/* **********
 *
 * This software is released under the provisions of the GPL version 2.
 * see file "COPYING".  If that file is not available, the full statement
 * of the license can be found at
 *
 * http://www.fsf.org/licensing/licenses/gpl.txt
 *
 * Copyright (c) Paul Cifarelli 2005
 * Portions Copyright (c) 1995-2002 RealNetworks, Inc. All Rights Reserved.
 * PCM time-domain equalizer:
 *    (c) 2002 Felipe Rivera <liebremx at users sourceforge net>
 *    (c) 2004 Mark Kretschmann <markey@web.de>
 *
 * ********** */
#include <math.h>
#include <stdlib.h>

#include "hxcomm.h"
#include "hxcore.h"
#include "hxprefs.h"
#include "hxstrutl.h"
#include "hxvsrc.h"
#include "hxresult.h"
#include "hxausvc.h"
#include "helix-sp.h"
#ifndef HELIX_SW_VOLUME_INTERFACE
#include "gain.h"
#endif
#include "hsphook.h"
#include "iir_cf.h"         // IIR filter coefficients
#include "hspalsadevice.h"

#define SCOPESIZE 512

HSPPreMixAudioHook::HSPPreMixAudioHook(HelixSimplePlayer *player, int playerIndex, IHXAudioStream *pAudioStream,
                                       bool fadein, unsigned long fadelength) :
   m_Player(player), m_lRefCount(0), m_index(playerIndex), m_stream(pAudioStream), m_count(0),
   m_gaintool(0), m_gaindb(0), m_fadein(fadein), m_fadeout(false), m_fadelength(fadelength)
{
   AddRef();
}

HSPPreMixAudioHook::~HSPPreMixAudioHook()
{
   if (m_gaintool)
      gainFree(m_gaintool);
}

STDMETHODIMP
HSPPreMixAudioHook::QueryInterface(REFIID riid, void**ppvObj)
{
    if(IsEqualIID(riid, IID_IUnknown))
    {
        AddRef();
        *ppvObj = (IUnknown*)(IHXAudioHook *)this;
        return HXR_OK;
    }
    else if(IsEqualIID(riid, IID_IHXAudioHook))
    {
        AddRef();
        *ppvObj = (IHXAudioHook *)this;
        return HXR_OK;
    }
    *ppvObj = NULL;
    return HXR_NOINTERFACE;
}

STDMETHODIMP_(UINT32)
HSPPreMixAudioHook::AddRef()
{
    return InterlockedIncrement(&m_lRefCount);
}

STDMETHODIMP_(UINT32)
HSPPreMixAudioHook::Release()
{
    if (InterlockedDecrement(&m_lRefCount) > 0)
    {
        return m_lRefCount;
    }

    delete this;
    return 0;
}

int HSPPreMixAudioHook::volumeize(unsigned char *data, unsigned char *outbuf, size_t len)
{
   gainFeed(data, outbuf, len, m_gaintool);

   return len;
}

void HSPPreMixAudioHook::setFadeout(bool fadeout)
{
   m_fadeout = fadeout;
   if (m_fadeout)
   {
      // the "time constant" (ms) is the time it takes to reach +/- 6db of the original
      gainSetTimeConstant((float) m_fadelength / 8.0, m_gaintool);
      gainSetSmoothdB(FADE_MIN_dB, m_gaintool);
   }
}

STDMETHODIMP HSPPreMixAudioHook::OnBuffer(HXAudioData *pAudioInData, HXAudioData *pAudioOutData)
{
   m_count++;

