp
index 47c4774..90c619c 100644
--- a/media/libaaudio/src/client/AudioEndpoint.cpp
+++ b/media/libaaudio/src/client/AudioEndpoint.cpp
@@ -19,7 +19,7 @@
#include <utils/Log.h>
#include <cassert>
-#include <aaudio/AAudioDefinitions.h>
+#include <aaudio/AAudio.h>
#include "AudioEndpointParcelable.h"
#include "AudioEndpoint.h"
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 54f4870..1f9ce4f 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -18,23 +18,19 @@
//#define LOG_NDEBUG 0
#include <utils/Log.h>
-#include <stdint.h>
#include <assert.h>
#include <binder/IServiceManager.h>
#include <utils/Mutex.h>
#include <aaudio/AAudio.h>
+#include <utils/String16.h>
-#include "AudioClock.h"
-#include "AudioEndpointParcelable.h"
-#include "binding/AAudioStreamRequest.h"
-#include "binding/AAudioStreamConfiguration.h"
-#include "binding/IAAudioService.h"
+#include "utility/AudioClock.h"
+#include "AudioStreamInternal.h"
#include "binding/AAudioServiceMessage.h"
#include "core/AudioStreamBuilder.h"
-#include "AudioStreamInternal.h"
#define LOG_TIMESTAMPS 0
@@ -51,6 +47,11 @@
#define AAUDIO_SERVICE_NAME "AAudioService"
+#define MIN_TIMEOUT_NANOS (1000 * AAUDIO_NANOS_PER_MILLISECOND)
+
+// Wait at least this many times longer than the operation should take.
+#define MIN_TIMEOUT_OPERATIONS 4
+
// Helper function to get access to the "AAudioService" service.
// This code was modeled after frameworks/av/media/libaudioclient/AudioSystem.cpp
static const sp<IAAudioService> getAAudioService() {
@@ -151,6 +152,29 @@
mClockModel.setSampleRate(getSampleRate());
mClockModel.setFramesPerBurst(mFramesPerBurst);
+ if (getDataCallbackProc()) {
+ mCallbackFrames = builder.getFramesPerDataCallback();
+ if (mCallbackFrames > getBufferCapacity() / 2) {
+ ALOGE("AudioStreamInternal.open(): framesPerCallback too large");
+ service->closeStream(mServiceStreamHandle);
+ return AAUDIO_ERROR_OUT_OF_RANGE;
+
+ } else if (mCallbackFrames < 0) {
+ ALOGE("AudioStreamInternal.open(): framesPerCallback negative");
+ service->closeStream(mServiceStreamHandle);
+ return AAUDIO_ERROR_OUT_OF_RANGE;
+
+ }
+ if (mCallbackFrames == AAUDIO_UNSPECIFIED) {
+ mCallbackFrames = mFramesPerBurst;
+ }
+
+ int32_t bytesPerFrame = getSamplesPerFrame()
+ * AAudioConvert_formatToSizeInBytes(getFormat());
+ int32_t callbackBufferSize = mCallbackFrames * bytesPerFrame;
+ mCallbackBuffer = new uint8_t[callbackBufferSize];
+ }
+
setState(AAUDIO_STREAM_STATE_OPEN);
}
return result;
@@ -164,12 +188,69 @@
const sp<IAAudioService>& aaudioService = getAAudioService();
if (aaudioService == 0) return AAUDIO_ERROR_NO_SERVICE;
aaudioService->closeStream(serviceStreamHandle);
+ delete[] mCallbackBuffer;
return AAUDIO_OK;
} else {
return AAUDIO_ERROR_INVALID_HANDLE;
}
}
+// Render audio in the application callback and then write the data to the stream.
+void *AudioStreamInternal::callbackLoop() {
+ aaudio_result_t result = AAUDIO_OK;
+ aaudio_data_callback_result_t callbackResult = AAUDIO_CALLBACK_RESULT_CONTINUE;
+ int32_t framesWritten = 0;
+ AAudioStream_dataCallback appCallback = getDataCallbackProc();
+ if (appCallback == nullptr) return NULL;
+
+ while (mCallbackEnabled.load() && isPlaying() && (result >= 0)) { // result might be a frame count
+ // Call application using the AAudio callback interface.
