liboboe: rename Oboe to AAudio

All of the edits were done using scripts in media/liboboe/scripts.
The conversion is done using SED, which is called from
convert_oboe_aaudio.sh
The conversion can be reverted when debugging using revert_all_aaudio.sh
The string substitutions are in oboe_to_aaudio.sed

Bug: 34749573
Test: cts/tests/tests/nativemedia/aaudio

Change-Id: Ia10b34472a90df2652b87607c99156e9084e57aa
Signed-off-by: Phil Burk <philburk@google.com>
diff --git a/services/oboeservice/AAudioServiceStreamFakeHal.cpp b/services/oboeservice/AAudioServiceStreamFakeHal.cpp
new file mode 100644
index 0000000..627a504
--- /dev/null
+++ b/services/oboeservice/AAudioServiceStreamFakeHal.cpp
@@ -0,0 +1,202 @@
+/*
+ * Copyright (C) 2016 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 <atomic>
+
+#include "AudioClock.h"
+#include "AudioEndpointParcelable.h"
+
+#include "AAudioServiceStreamBase.h"
+#include "AAudioServiceStreamFakeHal.h"
+
+#include "FakeAudioHal.h"
+
+using namespace android;
+using namespace aaudio;
+
+// HACK values for Marlin
+#define CARD_ID              0
+#define DEVICE_ID            19
+
+/**
+ * Construct the audio message queuues and message queues.
+ */
+
+AAudioServiceStreamFakeHal::AAudioServiceStreamFakeHal()
+        : AAudioServiceStreamBase()
+        , mStreamId(nullptr)
+        , mPreviousFrameCounter(0)
+        , mAAudioThread()
+{
+}
+
+AAudioServiceStreamFakeHal::~AAudioServiceStreamFakeHal() {
+    ALOGD("AAudioServiceStreamFakeHal::~AAudioServiceStreamFakeHal() call close()");
+    close();
+}
+
+aaudio_result_t AAudioServiceStreamFakeHal::open(aaudio::AAudioStreamRequest &request,
+                                             aaudio::AAudioStreamConfiguration &configuration) {
+    // Open stream on HAL and pass information about the ring buffer to the client.
+    mmap_buffer_info mmapInfo;
+    aaudio_result_t error;
+
+    // Open HAL
+    error = fake_hal_open(CARD_ID, DEVICE_ID, &mStreamId);
+    if(error < 0) {
+        ALOGE("Could not open card %d, device %d", CARD_ID, DEVICE_ID);
+        return error;
+    }
+
+    // Get information about the shared audio buffer.
+    error = fake_hal_get_mmap_info(mStreamId, &mmapInfo);
+    if (error < 0) {
+        ALOGE("fake_hal_get_mmap_info returned %d", error);
+        fake_hal_close(mStreamId);
+        mStreamId = nullptr;
+        return error;
+    }
+    mHalFileDescriptor = mmapInfo.fd;
+    mFramesPerBurst = mmapInfo.burst_size_in_frames;
+    mCapacityInFrames = mmapInfo.buffer_capacity_in_frames;
+    mCapacityInBytes = mmapInfo.buffer_capacity_in_bytes;
+    mSampleRate = mmapInfo.sample_rate;
+    mBytesPerFrame = mmapInfo.channel_count * sizeof(int16_t); // FIXME based on data format
+    ALOGD("AAudioServiceStreamFakeHal::open() mmapInfo.burst_size_in_frames = %d",
+         mmapInfo.burst_size_in_frames);
+    ALOGD("AAudioServiceStreamFakeHal::open() mmapInfo.buffer_capacity_in_frames = %d",
+         mmapInfo.buffer_capacity_in_frames);
+    ALOGD("AAudioServiceStreamFakeHal::open() mmapInfo.buffer_capacity_in_bytes = %d",
+         mmapInfo.buffer_capacity_in_bytes);
+
+    // Fill in AAudioStreamConfiguration
+    configuration.setSampleRate(mSampleRate);
+    configuration.setSamplesPerFrame(mmapInfo.channel_count);
+    configuration.setAudioFormat(AAUDIO_FORMAT_PCM_I16);
+
+    return AAUDIO_OK;
+}
+
+/**
+ * Get an immutable description of the in-memory queues
+ * used to communicate with the underlying HAL or Service.
+ */
+aaudio_result_t AAudioServiceStreamFakeHal::getDescription(AudioEndpointParcelable &parcelable) {
+    // Gather information on the message queue.
+    mUpMessageQueue->fillParcelable(parcelable,
+                                    parcelable.mUpMessageQueueParcelable);
