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
Bug: 33398120
Test: CTS test_aaudio.cpp
Change-Id: I44c7e4ecae4ce205611b6b73a72e0ae8a5b243e5
Signed-off-by: Phil Burk <philburk@google.com>
(cherry picked from commit 7f6b40d78b1976c78d1300e8a51fda36eeb50c5d)
diff --git a/services/oboeservice/AAudioMixer.cpp b/services/oboeservice/AAudioMixer.cpp
new file mode 100644
index 0000000..70da339
--- /dev/null
+++ b/services/oboeservice/AAudioMixer.cpp
@@ -0,0 +1,85 @@
+/*
+ * 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 <cstring>
+#include "AAudioMixer.h"
+
+using android::WrappingBuffer;
+using android::FifoBuffer;
+using android::fifo_frames_t;
+
+AAudioMixer::~AAudioMixer() {
+ delete[] mOutputBuffer;
+}
+
+void AAudioMixer::allocate(int32_t samplesPerFrame, int32_t framesPerBurst) {
+ mSamplesPerFrame = samplesPerFrame;
+ mFramesPerBurst = framesPerBurst;
+ int32_t samplesPerBuffer = samplesPerFrame * framesPerBurst;
+ mOutputBuffer = new float[samplesPerBuffer];
+ mBufferSizeInBytes = samplesPerBuffer * sizeof(float);
+}
+
+void AAudioMixer::clear() {
+ memset(mOutputBuffer, 0, mBufferSizeInBytes);
+}
+
+void AAudioMixer::mix(FifoBuffer *fifo, float volume) {
+ WrappingBuffer wrappingBuffer;
+ float *destination = mOutputBuffer;
+ fifo_frames_t framesLeft = mFramesPerBurst;
+
+ // Gather the data from the client. May be in two parts.
+ fifo->getFullDataAvailable(&wrappingBuffer);
+
+ // Mix data in one or two parts.
+ int partIndex = 0;
+ while (framesLeft > 0 && partIndex < WrappingBuffer::SIZE) {
+ fifo_frames_t framesToMix = framesLeft;
+ fifo_frames_t framesAvailable = wrappingBuffer.numFrames[partIndex];
+ if (framesAvailable > 0) {
+ if (framesToMix > framesAvailable) {
+ framesToMix = framesAvailable;
+ }
+ mixPart(destination, (float *)wrappingBuffer.data[partIndex], framesToMix, volume);
+
+ destination += framesToMix * mSamplesPerFrame;
+ framesLeft -= framesToMix;
+ }
+ partIndex++;
+ }
+ fifo->getFifoControllerBase()->advanceReadIndex(mFramesPerBurst - framesLeft);
+ if (framesLeft > 0) {
+ ALOGW("AAudioMixer::mix() UNDERFLOW by %d / %d frames ----- UNDERFLOW !!!!!!!!!!",
+ framesLeft, mFramesPerBurst);
+ }
+}
+
+void AAudioMixer::mixPart(float *destination, float *source, int32_t numFrames, float volume) {
+ int32_t numSamples = numFrames * mSamplesPerFrame;
+ // TODO maybe optimize using SIMD
+ for (int sampleIndex = 0; sampleIndex < numSamples; sampleIndex++) {
+ *destination++ += *source++ * volume;
+ }
+}
+
+float *AAudioMixer::getOutputBuffer() {
+ return mOutputBuffer;
+}