+ callbackResult = (*appCallback)(
+ (AAudioStream *) this,
+ getDataCallbackUserData(),
+ mCallbackBuffer,
+ mCallbackFrames);
+
+ if (callbackResult == AAUDIO_CALLBACK_RESULT_CONTINUE) {
+ // Write audio data to stream
+ int64_t timeoutNanos = calculateReasonableTimeout(mCallbackFrames);
+ result = write(mCallbackBuffer, mCallbackFrames, timeoutNanos);
+ if (result == AAUDIO_ERROR_DISCONNECTED) {
+ if (getErrorCallbackProc() != nullptr) {
+ ALOGD("AudioStreamAAudio(): callbackLoop() stream disconnected");
+ (*getErrorCallbackProc())(
+ (AAudioStream *) this,
+ getErrorCallbackUserData(),
+ AAUDIO_OK);
+ }
+ break;
+ } else if (result != mCallbackFrames) {
+ ALOGE("AudioStreamAAudio(): callbackLoop() wrote %d / %d",
+ framesWritten, mCallbackFrames);
+ break;
+ }
+ } else if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
+ ALOGD("AudioStreamAAudio(): callback returned AAUDIO_CALLBACK_RESULT_STOP");
+ break;
+ }
+ }
+
+ ALOGD("AudioStreamAAudio(): callbackLoop() exiting, result = %d, isPlaying() = %d",
+ result, (int) isPlaying());
+ return NULL; // TODO review
+}
+
+static void *aaudio_callback_thread_proc(void *context)
+{
+ AudioStreamInternal *stream = (AudioStreamInternal *)context;
+ //LOGD("AudioStreamAAudio(): oboe_callback_thread, stream = %p", stream);
+ if (stream != NULL) {
+ return stream->callbackLoop();
+ } else {
+ return NULL;
+ }
+}
+
aaudio_result_t AudioStreamInternal::requestStart()
{
int64_t startTime;
@@ -178,35 +259,81 @@
return AAUDIO_ERROR_INVALID_STATE;
}
const sp<IAAudioService>& aaudioService = getAAudioService();
- if (aaudioService == 0) return AAUDIO_ERROR_NO_SERVICE;
+ if (aaudioService == 0) {
+ return AAUDIO_ERROR_NO_SERVICE;
+ }
startTime = AudioClock::getNanoseconds();
mClockModel.start(startTime);
processTimestamp(0, startTime);
setState(AAUDIO_STREAM_STATE_STARTING);
- return aaudioService->startStream(mServiceStreamHandle);
+ aaudio_result_t result = aaudioService->startStream(mServiceStreamHandle);
+
+ if (result == AAUDIO_OK && getDataCallbackProc() != nullptr) {
+ // Launch the callback loop thread.
+ int64_t periodNanos = mCallbackFrames
+ * AAUDIO_NANOS_PER_SECOND
+ / getSampleRate();
+ mCallbackEnabled.store(true);
+ result = createThread(periodNanos, aaudio_callback_thread_proc, this);
+ }
+ return result;
}
-aaudio_result_t AudioStreamInternal::requestPause()
+int64_t AudioStreamInternal::calculateReasonableTimeout(int32_t framesPerOperation) {
+
+ // Wait for at least a second or some number of callbacks to join the thread.
