AAudioService: integrated with audioserver
Call the MmapStreamInterface from AudioFlinger instead of the FakeHAL.
Fix sending timestamps from the thread.
Add shared mode in service.
Bug: 35260844
Test: CTS test_aaudio.cpp
Change-Id: I44c7e4ecae4ce205611b6b73a72e0ae8a5b243e5
Signed-off-by: Phil Burk <philburk@google.com>
diff --git a/services/oboeservice/AAudioServiceEndpoint.cpp b/services/oboeservice/AAudioServiceEndpoint.cpp
new file mode 100644
index 0000000..80551c9
--- /dev/null
+++ b/services/oboeservice/AAudioServiceEndpoint.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.
+ */
+
+#include <algorithm>
+#include <mutex>
+#include <vector>
+
+#include "core/AudioStreamBuilder.h"
+#include "AAudioServiceEndpoint.h"
+#include "AAudioServiceStreamShared.h"
+
+using namespace android; // TODO just import names needed
+using namespace aaudio; // TODO just import names needed
+
+#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
+
+// The mStreamInternal will use a service interface that does not go through Binder.
+AAudioServiceEndpoint::AAudioServiceEndpoint(AAudioService &audioService)
+ : mStreamInternal(audioService, true)
+ {
+}
+
+AAudioServiceEndpoint::~AAudioServiceEndpoint() {
+}
+
+// Set up an EXCLUSIVE MMAP stream that will be shared.
+aaudio_result_t AAudioServiceEndpoint::open(int32_t deviceId, aaudio_direction_t direction) {
+ AudioStreamBuilder builder;
+ builder.setSharingMode(AAUDIO_SHARING_MODE_EXCLUSIVE);
+ builder.setDeviceId(deviceId);
+ builder.setDirection(direction);
+ aaudio_result_t result = mStreamInternal.open(builder);
+ if (result == AAUDIO_OK) {
+ mMixer.allocate(mStreamInternal.getSamplesPerFrame(), mStreamInternal.getFramesPerBurst());
+ }
+ return result;
+}
+
+aaudio_result_t AAudioServiceEndpoint::close() {
+ return mStreamInternal.close();
+}
+
+// TODO, maybe use an interface to reduce exposure
+aaudio_result_t AAudioServiceEndpoint::registerStream(AAudioServiceStreamShared *sharedStream) {
+ ALOGD("AAudioServiceEndpoint::registerStream(%p)", sharedStream);
+ // TODO use real-time technique to avoid mutex, eg. atomic command FIFO
+ std::lock_guard<std::mutex> lock(mLockStreams);
+ mRegisteredStreams.push_back(sharedStream);
+ return AAUDIO_OK;
+}
+
+aaudio_result_t AAudioServiceEndpoint::unregisterStream(AAudioServiceStreamShared *sharedStream) {
+ ALOGD("AAudioServiceEndpoint::unregisterStream(%p)", sharedStream);
+ std::lock_guard<std::mutex> lock(mLockStreams);
+ mRegisteredStreams.erase(std::remove(mRegisteredStreams.begin(), mRegisteredStreams.end(), sharedStream),
+ mRegisteredStreams.end());
+ return AAUDIO_OK;
+}
+
+aaudio_result_t AAudioServiceEndpoint::startStream(AAudioServiceStreamShared *sharedStream) {
+ // TODO use real-time technique to avoid mutex, eg. atomic command FIFO
+ ALOGD("AAudioServiceEndpoint(): startStream() entering");
+ std::lock_guard<std::mutex> lock(mLockStreams);
+ mRunningStreams.push_back(sharedStream);
+ if (mRunningStreams.size() == 1) {
+ startMixer_l();
+ }
+ return AAUDIO_OK;
+}
+
+aaudio_result_t AAudioServiceEndpoint::stopStream(AAudioServiceStreamShared *sharedStream) {
+ std::lock_guard<std::mutex> lock(mLockStreams);
+ mRunningStreams.erase(std::remove(mRunningStreams.begin(), mRunningStreams.end(), sharedStream),
+ mRunningStreams.end());
+ if (mRunningStreams.size() == 0) {
+ stopMixer_l();
+ }
+ return AAUDIO_OK;
+}
+
+static void *aaudio_mixer_thread_proc(void *context) {
+ AAudioServiceEndpoint *stream = (AAudioServiceEndpoint *) context;
+ //LOGD("AudioStreamAAudio(): oboe_callback_thread, stream = %p", stream);
+ if (stream != NULL) {
+ return stream->callbackLoop();
+ } else {
+ return NULL;
+ }
+}
+
+// Render audio in the application callback and then write the data to the stream.