#ifdef DEBUG_PURPOSES_ONLY
   if (!(m_count % 100))
   {
      m_Player->print2stderr("PRE: time: %d  ", pAudioInData->ulAudioTime);
      switch (pAudioInData->uAudioStreamType)
      {
         case INSTANTANEOUS_AUDIO:
            m_Player->print2stderr(" INSTANTANEOUS_AUDIO ");
            break;
         case STREAMING_AUDIO:
            m_Player->print2stderr(" STREAMING_AUDIO ");
            break;
         case TIMED_AUDIO:
            m_Player->print2stderr(" TIMED_AUDIO ");
            break;
         case STREAMING_INSTANTANEOUS_AUDIO:
            m_Player->print2stderr(" STREAMING_INSTANTANEOUS_AUDIO ");
            break;
      }
      m_Player->print2stderr("pAudioOutData %lx, data %lx\n", pAudioOutData, pAudioOutData->pData);
   }
#endif

   unsigned char *outbuf;
   IHXBuffer *ibuf;
   unsigned long len;
   unsigned char *data;

   pAudioInData->pData->Get(data, len);

   // provide a little margin to prevent a slight but noticeable jump in vol when the fadein ends
   if ((m_fadein && pAudioInData->ulAudioTime < 2*m_fadelength) || m_fadeout)
   {
      m_Player->pCommonClassFactory->CreateInstance(CLSID_IHXBuffer, (void **) &ibuf);
      if (ibuf)
      {
         ibuf->SetSize(len);
         outbuf = ibuf->GetBuffer();

         len = volumeize(data, outbuf, len);

         pAudioOutData->pData = ibuf;
         pAudioOutData->ulAudioTime = pAudioInData->ulAudioTime;
         pAudioOutData->uAudioStreamType = pAudioInData->uAudioStreamType;
      }
   }

   return 0;
}

STDMETHODIMP HSPPreMixAudioHook::OnInit(HXAudioFormat *pFormat)
{
   m_Player->print2stderr("PRE MIX HOOK OnInit AudioFormat: ch %d, bps %d, sps %ld, mbs %d\n", pFormat->uChannels,
          pFormat->uBitsPerSample,
          pFormat->ulSamplesPerSec,
          pFormat->uMaxBlockSize);

   m_format = *pFormat;

   int bps = pFormat->uBitsPerSample / 8;
   m_gaintool = gainInit(pFormat->ulSamplesPerSec, pFormat->uChannels, bps);
   gainSetImmediatedB(0, m_gaintool);

   if (m_fadein)
   {
      gainSetImmediatedB(FADE_MIN_dB, m_gaintool);
      // the "time constant" (ms) is the time it takes to reach -6db of the target
      gainSetTimeConstant((float) m_fadelength / 2.0, m_gaintool);
      gainSetSmoothdB(0, m_gaintool);
   }

   return 0;
}


///////////////////////////////////////////////////////////////////////////////////////////////////

HSPPostProcessor::HSPPostProcessor(HelixSimplePlayer *player, int playerIndex) :
   m_Player(player), m_lRefCount(0), m_index(playerIndex), m_count(0), m_item(0),
   m_current(0), m_prevtime(0), m_i(0), m_j(2), m_k(1)
#ifndef HELIX_SW_VOLUME_INTERFACE
   , m_gaintool(0), m_gaindB(0.0)
#endif
{
   AddRef();
   memset(&m_format, 0, sizeof(m_format));

   // zero the data_history, to eliminate the buzz on playing the first track after enabling the equalizer
   memset(&data_history, 0, sizeof(data_history));
}

HSPPostProcessor::~HSPPostProcessor()
{
#ifndef HELIX_SW_VOLUME_INTERFACE
   if (m_gaintool)
      gainFree(m_gaintool);
#endif
}

STDMETHODIMP
HSPPostProcessor::QueryInterface(REFIID riid, void**ppvObj)
{
    if(IsEqualIID(riid, IID_IUnknown))
    {
        AddRef();
        *ppvObj = (IUnknown*)(IHXAudioHook *)this;
        return HXR_OK;
    }
    else if(IsEqualIID(riid, IID_IHXAudioHook))
    {
        AddRef();
        *ppvObj = (IHXAudioHook *)this;
        return HXR_OK;
    }
    *ppvObj = NULL;
    return HXR_NOINTERFACE;
}

STDMETHODIMP_(UINT32)
HSPPostProcessor::AddRef()
{
    return InterlockedIncrement(&m_lRefCount);
}

