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/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
new file mode 100644
index 0000000..93693bd
--- /dev/null
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -0,0 +1,234 @@
+/*
+ * 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 "AAudio"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include <aaudio/AAudio.h>
+
+#include "client/AudioStreamInternalCapture.h"
+#include "utility/AudioClock.h"
+
+using android::WrappingBuffer;
+
+using namespace aaudio;
+
+AudioStreamInternalCapture::AudioStreamInternalCapture(AAudioServiceInterface &serviceInterface,
+ bool inService)
+ : AudioStreamInternal(serviceInterface, inService) {
+
+}
+
+AudioStreamInternalCapture::~AudioStreamInternalCapture() {}
+
+
+// Write the data, block if needed and timeoutMillis > 0
+aaudio_result_t AudioStreamInternalCapture::read(void *buffer, int32_t numFrames,
+ int64_t timeoutNanoseconds)
+{
+ return processData(buffer, numFrames, timeoutNanoseconds);
+}
+
+// Read as much data as we can without blocking.
+aaudio_result_t AudioStreamInternalCapture::processDataNow(void *buffer, int32_t numFrames,
+ int64_t currentNanoTime, int64_t *wakeTimePtr) {
+ aaudio_result_t result = processCommands();
+ if (result != AAUDIO_OK) {
+ return result;
+ }
+
+ if (mAudioEndpoint.isFreeRunning()) {
+ //ALOGD("AudioStreamInternalCapture::processDataNow() - update remote counter");
+ // Update data queue based on the timing model.
+ int64_t estimatedRemoteCounter = mClockModel.convertTimeToPosition(currentNanoTime);
+ // TODO refactor, maybe use setRemoteCounter()
+ mAudioEndpoint.setDataWriteCounter(estimatedRemoteCounter);
+ }
+
+ // If the write index passed the read index then consider it an overrun.
+ if (mAudioEndpoint.getEmptyFramesAvailable() < 0) {
+ mXRunCount++;
+ }
+
+ // Read some data from the buffer.
+ //ALOGD("AudioStreamInternalCapture::processDataNow() - readNowWithConversion(%d)", numFrames);
+ int32_t framesProcessed = readNowWithConversion(buffer, numFrames);
+ //ALOGD("AudioStreamInternalCapture::processDataNow() - tried to read %d frames, read %d",
+ // numFrames, framesProcessed);
+
+ // Calculate an ideal time to wake up.
+ if (wakeTimePtr != nullptr && framesProcessed >= 0) {
+ // By default wake up a few milliseconds from now. // TODO review
+ int64_t wakeTime = currentNanoTime + (1 * AAUDIO_NANOS_PER_MILLISECOND);
+ aaudio_stream_state_t state = getState();
+ //ALOGD("AudioStreamInternalCapture::processDataNow() - wakeTime based on %s",
+ // AAudio_convertStreamStateToText(state));
+ switch (state) {
+ case AAUDIO_STREAM_STATE_OPEN:
+ case AAUDIO_STREAM_STATE_STARTING:
+ break;
+ case AAUDIO_STREAM_STATE_STARTED: // When do we expect the next read burst to occur?
+ {
+ uint32_t burstSize = mFramesPerBurst;
+ if (burstSize < 32) {
+ burstSize = 32; // TODO review
+ }
+
+ uint64_t nextReadPosition = mAudioEndpoint.getDataWriteCounter() + burstSize;
+ wakeTime = mClockModel.convertPositionToTime(nextReadPosition);
+ }
+ break;
+ default:
+ break;
+ }
+ *wakeTimePtr = wakeTime;
+
+ }
+// ALOGD("AudioStreamInternalCapture::readNow finished: now = %llu, read# = %llu, wrote# = %llu",
+// (unsigned long long)currentNanoTime,
+// (unsigned long long)mAudioEndpoint.getDataReadCounter(),
+// (unsigned long long)mAudioEndpoint.getDownDataWriteCounter());
+ return framesProcessed;
+}
+
+aaudio_result_t AudioStreamInternalCapture::readNowWithConversion(void *buffer,
+ int32_t numFrames) {
+ // ALOGD("AudioStreamInternalCapture::readNowWithConversion(%p, %d)",
+ // buffer, numFrames);
+ WrappingBuffer wrappingBuffer;
+ uint8_t *destination = (uint8_t *) buffer;
+ int32_t framesLeft = numFrames;
+
+ mAudioEndpoint.getFullFramesAvailable(&wrappingBuffer);
+
+ // Read data in one or two parts.
