Sync AAudio flowgraph with Oboe
The flowgraph code in AAudio and Oboe should be kept in parity.
The code has diverged and this cl is meant to sync them again.
Bug: 208689965
Test: Build libaaudio/oboeservice and test changes
Change-Id: I1e0cf04120305ffcb1e64df1930ffa079d3c4a94
diff --git a/media/libaaudio/src/flowgraph/SampleRateConverter.cpp b/media/libaaudio/src/flowgraph/SampleRateConverter.cpp
new file mode 100644
index 0000000..5c3ed1f
--- /dev/null
+++ b/media/libaaudio/src/flowgraph/SampleRateConverter.cpp
@@ -0,0 +1,70 @@
+/*
+ * Copyright 2019 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 "SampleRateConverter.h"
+
+using namespace flowgraph;
+using namespace resampler;
+
+SampleRateConverter::SampleRateConverter(int32_t channelCount, MultiChannelResampler &resampler)
+ : FlowGraphFilter(channelCount)
+ , mResampler(resampler) {
+ setDataPulledAutomatically(false);
+}
+
+void SampleRateConverter::reset() {
+ FlowGraphNode::reset();
+ mInputCursor = kInitialCallCount;
+}
+
+// Return true if there is a sample available.
+bool SampleRateConverter::isInputAvailable() {
+ // If we have consumed all of the input data then go out and get some more.
+ if (mInputCursor >= mNumValidInputFrames) {
+ mInputCallCount++;
+ mNumValidInputFrames = input.pullData(mInputCallCount, input.getFramesPerBuffer());
+ mInputCursor = 0;
+ }
+ return (mInputCursor < mNumValidInputFrames);
+}
+
+const float *SampleRateConverter::getNextInputFrame() {
+ const float *inputBuffer = input.getBuffer();
+ return &inputBuffer[mInputCursor++ * input.getSamplesPerFrame()];
+}
+
+int32_t SampleRateConverter::onProcess(int32_t numFrames) {
+ float *outputBuffer = output.getBuffer();
+ int32_t channelCount = output.getSamplesPerFrame();
+ int framesLeft = numFrames;
+ while (framesLeft > 0) {
+ // Gather input samples as needed.
+ if(mResampler.isWriteNeeded()) {
+ if (isInputAvailable()) {
+ const float *frame = getNextInputFrame();
+ mResampler.writeNextFrame(frame);
+ } else {
+ break;
+ }
+ } else {
+ // Output frame is interpolated from input samples.
+ mResampler.readNextFrame(outputBuffer);
+ outputBuffer += channelCount;
+ framesLeft--;
+ }
+ }
+ return numFrames - framesLeft;
+}