+void *AAudioServiceEndpoint::callbackLoop() {
+ aaudio_result_t result = AAUDIO_OK;
+
+ ALOGD("AAudioServiceEndpoint(): callbackLoop() entering");
+
+ result = mStreamInternal.requestStart();
+ ALOGD("AAudioServiceEndpoint(): callbackLoop() after requestStart() %d, isPlaying() = %d",
+ result, (int) mStreamInternal.isPlaying());
+
+ // result might be a frame count
+ while (mCallbackEnabled.load() && mStreamInternal.isPlaying() && (result >= 0)) {
+ // Mix data from each active stream.
+ {
+ mMixer.clear();
+ std::lock_guard<std::mutex> lock(mLockStreams);
+ for(AAudioServiceStreamShared *sharedStream : mRunningStreams) {
+ FifoBuffer *fifo = sharedStream->getDataFifoBuffer();
+ float volume = 0.5; // TODO get from system
+ mMixer.mix(fifo, volume);
+ }
+ }
+
+ // Write audio data to stream using a blocking write.
+ ALOGD("AAudioServiceEndpoint(): callbackLoop() write(%d)", getFramesPerBurst());
+ int64_t timeoutNanos = calculateReasonableTimeout(mStreamInternal.getFramesPerBurst());
+ result = mStreamInternal.write(mMixer.getOutputBuffer(), getFramesPerBurst(), timeoutNanos);
+ if (result == AAUDIO_ERROR_DISCONNECTED) {
+ disconnectRegisteredStreams();
+ break;
+ } else if (result != getFramesPerBurst()) {
+ ALOGW("AAudioServiceEndpoint(): callbackLoop() wrote %d / %d",
+ result, getFramesPerBurst());
+ break;
+ }
+ }
+
+ ALOGD("AAudioServiceEndpoint(): callbackLoop() exiting, result = %d, isPlaying() = %d",
+ result, (int) mStreamInternal.isPlaying());
+
+ result = mStreamInternal.requestStop();
+
+ return NULL; // TODO review
+}
+
+aaudio_result_t AAudioServiceEndpoint::startMixer_l() {
+ // Launch the callback loop thread.
+ int64_t periodNanos = mStreamInternal.getFramesPerBurst()
+ * AAUDIO_NANOS_PER_SECOND
+ / getSampleRate();
+ mCallbackEnabled.store(true);
+ return mStreamInternal.createThread(periodNanos, aaudio_mixer_thread_proc, this);
+}
+
+aaudio_result_t AAudioServiceEndpoint::stopMixer_l() {
+ mCallbackEnabled.store(false);
+ return mStreamInternal.joinThread(NULL, calculateReasonableTimeout(mStreamInternal.getFramesPerBurst()));
+}
+
+// TODO Call method in AudioStreamInternal when that callback CL is merged.
+int64_t AAudioServiceEndpoint::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;
+}
+
+void AAudioServiceEndpoint::disconnectRegisteredStreams() {
+ std::lock_guard<std::mutex> lock(mLockStreams);
+ for(AAudioServiceStreamShared *sharedStream : mRunningStreams) {
+ sharedStream->onStop();
+ }
+ mRunningStreams.clear();
+ for(AAudioServiceStreamShared *sharedStream : mRegisteredStreams) {
+ sharedStream->onDisconnect();
+ }
+ mRegisteredStreams.clear();
+}