/******************************************************************************
    QtAV:  Multimedia framework based on Qt and FFmpeg
    Copyright (C) 2012-2016 Wang Bin <wbsecg1@gmail.com>

*   This file is part of QtAV

    This library is free software; you can redistribute it and/or
    modify it under the terms of the GNU Lesser General Public
    License as published by the Free Software Foundation; either
    version 2.1 of the License, or (at your option) any later version.

    This library is distributed in the hope that it will be useful,
    but WITHOUT ANY WARRANTY; without even the implied warranty of
    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
    Lesser General Public License for more details.

    You should have received a copy of the GNU Lesser General Public
    License along with this library; if not, write to the Free Software
    Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301  USA
******************************************************************************/

#include "AudioThread.h"
#include "AVThread_p.h"
#include "AudioDecoder.h"
#include "Packet.h"
#include "AudioFormat.h"
#include "AudioOutput.h"
#include "AudioResampler.h"
#include "AVClock.h"
#include "Filter.h"
#include "OutputSet.h"
#include "AVCompat.h"
#include <QtCore/QCoreApplication>
#include <QtCore/QDateTime>
#include "Logger.h"

namespace FAV {

class AudioThreadPrivate : public AVThreadPrivate
{
public:
    void init() {
        resample = false;
        last_pts = 0;
    }

    bool resample;
    qreal last_pts; //used when audio output is not available, to calculate the aproximate sleeping time
};

AudioThread::AudioThread(QObject *parent)
    :AVThread(*new AudioThreadPrivate(), parent)
{
    stopPosition = -1;

}

void AudioThread::applyFilters(AudioFrame &frame)
{
    DPTR_D(AudioThread);
    //QMutexLocker locker(&d.mutex);
    //Q_UNUSED(locker);
    if (!d.filters.isEmpty()) {
        //sort filters by format. vo->defaultFormat() is the last
        foreach (Filter *filter, d.filters) {
            AudioFilter *af = static_cast<AudioFilter*>(filter);
            if (!af->isEnabled())
                continue;
            af->apply(d.statistics, &frame);
        }
    }
}
/*
 *TODO:
 * if output is null or dummy, the use duration to wait
 */

#define LOG_AUDIO_THREAD qInfo // qInfo
//#define CHECK_AUDIO_BREAK 1     //  오디오 종료,끊기는 지점 확인

void AudioThread::run()
{
    DPTR_D(AudioThread);
    //No decoder or output. No audio output is ok, just display picture
    if (!d.dec || !d.dec->isAvailable() || !d.outputSet)
        return;
    resetState();
    Q_ASSERT(d.clock != 0);
    d.init();
    Packet pkt;
    qint64 fake_duration = 0LL;
    qint64 fake_pts = 0LL;
    int sync_id = 0;

#if (FORCE_BREAK_EOF)
    forceEnd = false;
    aboutToEnd = false;
#endif

    while (!d.stop)
    {

#if (USE_DURATION_IN_THREAD)
        usleep(1);  // 32x speed 에서 처리되지 않음
        if(durationSec > 0 && d.clock->value() > durationSec)
        {
            clock()->updateValue(durationSec); //external clock?
            msleep(100);
            break;
        }
#endif
        processNextTask();

#if (FORCE_BREAK_EOF)
        if(forceEnd)
        {
#if (CHECK_AUDIO_BREAK)
            qInfo() << "AUDIO BREAK(forceEnd) LN:" << __LINE__;
#endif
            break;
        }
#endif


        const bool is_external_clock = d.clock->clockType() == AVClock::ExternalClock;
        if (d.render_pts0 < 0) { // no pause when seeking
            if (tryPause()) { //DO NOT continue, or stepForward() will fail
                if (d.stop)
                {
#if (CHECK_AUDIO_BREAK)
                    qInfo() << "AUDIO BREAK(d.stop) LN:" << __LINE__;
#endif
                    break; //the queue is empty and may block. should setBlocking(false) wake up cond empty?
                }
            } else {
                if (isPaused())
                {
#if (CHECK_AUDIO_BREAK)
                    qInfo() << "AUDIO CONTINUE(isPaused()) LN:" << __LINE__;
#endif
                    continue;
                }
            }
        }