STDMETHODIMP_(UINT32)
HSPPostProcessor::Release()
{
    if (InterlockedDecrement(&m_lRefCount) > 0)
    {
        return m_lRefCount;
    }

    delete this;
    return 0;
}

STDMETHODIMP HSPPostProcessor::OnBuffer(HXAudioData *pAudioInData, HXAudioData *pAudioOutData)
{
   unsigned long len;
   unsigned char *data;

   pAudioInData->pData->Get(data, len);

   m_count++;

#ifdef DEBUG_PURPOSES_ONLY
   if (!(m_count % 100))
   {
      m_Player->print2stderr("POST: time: %d  ", pAudioInData->ulAudioTime);
      switch (pAudioInData->uAudioStreamType)
      {
         case INSTANTANEOUS_AUDIO:
            m_Player->print2stderr(" INSTANTANEOUS_AUDIO ");
            break;
         case STREAMING_AUDIO:
            m_Player->print2stderr(" STREAMING_AUDIO ");
            break;
         case TIMED_AUDIO:
            m_Player->print2stderr(" TIMED_AUDIO ");
            break;
         case STREAMING_INSTANTANEOUS_AUDIO:
            m_Player->print2stderr(" STREAMING_INSTANTANEOUS_AUDIO ");
            break;
      }
      m_Player->print2stderr("len %d\n", len);
      m_Player->print2stderr("pAudioOutData %lx, data %lx\n", pAudioOutData, pAudioOutData->pData);

      m_Player->print2stderr("Volume is %d\n",m_Player->getDirectHWVolume());
   }

#endif


#ifndef HELIX_SW_VOLUME_INTERFACE
   unsigned char *outbuf;
   IHXBuffer *ibuf;

   m_Player->pCommonClassFactory->CreateInstance(CLSID_IHXBuffer, (void **) &ibuf);
   if (ibuf)
   {
      ibuf->SetSize(len);
      outbuf = ibuf->GetBuffer();

      // equalize
      if (m_Player->ppctrl[m_index]->volume && m_Player->isEQenabled() && m_format.uBitsPerSample == 16)
      {
         equalize(data, outbuf, len);

         // finally adjust the volume
         len = volumeize(outbuf, len);
      }
      else
         // finally adjust the volume
         len = volumeize(data, outbuf, len);

      pAudioOutData->pData = ibuf;
      pAudioOutData->ulAudioTime = pAudioInData->ulAudioTime;
      pAudioOutData->uAudioStreamType = pAudioInData->uAudioStreamType;
   }
#else
   // equalize
   if (m_Player->ppctrl[m_index]->volume && m_Player->isEQenabled() && m_format.uBitsPerSample == 16)
   {
      unsigned char *outbuf;
      IHXBuffer *ibuf;

      m_Player->pCommonClassFactory->CreateInstance(CLSID_IHXBuffer, (void **) &ibuf);
      if (ibuf)
      {
         ibuf->SetSize(len);
         outbuf = ibuf->GetBuffer();
         equalize(data, outbuf, len);
         pAudioOutData->pData = ibuf;
         pAudioOutData->ulAudioTime = pAudioInData->ulAudioTime;
         pAudioOutData->uAudioStreamType = pAudioInData->uAudioStreamType;
         pAudioInData->pData->Release();
      }
   }
#endif

   return 0;
}

STDMETHODIMP HSPPostProcessor::OnInit(HXAudioFormat *pFormat)
{
   m_format = *pFormat;
   m_count = 0;
   m_prevtime = 0;

   // set the filter coefficients, in case we need to use the equalizer
   switch(pFormat->ulSamplesPerSec)
   {
      case 8000:
         //iir_cf = iir_cf10_8000;  <-- doesn't work
         iir_cf = iir_cf10_11k_11025; // works
         break;

      case 11025:
         //iir_cf = iir_cf10_11025; <-- not tested (cant get an encoder to give me this sfreq)
         iir_cf = iir_cf10_11k_11025; // not tested, but works for 8k
         break;

      case 16000:
         //iir_cf = iir_cf10_16000; <-- doesn't work
         iir_cf = iir_cf10_22k_22050; // works
         break;

      case 22050:
         //iir_cf = iir_cf10_22050;
         iir_cf = iir_cf10_22k_22050; // this set actually works...
         break;

      case 32000:
         iir_cf = iir_cf10_32000; // works
         break;

      case 48000:
         iir_cf = iir_cf10_48000;  // not tested
         break;

      case 44100:
      default:
         iir_cf = iir_cf10_44100; // works
         break;
   }

   m_i = 0;
   m_j = 2;
   m_k = 1;

   memset(&data_history, 0, sizeof(data_history));