+
+    // Gather information on the data queue.
+    // TODO refactor into a SharedRingBuffer?
+    int fdIndex = parcelable.addFileDescriptor(mHalFileDescriptor, mCapacityInBytes);
+    parcelable.mDownDataQueueParcelable.setupMemory(fdIndex, 0, mCapacityInBytes);
+    parcelable.mDownDataQueueParcelable.setBytesPerFrame(mBytesPerFrame);
+    parcelable.mDownDataQueueParcelable.setFramesPerBurst(mFramesPerBurst);
+    parcelable.mDownDataQueueParcelable.setCapacityInFrames(mCapacityInFrames);
+    return AAUDIO_OK;
+}
+
+/**
+ * Start the flow of data.
+ */
+aaudio_result_t AAudioServiceStreamFakeHal::start() {
+    if (mStreamId == nullptr) return AAUDIO_ERROR_NULL;
+    aaudio_result_t result = fake_hal_start(mStreamId);
+    sendServiceEvent(AAUDIO_SERVICE_EVENT_STARTED);
+    mState = AAUDIO_STREAM_STATE_STARTED;
+    if (result == AAUDIO_OK) {
+        mThreadEnabled.store(true);
+        result = mAAudioThread.start(this);
+    }
+    return result;
+}
+
+/**
+ * Stop the flow of data such that start() can resume with loss of data.
+ */
+aaudio_result_t AAudioServiceStreamFakeHal::pause() {
+    if (mStreamId == nullptr) return AAUDIO_ERROR_NULL;
+    sendCurrentTimestamp();
+    aaudio_result_t result = fake_hal_pause(mStreamId);
+    sendServiceEvent(AAUDIO_SERVICE_EVENT_PAUSED);
+    mState = AAUDIO_STREAM_STATE_PAUSED;
+    mFramesRead.reset32();
+    ALOGD("AAudioServiceStreamFakeHal::pause() sent AAUDIO_SERVICE_EVENT_PAUSED");
+    mThreadEnabled.store(false);
+    result = mAAudioThread.stop();
+    return result;
+}
+
+/**
+ *  Discard any data held by the underlying HAL or Service.
+ */
+aaudio_result_t AAudioServiceStreamFakeHal::flush() {
+    if (mStreamId == nullptr) return AAUDIO_ERROR_NULL;
+    // TODO how do we flush an MMAP/NOIRQ buffer? sync pointers?
+    ALOGD("AAudioServiceStreamFakeHal::pause() send AAUDIO_SERVICE_EVENT_FLUSHED");
+    sendServiceEvent(AAUDIO_SERVICE_EVENT_FLUSHED);
+    mState = AAUDIO_STREAM_STATE_FLUSHED;
+    return AAUDIO_OK;
+}
+
+aaudio_result_t AAudioServiceStreamFakeHal::close() {
+    aaudio_result_t result = AAUDIO_OK;
+    if (mStreamId != nullptr) {
+        result = fake_hal_close(mStreamId);
+        mStreamId = nullptr;
+    }
+    return result;
+}
+
+void AAudioServiceStreamFakeHal::sendCurrentTimestamp() {
+    int frameCounter = 0;
+    int error = fake_hal_get_frame_counter(mStreamId, &frameCounter);
+    if (error < 0) {
+        ALOGE("AAudioServiceStreamFakeHal::sendCurrentTimestamp() error %d",
+                error);
+    } else if (frameCounter != mPreviousFrameCounter) {
+        AAudioServiceMessage command;
+        command.what = AAudioServiceMessage::code::TIMESTAMP;
+        mFramesRead.update32(frameCounter);
+        command.timestamp.position = mFramesRead.get();
+        ALOGD("AAudioServiceStreamFakeHal::sendCurrentTimestamp() HAL frames = %d, pos = %d",
+                frameCounter, (int)mFramesRead.get());
+        command.timestamp.timestamp = AudioClock::getNanoseconds();
+        mUpMessageQueue->getFifoBuffer()->write(&command, 1);
+        mPreviousFrameCounter = frameCounter;
+    }
+}
+
+// implement Runnable
+void AAudioServiceStreamFakeHal::run() {
+    TimestampScheduler timestampScheduler;
+    timestampScheduler.setBurstPeriod(mFramesPerBurst, mSampleRate);
+    timestampScheduler.start(AudioClock::getNanoseconds());
+    while(mThreadEnabled.load()) {
+        aaudio_nanoseconds_t nextTime = timestampScheduler.nextAbsoluteTime();
+        if (AudioClock::getNanoseconds() >= nextTime) {
+            sendCurrentTimestamp();
+        } else  {
+            // Sleep until it is time to send the next timestamp.
+            AudioClock::sleepUntilNanoTime(nextTime);
+        }
+    }
+}
+