        // PAUSE 에도 발생
        if (d.seek_requested)
        {
            d.seek_requested = false;
            // LOG_AUDIO_THREAD("request seek audio thread");
            pkt = Packet(); // last decode failed and pkt is valid, reset pkt to force take the next packet if seek is requested
            msleep(1);
        }
        else
        {
            // d.render_pts0 < 0 means seek finished here
            if (d.clock->syncId() > 0)
            {
                //qInfo("audio thread wait to sync end for sync id: %d", d.clock->syncId());
#if (FORCE_BREAK_EOF)
                if (d.render_pts0 < 0 && sync_id > 0 && aboutToEnd == false) {
#else
                if (d.render_pts0 < 0 && sync_id > 0) {
#endif
                    msleep(10);
#if (CHECK_AUDIO_BREAK)
                    qInfo() << "AUDIO CONTINUE(d.clock->syncId() > 0) LN:" << __LINE__;
#endif
                    continue;
                }
            } else {
                sync_id = 0;
            }
        }
        if (!pkt.isValid())
        {
            // PACKET INVALID 원인 확인
            // qInfo() << "AUDIO PACKET INVALID LN:" << __LINE__ << "isCorrupt:" << pkt.isCorrupt << " empty:" << pkt.data.isEmpty() << " pts:" << pkt.pts << " duration:" << pkt.duration;

            // can't seek back if eof packet is read
            //qDebug("eof pkt: %d valid: %d, aqueue size: %d, abuffer: %d %.3f %d, fake_duration: %lld", pkt.isEOF(), pkt.isValid(), d.packets.size(), d.packets.bufferValue(), d.packets.bufferMax(), d.packets.isFull(), fake_duration);
            // If seek requested but last decode failed
            if (!pkt.isEOF() && (fake_duration <= 0 || !d.packets.isEmpty()))
            {
                pkt = d.packets.take(); //wait to dequeue
                //qInfo() << "AUDIO PTS:" << pkt.dts << " CLOCK:" << d.clock->value();
            }
#if (FORCE_BREAK_EOF)
            if (pkt.isEOF() || forceEnd)
#else
            if (pkt.isEOF())
#endif
            {
                fake_duration = 0; //avoid endless wait
                qDebug("audio thread gets an eof packet. pkt.pts: %.3f, d.render_pts0:%.3f", pkt.pts, d.render_pts0);
            }
            if (!pkt.isValid())
            {
                // check seek first
                if (pkt.pts >= 0)
                {
                    qDebug("Invalid packet! flush audio codec context!!!!!!!! audio queue size=%d", d.packets.size());
                    QMutexLocker locker(&d.mutex);
                    Q_UNUSED(locker);
                    if (d.dec) //maybe set to null in setDecoder()
                    {
                        d.dec->flush();
                    }

                    // render_pts0 지정
                    d.render_pts0 = pkt.pts;
                    sync_id = pkt.position;
                    // LOG_AUDIO_THREAD("audio seek: %.3f, id: %d", d.render_pts0, sync_id);
                    pkt = Packet(); //mark invalid to take next
                    if (fake_duration > 0)
                    {
                        //qInfo("fake_duration update on seek: %ul + %ul - %.3f", fake_duration, fake_pts, d.render_pts0);
                        fake_duration = fake_duration + fake_pts - d.render_pts0*1000.0;
                        fake_pts = d.render_pts0*1000.0;
                    }

                    // External 일 경우 의미 없음
                    // LOG_AUDIO_THREAD() << "d.clock->updateValue(d.render_pts0)";
                    d.clock->updateValue(d.render_pts0);
                    d.clock->updateDelay(0);
                    continue;
                }
                if (pkt.duration > 0)
                {
                    fake_duration = pkt.duration * 1000.0;
                    fake_pts = d.last_pts*1000.0;
                    pkt = Packet(); //mark invalid to avoid run here in the next loop
                    //qDebug("get fake apkt: %.3f+%ul", pkt.pts, fake_duration);

