/*
 * 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 "AAudioServiceStreamBase"
//#define LOG_NDEBUG 0
#include <utils/Log.h>

#include <iomanip>
#include <iostream>
#include <mutex>

#include <media/MediaMetricsItem.h>
#include <media/TypeConverter.h>

#include "binding/IAAudioService.h"
#include "binding/AAudioServiceMessage.h"
#include "core/AudioGlobal.h"
#include "utility/AudioClock.h"

#include "AAudioEndpointManager.h"
#include "AAudioService.h"
#include "AAudioServiceEndpoint.h"
#include "AAudioServiceStreamBase.h"
#include "TimestampScheduler.h"

using namespace android;  // TODO just import names needed
using namespace aaudio;   // TODO just import names needed

/**
 * Base class for streams in the service.
 * @return
 */

AAudioServiceStreamBase::AAudioServiceStreamBase(AAudioService &audioService)
        : mUpMessageQueue(nullptr)
        , mTimestampThread("AATime")
        , mAtomicStreamTimestamp()
        , mAudioService(audioService) {
    mMmapClient.clientUid = -1;
    mMmapClient.clientPid = -1;
    mMmapClient.packageName = String16("");
}

AAudioServiceStreamBase::~AAudioServiceStreamBase() {
    // May not be set if open failed.
    if (mMetricsId.size() > 0) {
        mediametrics::LogItem(mMetricsId)
                .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_DTOR)
                .set(AMEDIAMETRICS_PROP_STATE, AudioGlobal_convertStreamStateToText(getState()))
                .record();
    }

    // If the stream is deleted when OPEN or in use then audio resources will leak.
    // This would indicate an internal error. So we want to find this ASAP.
    LOG_ALWAYS_FATAL_IF(!(getState() == AAUDIO_STREAM_STATE_CLOSED
                        || getState() == AAUDIO_STREAM_STATE_UNINITIALIZED
                        || getState() == AAUDIO_STREAM_STATE_DISCONNECTED),
                        "service stream %p still open, state = %d",
                        this, getState());
}

std::string AAudioServiceStreamBase::dumpHeader() {
    return std::string("    T   Handle   UId   Port Run State Format Burst Chan Capacity");
}

std::string AAudioServiceStreamBase::dump() const {
    std::stringstream result;

    result << "    0x" << std::setfill('0') << std::setw(8) << std::hex << mHandle
           << std::dec << std::setfill(' ') ;
    result << std::setw(6) << mMmapClient.clientUid;
    result << std::setw(7) << mClientHandle;
    result << std::setw(4) << (isRunning() ? "yes" : " no");
    result << std::setw(6) << getState();
    result << std::setw(7) << getFormat();
    result << std::setw(6) << mFramesPerBurst;
    result << std::setw(5) << getSamplesPerFrame();
    result << std::setw(9) << getBufferCapacity();

    return result.str();
}

void AAudioServiceStreamBase::logOpen(aaudio_handle_t streamHandle) {
    // This is the first log sent from the AAudio Service for a stream.
    mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_STREAM)
            + std::to_string(streamHandle);

    audio_attributes_t attributes = AAudioServiceEndpoint::getAudioAttributesFrom(this);

    // Once this item is logged by the server, the client with the same PID, UID
    // can also log properties.
    mediametrics::LogItem(mMetricsId)
        .setPid(getOwnerProcessId())
        .setUid(getOwnerUserId())
        .set(AMEDIAMETRICS_PROP_ALLOWUID, (int32_t)getOwnerUserId())
        .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_OPEN)
        // the following are immutable
        .set(AMEDIAMETRICS_PROP_BUFFERCAPACITYFRAMES, (int32_t)getBufferCapacity())
        .set(AMEDIAMETRICS_PROP_BURSTFRAMES, (int32_t)getFramesPerBurst())
        .set(AMEDIAMETRICS_PROP_CHANNELCOUNT, (int32_t)getSamplesPerFrame())
        .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(attributes.content_type).c_str())
        .set(AMEDIAMETRICS_PROP_DIRECTION,
                AudioGlobal_convertDirectionToText(getDirection()))
        .set(AMEDIAMETRICS_PROP_ENCODING, toString(getFormat()).c_str())
        .set(AMEDIAMETRICS_PROP_ROUTEDDEVICEID, (int32_t)getDeviceId())
        .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)getSampleRate())
        .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)getSessionId())
        .set(AMEDIAMETRICS_PROP_SOURCE, toString(attributes.source).c_str())
        .set(AMEDIAMETRICS_PROP_USAGE, toString(attributes.usage).c_str())
        .record();
}

