aaudio: lower latency using MMAP capture

MMAP can be enabled by setting system properties.

Bug: 38267780
Test: input_monitor.cpp
Change-Id: I5e86fd1d9baef4fe59837ccbca7971acbb54d8b5
Signed-off-by: Phil Burk <philburk@google.com>
diff --git a/services/oboeservice/AAudioServiceEndpointPlay.cpp b/services/oboeservice/AAudioServiceEndpointPlay.cpp
new file mode 100644
index 0000000..cc09cc3
--- /dev/null
+++ b/services/oboeservice/AAudioServiceEndpointPlay.cpp
@@ -0,0 +1,109 @@
+/*
+ * 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 "AAudioService"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include <assert.h>
+#include <map>
+#include <mutex>
+#include <utils/Singleton.h>
+
+#include "AAudioEndpointManager.h"
+#include "AAudioServiceEndpoint.h"
+#include <algorithm>
+#include <mutex>
+#include <vector>
+
+#include "core/AudioStreamBuilder.h"
+#include "AAudioServiceEndpoint.h"
+#include "AAudioServiceStreamShared.h"
+#include "AAudioServiceEndpointPlay.h"
+
+using namespace android;  // TODO just import names needed
+using namespace aaudio;   // TODO just import names needed
+
+#define BURSTS_PER_BUFFER_DEFAULT   2
+
+AAudioServiceEndpointPlay::AAudioServiceEndpointPlay(AAudioService &audioService)
+        : mStreamInternalPlay(audioService, true) {
+}
+
+AAudioServiceEndpointPlay::~AAudioServiceEndpointPlay() {
+}
+
+aaudio_result_t AAudioServiceEndpointPlay::open(int32_t deviceId) {
+    aaudio_result_t result = AAudioServiceEndpoint::open(deviceId);
+    if (result == AAUDIO_OK) {
+        mMixer.allocate(getStreamInternal()->getSamplesPerFrame(),
+                        getStreamInternal()->getFramesPerBurst());
+
+        int32_t burstsPerBuffer = AAudioProperty_getMixerBursts();
+        if (burstsPerBuffer == 0) {
+            mLatencyTuningEnabled = true;
+            burstsPerBuffer = BURSTS_PER_BUFFER_DEFAULT;
+        }
+        ALOGD("AAudioServiceEndpoint(): burstsPerBuffer = %d", burstsPerBuffer);
+        int32_t desiredBufferSize = burstsPerBuffer * getStreamInternal()->getFramesPerBurst();
+        getStreamInternal()->setBufferSize(desiredBufferSize);
+    }
+    return result;
+}
+
+// Mix data from each application stream and write result to the shared MMAP stream.
+void *AAudioServiceEndpointPlay::callbackLoop() {
+    ALOGD("AAudioServiceEndpointPlay(): callbackLoop() entering");
+    int32_t underflowCount = 0;
+
+    aaudio_result_t result = getStreamInternal()->requestStart();
+
+    int64_t timeoutNanos = getStreamInternal()->calculateReasonableTimeout();
+
+    // result might be a frame count
+    while (mCallbackEnabled.load() && getStreamInternal()->isActive() && (result >= 0)) {
+        // Mix data from each active stream.
+        mMixer.clear();
+        { // use lock guard
+            std::lock_guard <std::mutex> lock(mLockStreams);
+            for (AAudioServiceStreamShared *sharedStream : mRunningStreams) {
+                FifoBuffer *fifo = sharedStream->getDataFifoBuffer();
+                float volume = 0.5; // TODO get from system
+                bool underflowed = mMixer.mix(fifo, volume);
+                underflowCount += underflowed ? 1 : 0;
+                // TODO log underflows in each stream
+                sharedStream->markTransferTime(AudioClock::getNanoseconds());
+            }
+        }
+
+        // Write mixer output to stream using a blocking write.
+        result = getStreamInternal()->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;
+        }
+    }
+
+    result = getStreamInternal()->requestStop();
+
+    ALOGD("AAudioServiceEndpointPlay(): callbackLoop() exiting, %d underflows", underflowCount);
+    return NULL; // TODO review
+}