#if (CHECK_AUDIO_BREAK)
                    qInfo() << "AUDIO CONTINUE(pkt.duration > 0) LN:" << __LINE__ << " IS EOF:" << pkt.isEOF() << "forceEnd:" << forceEnd;
#endif
                    continue;
                }
            }

            if (fake_duration > 0)
            {
                static const ulong kSleepMs = 20;
                const ulong ms = qMin<qint64>(fake_duration, kSleepMs);
                fake_duration -= ms;
                fake_pts += ms;
                //LOG_AUDIO_THREAD("fake_wait: %ul, fake_duration: %lld, delay: %.3f", ms, fake_duration, d.clock->delay());
                d.clock->updateDelay(d.clock->delay() + qreal(ms)/1000.0);
                msleep(ms);
#if (CHECK_AUDIO_BREAK)
                qInfo() << "AUDIO CONTINUE(fake_duration > 0) LN:" << __LINE__ << " clock:" << d.clock->value() << " IS EOF:" << pkt.isEOF() << "forceEnd:" << forceEnd;
#endif
                continue;
            }
        }
        if (!pkt.isValid() && !pkt.isEOF()) // decode it will cause crash
        {
#if (CHECK_AUDIO_BREAK)
            qInfo() << "AUDIO CONTINUE(isNotValid+NotEOF) LN:" << __LINE__;
#endif
            continue;
        }


        qreal dts = pkt.dts; //FIXME: pts and dts
        // no key frame for audio. so if pts reaches, try decode and skip render if got frame pts does not reach
        bool skip_render = pkt.pts >= 0 && pkt.pts < d.render_pts0; // if audio stream is too short, seeking will fail and d.render_pts0 keeps >0
        // audio has no key frame, skip rendering equals to skip decoding
        // SEEK
        if (skip_render) {

            d.clock->updateValue(pkt.pts);

            // audio may be too fast than video if skip without sleep
            // a frame is about 20ms. sleep time must be << frame time
            qreal a_v = dts - d.clock->videoTime();

            // qInfo("skip audio decode at %f/%f v=%f a-v=%fms", dts, d.render_pts0, d.clock->videoTime(), a_v*1000.0);
            if (a_v > 0)
            {
                // LOG_AUDIO_THREAD() << "a_v:" << a_v;
                msleep(qMin((ulong)20, ulong(a_v*1000.0)));
                d.clock->updateValue(a_v);
            } else {
                // audio maybe too late compared with video packet before seeking backword. so just ignore
                // 64 로 처리하면 영상 딜레이가 발생함..!!!!!!
#if (MODEL_BBVIEWER)
                msleep(0); //original
#else
                msleep(0); //wait video seek done if audio done early
#endif
            }
            pkt = Packet(); //mark invalid to take next
#if (CHECK_AUDIO_BREAK)
            qInfo() << "AUDIO CONTINUE(skip_render) LN:" << __LINE__;
#endif
            continue;
        }
//        const bool is_external_clock = d.clock->clockType() == AVClock::ExternalClock;
        if (is_external_clock && !pkt.isEOF()) {
            d.delay = dts - d.clock->value();

            // qInfo() << "dts:" << dts << " d.delay:" << d.delay;

//          after seeking forward, a packet may be the old, v packet may be
//          the new packet, then the d.delay is very large, omit it.
//          TODO: 1. how to choose the value
//           2. use last delay when seeking
            // 오차가 작을경우
            if (qAbs(d.delay) < 2.0) {

                //qInfo() << "d.delay+:" << d.delay;
                // 느릴 경우 (뒤로 이동)
                if (d.delay < -kSyncThreshold) { //Speed up. drop frame? resample?