aaudio_result_t AAudioServiceStreamBase::open(const aaudio::AAudioStreamRequest &request) {
    AAudioEndpointManager &mEndpointManager = AAudioEndpointManager::getInstance();
    aaudio_result_t result = AAUDIO_OK;

    mMmapClient.clientUid = request.getUserId();
    mMmapClient.clientPid = request.getProcessId();
    mMmapClient.packageName.setTo(String16("")); // TODO What should we do here?

    // Limit scope of lock to avoid recursive lock in close().
    {
        std::lock_guard<std::mutex> lock(mUpMessageQueueLock);
        if (mUpMessageQueue != nullptr) {
            ALOGE("%s() called twice", __func__);
            return AAUDIO_ERROR_INVALID_STATE;
        }

        mUpMessageQueue = new SharedRingBuffer();
        result = mUpMessageQueue->allocate(sizeof(AAudioServiceMessage),
                                           QUEUE_UP_CAPACITY_COMMANDS);
        if (result != AAUDIO_OK) {
            goto error;
        }

        // This is not protected by a lock because the stream cannot be
        // referenced until the service returns a handle to the client.
        // So only one thread can open a stream.
        mServiceEndpoint = mEndpointManager.openEndpoint(mAudioService,
                                                         request);
        if (mServiceEndpoint == nullptr) {
            result = AAUDIO_ERROR_UNAVAILABLE;
            goto error;
        }
        // Save a weak pointer that we will use to access the endpoint.
        mServiceEndpointWeak = mServiceEndpoint;

        mFramesPerBurst = mServiceEndpoint->getFramesPerBurst();
        copyFrom(*mServiceEndpoint);
    }
    return result;

error:
    close();
    return result;
}

aaudio_result_t AAudioServiceStreamBase::close() {
    if (getState() == AAUDIO_STREAM_STATE_CLOSED) {
        return AAUDIO_OK;
    }

    stop();

    aaudio_result_t result = AAUDIO_OK;
    sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
    if (endpoint == nullptr) {
        result = AAUDIO_ERROR_INVALID_STATE;
    } else {
        endpoint->unregisterStream(this);
        AAudioEndpointManager &endpointManager = AAudioEndpointManager::getInstance();
        endpointManager.closeEndpoint(endpoint);

        // AAudioService::closeStream() prevents two threads from closing at the same time.
        mServiceEndpoint.clear(); // endpoint will hold the pointer until this method returns.
    }

    {
        std::lock_guard<std::mutex> lock(mUpMessageQueueLock);
        stopTimestampThread();
        delete mUpMessageQueue;
        mUpMessageQueue = nullptr;
    }

    setState(AAUDIO_STREAM_STATE_CLOSED);

    mediametrics::LogItem(mMetricsId)
        .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_CLOSE)
        .record();
    return result;
}

aaudio_result_t AAudioServiceStreamBase::startDevice() {
    mClientHandle = AUDIO_PORT_HANDLE_NONE;
    sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
    if (endpoint == nullptr) {
        ALOGE("%s() has no endpoint", __func__);
        return AAUDIO_ERROR_INVALID_STATE;
    }
    return endpoint->startStream(this, &mClientHandle);
}

/**
 * Start the flow of audio data.
 *
 * An AAUDIO_SERVICE_EVENT_STARTED will be sent to the client when complete.
 */
aaudio_result_t AAudioServiceStreamBase::start() {
    const int64_t beginNs = AudioClock::getNanoseconds();
    aaudio_result_t result = AAUDIO_OK;

    mediametrics::Defer defer([&] {
        mediametrics::LogItem(mMetricsId)
            .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_START)
            .set(AMEDIAMETRICS_PROP_DURATIONNS, (int64_t)(AudioClock::getNanoseconds() - beginNs))
            .set(AMEDIAMETRICS_PROP_STATE, AudioGlobal_convertStreamStateToText(getState()))
            .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)result)
            .record(); });

    if (isRunning()) {
        return AAUDIO_OK;
    }

    setFlowing(false);
    setSuspended(false);