#ifndef HELIX_SW_VOLUME_INTERFACE
   // setup the gain tool for volume
   if (m_gaintool)
      gainFree(m_gaintool);

   int bps = pFormat->uBitsPerSample / 8;
   m_gaintool = gainInit(pFormat->ulSamplesPerSec, pFormat->uChannels, bps);
   setGain(m_Player->ppctrl[m_index]->volume);
#endif

   return 0;
}


#ifndef HELIX_SW_VOLUME_INTERFACE
void HSPPostProcessor::setGain(int volume)
{
   if (m_gaintool)
   {
      if (volume == 0)
         gainSetMute(m_gaintool);
      else
      {
         //m_gaindB = GAIN_MIN_dB + (GAIN_MAX_dB - GAIN_MIN_dB) * (float) volume / 100.0;
         //m_Player->print2stderr("GAIN set to %f\n", m_gaindB);
         //gainSetImmediatedB(m_gaindB, m_gaintool);

         gainSetImmediate( (float) volume / 100.0, m_gaintool );
      }
   }
}
#endif


void HSPPostProcessor::scopeify(unsigned long time, unsigned char *data, size_t len)
{
   int bytes_per_sample = m_format.uBitsPerSample / 8;

   // TODO: 32 bit samples
   if ( (bytes_per_sample != 1 && bytes_per_sample != 2) )
      return; // no scope

   unsigned long scopebuf_timeinc = (unsigned long)(1000.0 * (double)len / ((double)m_format.ulSamplesPerSec * (double)bytes_per_sample));
   DelayQueue *item = new DelayQueue(len);
   memcpy(item->buf, data, len);
   item->len = len;
   item->time = time;
   item->etime = time + scopebuf_timeinc;

   m_prevtime = item->etime;

   item->nchan = m_format.uChannels;
   item->bps = bytes_per_sample;
   item->spb = len / item->nchan;
   item->spb /= bytes_per_sample;
   item->tps = (double) scopebuf_timeinc / (double) item->spb;
   m_Player->addScopeBuf(item, m_index);
}


void HSPPostProcessor::updateEQgains(int pamp, vector<int> &equalizerGains)
{
   for (int i=0; i<ETQ_CHANNELS; i++)
   {
      preamp[i] = (float) pamp * 0.01;


      for (int j=0; j<ETQ_MAX_BANDS; j++)
         gain[j][i] = (float)(equalizerGains[j]) * 0.012 - 0.2;
   }
}


void HSPPostProcessor::equalize(unsigned char *inbuf, unsigned char *outbuf, size_t length)
{
   int index, band, channel;
   int tempint, halflength;
   float out[ETQ_CHANNELS], pcm[ETQ_CHANNELS];
   short int *data = (short int *) inbuf;
   short int *dataout = (short int *) outbuf;

   /**
    * IIR filter equation is
    * y[n] = 2 * (alpha*(x[n]-x[n-2]) + gamma*y[n-1] - beta*y[n-2])
    *
    * NOTE: The 2 factor was introduced in the coefficients to save
    * 			a multiplication
    *
    * This algorithm cascades two filters to get nice filtering
    * at the expense of extra CPU cycles
    */
   /* 16bit, 2 bytes per sample, so divide by two the length of
    * the buffer (length is in bytes)
    */
   halflength = (length >> 1);
   for (index = 0; index < halflength; index+=m_format.uChannels)
   {
      /* For each channel */
      for (channel = 0; channel < m_format.uChannels; channel++)
      {
         pcm[channel] = (float) data[index+channel];