                    //qInfo() << "d.delay < -kSyncThreshold";
                    //msleep(1); // 추가 (노이즈 방지)
                    //qInfo("audio is late compared with external clock. skip decoding. %.3f-%.3f=%.3f", dts, d.clock->value(), d.delay);
                    pkt = Packet(); //mark invalid to take next
#if (CHECK_AUDIO_BREAK)
                    qInfo() << "AUDIO CONTINUE(qAbs(d.delay) < 2.0) LN:" << __LINE__;
#endif
                    continue;
                }
                // 빠를 경우 (전방이동)
                if (d.delay > 0)
                {
                    // 아래버그 처리하니 DELAY 는 해결됨 -> 노이즈가 생김...
                    //qInfo() << "!!>>>>W" << d.delay << dts << __FUNCTION__ << __LINE__;
                    waitAndCheck(d.delay, dts); //-> 버그로 보임
                    //waitAndCheck(ulong(d.delay * 100.0), dts);
                    //waitAndCheck(ulong(d.delay * 1000.0), dts);
                }
            }
            else { //when to drop off?

                // 오차가 클 경우 (SEEK)

                if (d.delay > 0)  // 전방이동
                {
                    msleep(64);

                    //qInfo() << "DELAY > 0 BIG" << d.delay;
                }
                else // 후방이동
                {
                    // DELAY 가 계속 커지면서 (-) 문제가 발생함
#if (MODEL_BBVIEWER)
                    //qInfo() << "backward d.delay-:" << d.delay;
                    msleep(32);
#else

                    // 128 일 경우 전방 이동시 부드럽게 시작?? 하지 않는다.
                    msleep(64); // 추가 (노이즈 방지) -> SLEEP 하지 않으면 VideoThread 처리가 늦어짐
#endif
                    //audio packet not cleaned up?
                    qDebug("audio is too late compared with external clock. skip decoding. %.3f-%.3f=%.3f", dts, d.clock->value(), d.delay);
                    pkt = Packet(); //mark invalid to take next
                    //qInfo() << "DELAY < 0 BIG" << d.delay;
#if (CHECK_AUDIO_BREAK)
                    qInfo() << "AUDIO CONTINUE(qAbs(d.delay) > 2.0) LN:" << __LINE__;
#endif
                    continue;
                }
            }
        }

        /* lock here to ensure decoder and ao can complete current work before they are changed
         * current packet maybe not supported by new decoder
         */
        // TODO: smaller scope
        QMutexLocker locker(&d.mutex);
        Q_UNUSED(locker);
        AudioDecoder *dec = static_cast<AudioDecoder*>(d.dec);
        if (!dec) {
            pkt = Packet(); //mark invalid to take next
#if (CHECK_AUDIO_BREAK)
            qInfo() << "AUDIO CONTINUE(!dec) LN:" << __LINE__;
#endif
            continue;
        }
        AudioOutput *ao = 0;
        // first() is not null even if list empty
        if (!d.outputSet->outputs().isEmpty())
        {
            ao = static_cast<AudioOutput*>(d.outputSet->outputs().first());
        }

        //DO NOT decode and convert if ao is not available or mute!
        bool has_ao = ao && ao->isAvailable();