    // Start with fresh presentation timestamps.
    mAtomicStreamTimestamp.clear();

    mClientHandle = AUDIO_PORT_HANDLE_NONE;
    result = startDevice();
    if (result != AAUDIO_OK) goto error;

    // This should happen at the end of the start.
    sendServiceEvent(AAUDIO_SERVICE_EVENT_STARTED);
    setState(AAUDIO_STREAM_STATE_STARTED);
    mThreadEnabled.store(true);
    result = mTimestampThread.start(this);
    if (result != AAUDIO_OK) goto error;

    return result;

error:
    disconnect();
    return result;
}

aaudio_result_t AAudioServiceStreamBase::pause() {
    const int64_t beginNs = AudioClock::getNanoseconds();
    aaudio_result_t result = AAUDIO_OK;
    if (!isRunning()) {
        return result;
    }

    mediametrics::Defer defer([&] {
        mediametrics::LogItem(mMetricsId)
            .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_PAUSE)
            .set(AMEDIAMETRICS_PROP_DURATIONNS, (int64_t)(AudioClock::getNanoseconds() - beginNs))
            .set(AMEDIAMETRICS_PROP_STATE, AudioGlobal_convertStreamStateToText(getState()))
            .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)result)
            .record(); });

    // Send it now because the timestamp gets rounded up when stopStream() is called below.
    // Also we don't need the timestamps while we are shutting down.
    sendCurrentTimestamp();

    result = stopTimestampThread();
    if (result != AAUDIO_OK) {
        disconnect();
        return result;
    }

    sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
    if (endpoint == nullptr) {
        ALOGE("%s() has no endpoint", __func__);
        result =  AAUDIO_ERROR_INVALID_STATE; // for MediaMetric tracking
        return result;
    }
    result = endpoint->stopStream(this, mClientHandle);
    if (result != AAUDIO_OK) {
        ALOGE("%s() mServiceEndpoint returned %d, %s", __func__, result, getTypeText());
        disconnect(); // TODO should we return or pause Base first?
    }

    sendServiceEvent(AAUDIO_SERVICE_EVENT_PAUSED);
    setState(AAUDIO_STREAM_STATE_PAUSED);
    return result;
}

aaudio_result_t AAudioServiceStreamBase::stop() {
    const int64_t beginNs = AudioClock::getNanoseconds();
    aaudio_result_t result = AAUDIO_OK;
    if (!isRunning()) {
        return result;
    }

    mediametrics::Defer defer([&] {
        mediametrics::LogItem(mMetricsId)
            .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_STOP)
            .set(AMEDIAMETRICS_PROP_DURATIONNS, (int64_t)(AudioClock::getNanoseconds() - beginNs))
            .set(AMEDIAMETRICS_PROP_STATE, AudioGlobal_convertStreamStateToText(getState()))
            .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)result)
            .record(); });

    setState(AAUDIO_STREAM_STATE_STOPPING);

    // Send it now because the timestamp gets rounded up when stopStream() is called below.
    // Also we don't need the timestamps while we are shutting down.
    sendCurrentTimestamp(); // warning - this calls a virtual function
    result = stopTimestampThread();
    if (result != AAUDIO_OK) {
        disconnect();
        return result;
    }

    sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
    if (endpoint == nullptr) {
        ALOGE("%s() has no endpoint", __func__);
        result =  AAUDIO_ERROR_INVALID_STATE; // for MediaMetric tracking
        return result;
    }
    // TODO wait for data to be played out
    result = endpoint->stopStream(this, mClientHandle);
    if (result != AAUDIO_OK) {
        ALOGE("%s() stopStream returned %d, %s", __func__, result, getTypeText());
        disconnect();
        // TODO what to do with result here?
    }

    sendServiceEvent(AAUDIO_SERVICE_EVENT_STOPPED);
    setState(AAUDIO_STREAM_STATE_STOPPED);
    return result;
}