         /* Preamp gain */
         pcm[channel] *= preamp[channel];

         out[channel] = 0.;
         /* For each band */
         for (band = 0; band < BAND_NUM; band++)
         {
            /* Store Xi(n) */
            data_history[band][channel].x[m_i] = pcm[channel];
            /* Calculate and store Yi(n) */
            data_history[band][channel].y[m_i] =
               (
                  /* = alpha * [x(n)-x(n-2)] */
                  iir_cf[band].alpha * ( data_history[band][channel].x[m_i]
                   -  data_history[band][channel].x[m_k])
                  /* + gamma * y(n-1) */
                  + iir_cf[band].gamma * data_history[band][channel].y[m_j]
                  /* - beta * y(n-2) */
                  - iir_cf[band].beta * data_history[band][channel].y[m_k]
                  );
            /*
             * The multiplication by 2.0 was 'moved' into the coefficients to save
             * CPU cycles here */
            /* Apply the gain  */
            out[channel] +=  data_history[band][channel].y[m_i]*gain[band][channel]; // * 2.0;
         } /* For each band */

         /* Volume stuff
            Scale down original PCM sample and add it to the filters
            output. This substitutes the multiplication by 0.25
            Go back to use the floating point multiplication before the
            conversion to give more dynamic range
         */
         out[channel] += pcm[channel]*0.25;

         /* Round and convert to integer */
         tempint = lrintf(out[channel]);

         /* Limit the output */
         if (tempint < -32768)
            dataout[index+channel] = -32768;
         else if (tempint > 32767)
            dataout[index+channel] = 32767;
         else
            dataout[index+channel] = tempint;
      } /* For each channel */

      m_i++; m_j++; m_k++;

      /* Wrap around the indexes */
      if (m_i == 3) m_i = 0;
      else if (m_j == 3) m_j = 0;
      else m_k = 0;
   }/* For each pair of samples */
}


#ifndef HELIX_SW_VOLUME_INTERFACE
int HSPPostProcessor::volumeize(unsigned char *data, size_t len)
{
   gainFeed(data, data, len, m_gaintool);

   return len;
}


int HSPPostProcessor::volumeize(unsigned char *data, unsigned char *outbuf, size_t len)
{
   gainFeed(data, outbuf, len, m_gaintool);

   return len;
}
#endif

HSPPostMixAudioHook::HSPPostMixAudioHook(HelixSimplePlayer *player, int playerIndex) :
   m_Player(player), m_index(playerIndex), m_lRefCount(0), m_processor(0)
{
   AddRef();

   m_Player->print2stderr("POST MIX HOOK CTOR\n");

   m_processor = new HSPPostProcessor(player, playerIndex);
}


HSPPostMixAudioHook::~HSPPostMixAudioHook()
{
   m_processor->Release();
}

/*
 *  IUnknown methods
 */
STDMETHODIMP
HSPPostMixAudioHook::QueryInterface(REFIID riid, void** ppvObj)
{
   if(IsEqualIID(riid, IID_IUnknown))
   {
      AddRef();
      *ppvObj = (IUnknown*)(IHXAudioHook *)this;
      return HXR_OK;
   }
   else if(IsEqualIID(riid, IID_IHXAudioHook))
   {
      AddRef();
      *ppvObj = (IHXAudioHook *)this;
      return HXR_OK;
   }
   *ppvObj = NULL;
   return HXR_NOINTERFACE;
}


STDMETHODIMP_(UINT32)
HSPPostMixAudioHook::AddRef()
{
    return InterlockedIncrement(&m_lRefCount);
}

STDMETHODIMP_(UINT32)
HSPPostMixAudioHook::Release()
{
    if (InterlockedDecrement(&m_lRefCount) > 0)
    {
        return m_lRefCount;
    }

    m_Player->print2stderr("DELETING POST MIX HOOK index %d\n", m_index);
    delete this;
    return 0;
}

STDMETHODIMP HSPPostMixAudioHook::OnBuffer(HXAudioData *pAudioInData, HXAudioData */*pAudioOutData*/)
{
   unsigned long len;
   unsigned char *data;

   pAudioInData->pData->Get(data, len);