+ int64_t timeoutNanoseconds = (MIN_TIMEOUT_OPERATIONS * framesPerOperation * AAUDIO_NANOS_PER_SECOND)
+ / getSampleRate();
+ if (timeoutNanoseconds < MIN_TIMEOUT_NANOS) { // arbitrary number of seconds
+ timeoutNanoseconds = MIN_TIMEOUT_NANOS;
+ }
+ return timeoutNanoseconds;
+}
+
+aaudio_result_t AudioStreamInternal::stopCallback()
+{
+ if (isDataCallbackActive()) {
+ mCallbackEnabled.store(false);
+ return joinThread(NULL, calculateReasonableTimeout(mCallbackFrames));
+ } else {
+ return AAUDIO_OK;
+ }
+}
+
+aaudio_result_t AudioStreamInternal::requestPauseInternal()
{
ALOGD("AudioStreamInternal(): pause()");
if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
return AAUDIO_ERROR_INVALID_STATE;
}
const sp<IAAudioService>& aaudioService = getAAudioService();
- if (aaudioService == 0) return AAUDIO_ERROR_NO_SERVICE;
+ if (aaudioService == 0) {
+ return AAUDIO_ERROR_NO_SERVICE;
+ }
mClockModel.stop(AudioClock::getNanoseconds());
setState(AAUDIO_STREAM_STATE_PAUSING);
return aaudioService->pauseStream(mServiceStreamHandle);
}
+aaudio_result_t AudioStreamInternal::requestPause()
+{
+ aaudio_result_t result = stopCallback();
+ if (result != AAUDIO_OK) {
+ return result;
+ }
+ return requestPauseInternal();
+}
+
aaudio_result_t AudioStreamInternal::requestFlush() {
ALOGD("AudioStreamInternal(): flush()");
if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
return AAUDIO_ERROR_INVALID_STATE;
}
const sp<IAAudioService>& aaudioService = getAAudioService();
- if (aaudioService == 0) return AAUDIO_ERROR_NO_SERVICE;
-setState(AAUDIO_STREAM_STATE_FLUSHING);
+ if (aaudioService == 0) {
+ return AAUDIO_ERROR_NO_SERVICE;
+ }
+ setState(AAUDIO_STREAM_STATE_FLUSHING);
return aaudioService->flushStream(mServiceStreamHandle);
}
@@ -260,18 +387,20 @@
return aaudioService->unregisterAudioThread(mServiceStreamHandle, gettid());
}
-// TODO use aaudio_clockid_t all the way down to AudioClock
aaudio_result_t AudioStreamInternal::getTimestamp(clockid_t clockId,
int64_t *framePosition,
int64_t *timeNanoseconds) {
-// TODO implement using real HAL
+ // TODO implement using real HAL
int64_t time = AudioClock::getNanoseconds();
*framePosition = mClockModel.convertTimeToPosition(time);
*timeNanoseconds = time + (10 * AAUDIO_NANOS_PER_MILLISECOND); // Fake hardware delay
return AAUDIO_OK;
}
-aaudio_result_t AudioStreamInternal::updateState() {
+aaudio_result_t AudioStreamInternal::updateStateWhileWaiting() {
+ if (isDataCallbackActive()) {
+ return AAUDIO_OK; // state is getting updated by the callback thread read/write call
+ }
return processCommands();
}
@@ -485,43 +614,6 @@
return framesWritten;
}
-aaudio_result_t AudioStreamInternal::waitForStateChange(aaudio_stream_state_t currentState,
- aaudio_stream_state_t *nextState,
- int64_t timeoutNanoseconds)
-
-{
- aaudio_result_t result = processCommands();
-// ALOGD("AudioStreamInternal::waitForStateChange() - processCommands() returned %d", result);
- if (result != AAUDIO_OK) {
- return result;
- }
- // TODO replace this polling with a timed sleep on a futex on the message queue
- int32_t durationNanos = 5 * AAUDIO_NANOS_PER_MILLISECOND;
- aaudio_stream_state_t state = getState();
-// ALOGD("AudioStreamInternal::waitForStateChange() - state = %d", state);
- while (state == currentState && timeoutNanoseconds > 0) {
- // TODO use futex from service message queue
- if (durationNanos > timeoutNanoseconds) {
- durationNanos = timeoutNanoseconds;
- }