aaudio_result_t AAudioServiceStreamBase::stopTimestampThread() {
    aaudio_result_t result = AAUDIO_OK;
    // clear flag that tells thread to loop
    if (mThreadEnabled.exchange(false)) {
        result = mTimestampThread.stop();
    }
    return result;
}

aaudio_result_t AAudioServiceStreamBase::flush() {
    const int64_t beginNs = AudioClock::getNanoseconds();
    aaudio_result_t result = AAudio_isFlushAllowed(getState());
    if (result != AAUDIO_OK) {
        return result;
    }

    mediametrics::Defer defer([&] {
        mediametrics::LogItem(mMetricsId)
            .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_FLUSH)
            .set(AMEDIAMETRICS_PROP_DURATIONNS, (int64_t)(AudioClock::getNanoseconds() - beginNs))
            .set(AMEDIAMETRICS_PROP_STATE, AudioGlobal_convertStreamStateToText(getState()))
            .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)result)
            .record(); });

    // Data will get flushed when the client receives the FLUSHED event.
    sendServiceEvent(AAUDIO_SERVICE_EVENT_FLUSHED);
    setState(AAUDIO_STREAM_STATE_FLUSHED);
    return AAUDIO_OK;
}

// implement Runnable, periodically send timestamps to client
__attribute__((no_sanitize("integer")))
void AAudioServiceStreamBase::run() {
    ALOGD("%s() %s entering >>>>>>>>>>>>>> TIMESTAMPS", __func__, getTypeText());
    TimestampScheduler timestampScheduler;
    timestampScheduler.setBurstPeriod(mFramesPerBurst, getSampleRate());
    timestampScheduler.start(AudioClock::getNanoseconds());
    int64_t nextTime = timestampScheduler.nextAbsoluteTime();
    int32_t loopCount = 0;
    while(mThreadEnabled.load()) {
        loopCount++;
        if (AudioClock::getNanoseconds() >= nextTime) {
            aaudio_result_t result = sendCurrentTimestamp();
            if (result != AAUDIO_OK) {
                ALOGE("%s() timestamp thread got result = %d", __func__, result);
                break;
            }
            nextTime = timestampScheduler.nextAbsoluteTime();
        } else  {
            // Sleep until it is time to send the next timestamp.
            // TODO Wait for a signal with a timeout so that we can stop more quickly.
            AudioClock::sleepUntilNanoTime(nextTime);
        }
    }
    ALOGD("%s() %s exiting after %d loops <<<<<<<<<<<<<< TIMESTAMPS",
          __func__, getTypeText(), loopCount);
}

void AAudioServiceStreamBase::disconnect() {
    if (getState() != AAUDIO_STREAM_STATE_DISCONNECTED) {
        mediametrics::LogItem(mMetricsId)
            .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_DISCONNECT)
            .set(AMEDIAMETRICS_PROP_STATE, AudioGlobal_convertStreamStateToText(getState()))
            .record();
        sendServiceEvent(AAUDIO_SERVICE_EVENT_DISCONNECTED);
        setState(AAUDIO_STREAM_STATE_DISCONNECTED);
    }
}

aaudio_result_t AAudioServiceStreamBase::sendServiceEvent(aaudio_service_event_t event,
                                                          double  dataDouble) {
    AAudioServiceMessage command;
    command.what = AAudioServiceMessage::code::EVENT;
    command.event.event = event;
    command.event.dataDouble = dataDouble;
    return writeUpMessageQueue(&command);
}

aaudio_result_t AAudioServiceStreamBase::sendServiceEvent(aaudio_service_event_t event,
                                                          int64_t dataLong) {
    AAudioServiceMessage command;
    command.what = AAudioServiceMessage::code::EVENT;
    command.event.event = event;
    command.event.dataLong = dataLong;
    return writeUpMessageQueue(&command);
}