   // feed the visualizations, if this is a local file (otherwise do it in the FinalHook)
   if (m_Player->isLocal(m_index))
      m_processor->scopeify(pAudioInData->ulAudioTime, data, len);

   return 0;
}

STDMETHODIMP HSPPostMixAudioHook::OnInit(HXAudioFormat *pFormat)
{
   m_Player->print2stderr("POST MIX HOOK OnInit AudioFormat: idx %d ch %d, bps %d, sps %ld, mbs %d\n", m_index, pFormat->uChannels,
          pFormat->uBitsPerSample,
          pFormat->ulSamplesPerSec,
          pFormat->uMaxBlockSize);


   return (m_processor->OnInit(pFormat));
}

void HSPPostMixAudioHook::updateEQgains(int preamp, vector<int> &equalizerGains)
{
   m_processor->updateEQgains(preamp, equalizerGains);
}


#ifndef HELIX_SW_VOLUME_INTERFACE

void HSPPostMixAudioHook::setGain(int volume)
{
   m_processor->setGain(volume);
}

#endif


HSPFinalAudioHook::HSPFinalAudioHook(HelixSimplePlayer *player) : m_Player(player), m_lRefCount(0), m_processor(0)
{
   AddRef();
   m_processor = new HSPPostProcessor(player, 0);
}


HSPFinalAudioHook::~HSPFinalAudioHook()
{
   m_processor->Release();
}

/*
 *  IUnknown methods
 */
STDMETHODIMP
HSPFinalAudioHook::QueryInterface(REFIID riid, void** ppvObj)
{
   if(IsEqualIID(riid, IID_IUnknown))
   {
      AddRef();
      *ppvObj = (IUnknown*)(IHXAudioHook *)this;
      return HXR_OK;
   }
   else if(IsEqualIID(riid, IID_IHXAudioHook))
   {
      AddRef();
      *ppvObj = (IHXAudioHook *)this;
      return HXR_OK;
   }
   *ppvObj = NULL;
   return HXR_NOINTERFACE;
}


STDMETHODIMP_(UINT32)
HSPFinalAudioHook::AddRef()
{
    return InterlockedIncrement(&m_lRefCount);
}

STDMETHODIMP_(UINT32)
HSPFinalAudioHook::Release()
{
    if (InterlockedDecrement(&m_lRefCount) > 0)
    {
        return m_lRefCount;
    }

    delete this;
    return 0;
}

STDMETHODIMP HSPFinalAudioHook::OnBuffer(HXAudioData *pAudioInData, HXAudioData *pAudioOutData)
{
   unsigned long len;
   unsigned char *data;

   pAudioInData->pData->Get(data, len);

   // feed the visualizations, if this is a live stream
   bool anyLocal = false;
   int i = 0;
   while (i< m_Player->numPlayers())
   {
      if (m_Player->isPlaying(i))
         m_processor->setIndex(i); // put the buffers on the queue of the player that's playing
      if (anyLocal = m_Player->isLocal(i))
         break;
      i++;
   }
   if (!anyLocal)
      m_processor->scopeify(pAudioInData->ulAudioTime, data, len);

   return (m_processor->OnBuffer(pAudioInData, pAudioOutData));
}

STDMETHODIMP HSPFinalAudioHook::OnInit(HXAudioFormat *pFormat)
{
   m_Player->print2stderr("FINAL HOOK OnInit AudioFormat: ch %d, bps %d, sps %ld, mbs %d\n",  pFormat->uChannels,
          pFormat->uBitsPerSample,
          pFormat->ulSamplesPerSec,
          pFormat->uMaxBlockSize);

   return (m_processor->OnInit(pFormat));
}

void HSPFinalAudioHook::updateEQgains(int preamp, vector<int> &equalizerGains)
{
   m_processor->updateEQgains(preamp, equalizerGains);
}


#ifndef HELIX_SW_VOLUME_INTERFACE

void HSPFinalAudioHook::setGain(int volume)
{
   m_processor->setGain(volume);
}

#endif