- AudioClock::sleepForNanos(durationNanos);
- timeoutNanoseconds -= durationNanos;
-
- result = processCommands();
- if (result != AAUDIO_OK) {
- return result;
- }
-
- state = getState();
-// ALOGD("AudioStreamInternal::waitForStateChange() - state = %d", state);
- }
- if (nextState != nullptr) {
- *nextState = state;
- }
- return (state == currentState) ? AAUDIO_ERROR_TIMEOUT : AAUDIO_OK;
-}
-
-
void AudioStreamInternal::processTimestamp(uint64_t position, int64_t time) {
mClockModel.processTimestamp( position, time);
}
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 6f3a7ac..9a15a9b 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -53,7 +53,7 @@
int64_t *timeNanoseconds) override;
- virtual aaudio_result_t updateState() override;
+ virtual aaudio_result_t updateStateWhileWaiting() override;
// =========== End ABSTRACT methods ===========================
virtual aaudio_result_t open(const AudioStreamBuilder &builder) override;
@@ -64,10 +64,6 @@
int32_t numFrames,
int64_t timeoutNanoseconds) override;
- virtual aaudio_result_t waitForStateChange(aaudio_stream_state_t currentState,
- aaudio_stream_state_t *nextState,
- int64_t timeoutNanoseconds) override;
-
virtual aaudio_result_t setBufferSize(int32_t requestedFrames) override;
virtual int32_t getBufferSize() const override;
@@ -86,10 +82,17 @@
virtual aaudio_result_t unregisterThread() override;
+ // Called internally from 'C'
+ void *callbackLoop();
+
protected:
aaudio_result_t processCommands();
+ aaudio_result_t requestPauseInternal();
+
+ aaudio_result_t stopCallback();
+
/**
* Low level write that will not block. It will just write as much as it can.
*
@@ -108,17 +111,22 @@
aaudio_result_t onTimestampFromServer(AAudioServiceMessage *message);
+ // Calculate timeout for an operation involving framesPerOperation.
+ int64_t calculateReasonableTimeout(int32_t framesPerOperation);
+
private:
IsochronousClockModel mClockModel;
AudioEndpoint mAudioEndpoint;
aaudio_handle_t mServiceStreamHandle;
EndpointDescriptor mEndpointDescriptor;
+ uint8_t *mCallbackBuffer = nullptr;
+ int32_t mCallbackFrames = 0;
+
// Offset from underlying frame position.
int64_t mFramesOffsetFromService = 0;
int64_t mLastFramesRead = 0;
int32_t mFramesPerBurst;
int32_t mXRunCount = 0;
-
void processTimestamp(uint64_t position, int64_t time);
};
diff --git a/media/libaaudio/src/client/IsochronousClockModel.cpp b/media/libaaudio/src/client/IsochronousClockModel.cpp
index 4c8aabc..c278c8b 100644
--- a/media/libaaudio/src/client/IsochronousClockModel.cpp
+++ b/media/libaaudio/src/client/IsochronousClockModel.cpp
@@ -19,7 +19,6 @@
#include <utils/Log.h>
#include <stdint.h>
-#include <aaudio/AAudioDefinitions.h>
#include "utility/AudioClock.h"
#include "IsochronousClockModel.h"
diff --git a/media/libaaudio/src/client/IsochronousClockModel.h b/media/libaaudio/src/client/IsochronousClockModel.h
index 524c286..205c341 100644
--- a/media/libaaudio/src/client/IsochronousClockModel.h
+++ b/media/libaaudio/src/client/IsochronousClockModel.h
@@ -14,11 +14,10 @@
* limitations under the License.
*/
-#ifndef AAUDIO_ISOCHRONOUSCLOCKMODEL_H
-#define AAUDIO_ISOCHRONOUSCLOCKMODEL_H
+#ifndef AAUDIO_ISOCHRONOUS_CLOCK_MODEL_H
+#define AAUDIO_ISOCHRONOUS_CLOCK_MODEL_H
#include <stdint.h>
-#include <aaudio/AAudio.h>
namespace aaudio {
@@ -107,4 +106,4 @@
} /* namespace aaudio */
-#endif //AAUDIO_ISOCHRONOUSCLOCKMODEL_H
+#endif //AAUDIO_ISOCHRONOUS_CLOCK_MODEL_H