bool AAudioServiceStreamBase::isUpMessageQueueBusy() {
    std::lock_guard<std::mutex> lock(mUpMessageQueueLock);
    if (mUpMessageQueue == nullptr) {
        ALOGE("%s(): mUpMessageQueue null! - stream not open", __func__);
        return true;
    }
    int32_t framesAvailable = mUpMessageQueue->getFifoBuffer()
        ->getFullFramesAvailable();
    int32_t capacity = mUpMessageQueue->getFifoBuffer()
        ->getBufferCapacityInFrames();
    // Is it half full or more
    return framesAvailable >= (capacity / 2);
}

aaudio_result_t AAudioServiceStreamBase::writeUpMessageQueue(AAudioServiceMessage *command) {
    std::lock_guard<std::mutex> lock(mUpMessageQueueLock);
    if (mUpMessageQueue == nullptr) {
        ALOGE("%s(): mUpMessageQueue null! - stream not open", __func__);
        return AAUDIO_ERROR_NULL;
    }
    int32_t count = mUpMessageQueue->getFifoBuffer()->write(command, 1);
    if (count != 1) {
        ALOGW("%s(): Queue full. Did client stop? Suspending stream. what = %u, %s",
              __func__, command->what, getTypeText());
        setSuspended(true);
        return AAUDIO_ERROR_WOULD_BLOCK;
    } else {
        return AAUDIO_OK;
    }
}

aaudio_result_t AAudioServiceStreamBase::sendXRunCount(int32_t xRunCount) {
    return sendServiceEvent(AAUDIO_SERVICE_EVENT_XRUN, (int64_t) xRunCount);
}

aaudio_result_t AAudioServiceStreamBase::sendCurrentTimestamp() {
    AAudioServiceMessage command;
    // It is not worth filling up the queue with timestamps.
    // That can cause the stream to get suspended.
    // So just drop the timestamp if the queue is getting full.
    if (isUpMessageQueueBusy()) {
        return AAUDIO_OK;
    }

    // Send a timestamp for the clock model.
    aaudio_result_t result = getFreeRunningPosition(&command.timestamp.position,
                                                    &command.timestamp.timestamp);
    if (result == AAUDIO_OK) {
        ALOGV("%s() SERVICE  %8lld at %lld", __func__,
              (long long) command.timestamp.position,
              (long long) command.timestamp.timestamp);
        command.what = AAudioServiceMessage::code::TIMESTAMP_SERVICE;
        result = writeUpMessageQueue(&command);

        if (result == AAUDIO_OK) {
            // Send a hardware timestamp for presentation time.
            result = getHardwareTimestamp(&command.timestamp.position,
                                          &command.timestamp.timestamp);
            if (result == AAUDIO_OK) {
                ALOGV("%s() HARDWARE %8lld at %lld", __func__,
                      (long long) command.timestamp.position,
                      (long long) command.timestamp.timestamp);
                command.what = AAudioServiceMessage::code::TIMESTAMP_HARDWARE;
                result = writeUpMessageQueue(&command);
            }
        }
    }

    if (result == AAUDIO_ERROR_UNAVAILABLE) { // TODO review best error code
        result = AAUDIO_OK; // just not available yet, try again later
    }
    return result;
}

/**
 * Get an immutable description of the in-memory queues
 * used to communicate with the underlying HAL or Service.
 */
aaudio_result_t AAudioServiceStreamBase::getDescription(AudioEndpointParcelable &parcelable) {
    {
        std::lock_guard<std::mutex> lock(mUpMessageQueueLock);
        if (mUpMessageQueue == nullptr) {
            ALOGE("%s(): mUpMessageQueue null! - stream not open", __func__);
            return AAUDIO_ERROR_NULL;
        }
        // Gather information on the message queue.
        mUpMessageQueue->fillParcelable(parcelable,
                                        parcelable.mUpMessageQueueParcelable);
    }
    return getAudioDataDescription(parcelable);
}

void AAudioServiceStreamBase::onVolumeChanged(float volume) {
    sendServiceEvent(AAUDIO_SERVICE_EVENT_VOLUME, volume);
}

int32_t AAudioServiceStreamBase::incrementServiceReferenceCount_l() {
    return ++mCallingCount;
}

int32_t AAudioServiceStreamBase::decrementServiceReferenceCount_l() {
    int32_t count = --mCallingCount;
    // Each call to increment should be balanced with one call to decrement.
    assert(count >= 0);
    return count;
}