        //if (!has_ao) {//do not decode?
        // TODO: move resampler to AudioFrame, like VideoFrame does
        if (has_ao && dec->resampler()) {
            if (dec->resampler()->speed() != ao->speed()
                    || dec->resampler()->outAudioFormat() != ao->audioFormat()) {
                //resample later to ensure thread safe. TODO: test
                if (d.resample) {
                    qDebug() << "ao.format " << ao->audioFormat();
                    qDebug() << "swr.format " << dec->resampler()->outAudioFormat();
                    qDebug("decoder set speed: %.2f", ao->speed());
                    dec->resampler()->setOutAudioFormat(ao->audioFormat());
                    dec->resampler()->setSpeed(ao->speed());
                    dec->resampler()->prepare();
                    d.resample = false;
                } else {
                    d.resample = true;
                }
            }
        } else {
            if (dec->resampler() && dec->resampler()->speed() != d.clock->speed()) {
                if (d.resample) {
                    qDebug("decoder set speed: %.2f", d.clock->speed());
                    dec->resampler()->setSpeed(d.clock->speed());
                    dec->resampler()->prepare();
                    d.resample = false;
                } else {
                    d.resample = true;
                }
            }
        }
        if (d.stop) {
            qDebug("audio thread stop before decode()");
#if (CHECK_AUDIO_BREAK)
            qInfo() << "AUDIO BREAK(d.stop) LN:" << __LINE__;
#endif
            break;
        }
        //qDebug("apkt: %.3f, %lld %p", pkt.pts, pkt.asAVPacket()->pts, pkt.asAVPacket()->data);
        if (!dec->decode(pkt)) {
            qWarning("Decode audio failed. undecoded: %d", dec->undecodedSize());
            if (pkt.isEOF()) {
                qDebug("audio decode eof done");
                Q_EMIT eofDecoded();
                if (d.render_pts0 >= 0) {
                    qDebug("audio seek done at eof pts: %.3f. id: %d", pkt.pts, sync_id);
                    d.render_pts0 = -1;
                    d.clock->syncEndOnce(sync_id);
#if (PLAY_SYNC_FIX2)
                    Q_EMIT seekFinished(qint64(pkt.pts*1000.0),qint64(d.render_pts0*1000.0)); //TODO: pts
#else
                    Q_EMIT seekFinished(qint64(pkt.pts*1000.0)); //TODO: pts
#endif
                }
                if (!pkt.position)
                {
#if (CHECK_AUDIO_BREAK)
                    qInfo() << "AUDIO BREAK(!pkt.position) LN:" << __LINE__;
#endif
                    break;
                }
            }
            qreal dt = dts - d.last_pts;
            if (dt > 0.5 || dt < 0) {
                dt = 0;
            }
            if (!qFuzzyIsNull(dt)) {
                msleep((unsigned long)(dt*1000.0));
            }
            pkt = Packet();
            d.last_pts = d.clock->value(); //not pkt.pts! the delay is updated!
#if (CHECK_AUDIO_BREAK)
            qInfo() << "AUDIO CONTINUE(!dec->decode(pkt)) LN:" << __LINE__;
#endif
            continue;
        }
        // reduce here to ensure to decode the rest data in the next loop
        if (!pkt.isEOF()) {
            pkt.skip(pkt.data.size() - dec->undecodedSize());
        }
#if (USE_AUDIO_FRAME)
        AudioFrame frame(dec->frame());
        if (!frame)
        {
#if (CHECK_AUDIO_BREAK)
            qInfo() << "AUDIO CONTINUE(!frame) LN:" << __LINE__;
#endif // CHECK_AUDIO_BREAK
            continue; //pkt data is updated after decode, no reset here
        }
        if (frame.pts() <= 0)
        {
            frame.setPTS(pkt.pts); // pkt.pts is wrong. >= real timestamp
        }
        if (d.render_pts0 >= 0.0) { // seeking
            d.clock->updateValue(frame.pts());

            if (frame.pts() < d.render_pts0) {
                qDebug("skip audio rendering: %f-%f", frame.pts(), d.render_pts0);
#if (CHECK_AUDIO_BREAK)
            qInfo() << "AUDIO CONTINUE(frame.timestamp() < d.render_pts0) LN:" << __LINE__;
#endif // CHECK_AUDIO_BREAK
                continue; //pkt data is updated after decode, no reset here
            }
            qDebug("audio seek finished @%.3f. id: %d", frame.pts(), sync_id);
#if (PLAY_SYNC_FIX2)
            qreal requested = d.render_pts0;
#endif // PLAY_SYNC_FIX2
            d.render_pts0 = -1.0;
            d.clock->syncEndOnce(sync_id);
#if (PLAY_SYNC_FIX2)
            Q_EMIT seekFinished(qint64(frame.pts()*1000.0),qint64(requested*1000.0));
#else // PLAY_SYNC_FIX2
            Q_EMIT seekFinished(qint64(frame.pts()*1000.0));
#endif // PLAY_SYNC_FIX2
            if (has_ao) {
                ao->clear();
            }
        }
        if (has_ao) {
            applyFilters(frame);
            frame.setAudioResampler(dec->resampler()); //!!!
            // FIXME: resample ONCE is required for audio frames from ffmpeg
            //if (ao->audioFormat() != frame.format()) {
                frame = frame.to(ao->audioFormat());
            //}
        }