+ for (int partIndex = 0; framesLeft > 0 && partIndex < WrappingBuffer::SIZE; partIndex++) {
+ int32_t framesToProcess = framesLeft;
+ int32_t framesAvailable = wrappingBuffer.numFrames[partIndex];
+ if (framesAvailable <= 0) break;
+
+ if (framesToProcess > framesAvailable) {
+ framesToProcess = framesAvailable;
+ }
+
+ int32_t numBytes = getBytesPerFrame() * framesToProcess;
+ int32_t numSamples = framesToProcess * getSamplesPerFrame();
+
+ // TODO factor this out into a utility function
+ if (mDeviceFormat == getFormat()) {
+ memcpy(destination, wrappingBuffer.data[partIndex], numBytes);
+ } else if (mDeviceFormat == AAUDIO_FORMAT_PCM_I16
+ && getFormat() == AAUDIO_FORMAT_PCM_FLOAT) {
+ AAudioConvert_pcm16ToFloat(
+ (const int16_t *) wrappingBuffer.data[partIndex],
+ (float *) destination,
+ numSamples,
+ 1.0f);
+ } else if (mDeviceFormat == AAUDIO_FORMAT_PCM_FLOAT
+ && getFormat() == AAUDIO_FORMAT_PCM_I16) {
+ AAudioConvert_floatToPcm16(
+ (const float *) wrappingBuffer.data[partIndex],
+ (int16_t *) destination,
+ numSamples,
+ 1.0f);
+ } else {
+ ALOGE("Format conversion not supported!");
+ return AAUDIO_ERROR_INVALID_FORMAT;
+ }
+ destination += numBytes;
+ framesLeft -= framesToProcess;
+ }
+
+ int32_t framesProcessed = numFrames - framesLeft;
+ mAudioEndpoint.advanceReadIndex(framesProcessed);
+ incrementFramesRead(framesProcessed);
+
+ //ALOGD("AudioStreamInternalCapture::readNowWithConversion() returns %d", framesProcessed);
+ return framesProcessed;
+}
+
+int64_t AudioStreamInternalCapture::getFramesWritten()
+{
+ int64_t frames =
+ mClockModel.convertTimeToPosition(AudioClock::getNanoseconds())
+ + mFramesOffsetFromService;
+ // Prevent retrograde motion.
+ if (frames < mLastFramesWritten) {
+ frames = mLastFramesWritten;
+ } else {
+ mLastFramesWritten = frames;
+ }
+ //ALOGD("AudioStreamInternalCapture::getFramesWritten() returns %lld", (long long)frames);
+ return frames;
+}
+
+int64_t AudioStreamInternalCapture::getFramesRead()
+{
+ int64_t frames = mAudioEndpoint.getDataWriteCounter()
+ + mFramesOffsetFromService;
+ //ALOGD("AudioStreamInternalCapture::getFramesRead() returns %lld", (long long)frames);
+ return frames;
+}
+
+// Read data from the stream and pass it to the callback for processing.
+void *AudioStreamInternalCapture::callbackLoop() {
+ aaudio_result_t result = AAUDIO_OK;
+ aaudio_data_callback_result_t callbackResult = AAUDIO_CALLBACK_RESULT_CONTINUE;
+ AAudioStream_dataCallback appCallback = getDataCallbackProc();
+ if (appCallback == nullptr) return NULL;
+
+ // result might be a frame count
+ while (mCallbackEnabled.load() && isActive() && (result >= 0)) {
+
+ // Read audio data from stream.
+ int64_t timeoutNanos = calculateReasonableTimeout(mCallbackFrames);
+
+ // This is a BLOCKING READ!
+ result = read(mCallbackBuffer, mCallbackFrames, timeoutNanos);
+ if ((result != mCallbackFrames)) {
+ ALOGE("AudioStreamInternalCapture(): callbackLoop: read() returned %d", result);
+ if (result >= 0) {
+ // Only read some of the frames requested. Must have timed out.
+ result = AAUDIO_ERROR_TIMEOUT;
+ }
+ AAudioStream_errorCallback errorCallback = getErrorCallbackProc();
+ if (errorCallback != nullptr) {
+ (*errorCallback)(
+ (AAudioStream *) this,
+ getErrorCallbackUserData(),
+ result);
+ }
+ break;
+ }
+
+ // Call application using the AAudio callback interface.
+ callbackResult = (*appCallback)(
+ (AAudioStream *) this,
+ getDataCallbackUserData(),
+ mCallbackBuffer,
+ mCallbackFrames);
+
+ if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
+ ALOGD("AudioStreamInternalCapture(): callback returned AAUDIO_CALLBACK_RESULT_STOP");
+ break;
+ }
+ }
+
+ ALOGD("AudioStreamInternalCapture(): callbackLoop() exiting, result = %d, isActive() = %d",
+ result, (int) isActive());
+ return NULL;
+}