#if (FIXED_FPS_DURATION)
        bool isStopPosition = (stopPosition > 0 && (qint64)(frame.pts() * 1000000) >= stopPosition);
        // 후방 깨진영역 플레이 하지 않는다??
        if(isStopPosition == true && playerID == 0)
        {
#if (CHECK_AUDIO_BREAK)
            qInfo() << "AUDIO CONTINUE(isStopPosition) LN:" << __LINE__;
#endif // CHECK_AUDIO_BREAK
            msleep(1);
            continue;
        }
#endif // FIXED_FPS_DURATION

        QByteArray decoded(frame.data());
#else // AUDIO FRAME
        QByteArray decoded(dec->data());
#endif // AUDIO FRAME
        int decodedSize = decoded.size();
        int decodedPos = 0;
        qreal delay = 0;
        const qreal byte_rate = frame.format().bytesPerSecond();
        qreal pts = frame.pts();
        //qDebug("frame samples: %d @%.3f+%lld", frame.samplesPerChannel()*frame.channelCount(), frame.timestamp(), frame.duration()/1000LL);
        while (decodedSize > 0) {
            if (d.stop)
            {
#if (CHECK_AUDIO_BREAK)
                qInfo() << "AUDIO BREAK(d.stop) LN:" << __LINE__;
#endif
                break;
            }
            const int chunk = qMin(decodedSize, has_ao ? ao->bufferSize() : 512*frame.format().bytesPerFrame());//int(max_len*byte_rate));
            //AudioFormat.bytesForDuration
            const qreal chunk_delay = (qreal)chunk/(qreal)byte_rate;

            if (has_ao && ao->isOpen()) {
                QByteArray decodedChunk = QByteArray::fromRawData(decoded.constData() + decodedPos, chunk);

                //LOG_AUDIO_THREAD() << "A:" << ao->timestamp();
                //ao->play(decodedChunk, pts);
                // qInfo() << playerID << "pts" << pts << "duration" << duration << __FUNCTION__;

                // 사운드가 재생이 안될 경우 duration 확인
                if(pts < duration) {
                    ao->play(decodedChunk, pts);
                    // qInfo() << "SOUND PLAY:" << ao->play(decodedChunk, pts);
                }

                // PAUSE + PLAY 시 timestamp 가 0.1초 정도 (전,후방) 이동함
                // 7.794 -> 0000 -> 7.691
                //qInfo("ao.timestamp: %.3f, pts: %.3f, pktpts: %.3f", ao->timestamp(), pts, pkt.pts);

                //if (!is_external_clock && ao->timestamp() > 0)
                if (!is_external_clock && ao->timestamp() > 0)
                {   //TODO: clear ao buffer
                   // const qreal da = qAbs(pts - ao->timestamp());
                   // if (da > 1.0) { // what if frame duration is long?
                   // }
                    // TODO: check seek_requested(atomic bool)
                    d.clock->updateValue(ao->timestamp());
                }
            }
            else // 발생하지 않는다.
            {
                d.clock->updateDelay(delay += chunk_delay);
                // why need this even if we add delay? and usleep sounds weird
                // the advantage is if no audio device, the play speed is ok too
                // So is portaudio blocking the thread when playing?
                // TODO: avoid acummulative error. External clock?
                msleep((unsigned long)(chunk_delay * 1000.0));

                // 누적에러 처리???
                // LOG_AUDIO_THREAD() << "chunk_delay:" << chunk_delay;
            }
            decodedPos += chunk;
            decodedSize -= chunk;
            pts += chunk_delay;
            pkt.pts += chunk_delay; // packet not fully decoded, use new pts in the next decoding
            pkt.dts += chunk_delay;
        }
        if (has_ao)
        {
            emit frameDelivered();
        }
        d.last_pts = d.clock->value(); //not pkt.pts! the delay is updated!
        //LOG_AUDIO_THREAD() << "d.last_pts:" << d.last_pts;
    }
    d.packets.clear();

#if (CHECK_AUDIO_BREAK)
    qInfo() << "AUDIO END LN:" << __LINE__;
#endif


    qDebug("Audio thread stops running...");
    //qInfo() << "AUDIO STOP!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!";

    //FIXED_SLEEP; //qInfo() << "EXIT THREAD 0 AUDIO" << __FUNCTION__ << __LINE__;
}

} //namespace FAV