/***************************************************************************
 *             __________               __   ___.
 *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
 *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
 *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
 *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
 *                     \/            \/     \/    \/            \/
 * $Id$
 *
 * Copyright (C) 2005 Dave Chapman
 *
 * All files in this archive are subject to the GNU General Public License.
 * See the file COPYING in the source tree root for full license agreement.
 *
 * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
 * KIND, either express or implied.
 *
 ****************************************************************************/

#include "plugin.h"

#include <codecs/libmad/mad.h>

#include "playback.h"
#include "mp3data.h"
#include "lib/codeclib.h"

static struct plugin_api* rb;

struct mad_stream  Stream IDATA_ATTR;
struct mad_frame  Frame IDATA_ATTR;
struct mad_synth  Synth IDATA_ATTR;
mad_timer_t      Timer;
struct dither d0, d1;

/* The following function is used inside libmad - let's hope it's never
   called. 
*/

void abort(void) {
}

/* The "dither" code to convert the 24-bit samples produced by libmad was
   taken from the coolplayer project - coolplayer.sourceforge.net */

struct dither {
	mad_fixed_t error[3];
	mad_fixed_t random;
};

# define SAMPLE_DEPTH	16
# define scale(x, y)	dither((x), (y))

/*
 * NAME:		prng()
 * DESCRIPTION:	32-bit pseudo-random number generator
 */
static __inline
unsigned long prng(unsigned long state)
{
  return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;
}

/*
 * NAME:        dither()
 * DESCRIPTION:	dither and scale sample
 */
static __inline
signed int dither(mad_fixed_t sample, struct dither *dither)
{
  unsigned int scalebits;
  mad_fixed_t output, mask, random;

  enum {
    MIN = -MAD_F_ONE,
    MAX =  MAD_F_ONE - 1
  };

  /* noise shape */
  sample += dither->error[0] - dither->error[1] + dither->error[2];

  dither->error[2] = dither->error[1];
  dither->error[1] = dither->error[0] / 2;

  /* bias */
  output = sample + (1L << (MAD_F_FRACBITS + 1 - SAMPLE_DEPTH - 1));

  scalebits = MAD_F_FRACBITS + 1 - SAMPLE_DEPTH;
  mask = (1L << scalebits) - 1;

  /* dither */
  random  = prng(dither->random);
  output += (random & mask) - (dither->random & mask);

  //dither->random = random;

  /* clip */
  if (output > MAX) {
    output = MAX;

    if (sample > MAX)
      sample = MAX;
  }
  else if (output < MIN) {
    output = MIN;

    if (sample < MIN)
      sample = MIN;
  }

  /* quantize */
  output &= ~mask;

  /* error feedback */
  dither->error[0] = sample - output;

  /* scale */
  return output >> scalebits;
}

static __inline
signed int detect_silence(mad_fixed_t sample)
{
  unsigned int scalebits;
  mad_fixed_t output, mask;

  enum {
    MIN = -MAD_F_ONE,
    MAX =  MAD_F_ONE - 1
  };

  /* bias */
  output = sample + (1L << (MAD_F_FRACBITS + 1 - SAMPLE_DEPTH - 1));

  scalebits = MAD_F_FRACBITS + 1 - SAMPLE_DEPTH;
  mask = (1L << scalebits) - 1;

  /* clip */
  if (output > MAX) {
    output = MAX;

    if (sample > MAX)
      sample = MAX;
  }
  else if (output < MIN) {
    output = MIN;

    if (sample < MIN)
      sample = MIN;
  }

  /* quantize */
  output &= ~mask;

  /* scale */
  output >>= scalebits + 4;
  
  if (output == 0x00 || output == 0xff)
    return 1;
    
  return 0;
}
#define SHRT_MAX 32767

#define INPUT_CHUNK_SIZE        8192
#define OUTPUT_BUFFER_SIZE	65536 /* Must be an integer multiple of 4. */

unsigned char OutputBuffer[OUTPUT_BUFFER_SIZE];
unsigned char *OutputPtr;
unsigned char *GuardPtr=NULL;
const unsigned char *OutputBufferEnd=OutputBuffer+OUTPUT_BUFFER_SIZE;

mad_fixed_t mad_frame_overlap[2][32][18] IDATA_ATTR;
unsigned char mad_main_data[MAD_BUFFER_MDLEN] IDATA_ATTR;
/* TODO: what latency does layer 1 have? */
int mpeg_latency[3] = { 0, 481, 529 };
#ifdef USE_IRAM
extern char iramcopy[];
extern char iramstart[];
extern char iramend[];
#endif

#undef DEBUG_GAPLESS

/* this is the plugin entry point */
enum plugin_status plugin_start(struct plugin_api* api, void* parm)
{
  struct codec_api *ci = (struct codec_api *)parm;
  struct mp3info *info;
  int Status=0;
  size_t size;
  int file_end;
  unsigned short Sample;
  char *InputBuffer;
  unsigned int samplecount;
  unsigned int samplesdone;
  bool first_frame;
#ifdef DEBUG_GAPLESS
  bool first = true;
  int fd;
#endif
  int i;
  int yieldcounter = 0;
  int stop_skip, start_skip;
  
  /* Generic plugin inititialisation */

  TEST_PLUGIN_API(api);
  rb = api;

#ifdef USE_IRAM
  rb->memcpy(iramstart, iramcopy, iramend-iramstart);
#endif

  /* This function sets up the buffers and reads the file into RAM */

  if (codec_init(api, ci)) {
    return PLUGIN_ERROR;
  }

  /* Create a decoder instance */

  ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*2));
  ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*16));
  
  memset(&Stream, 0, sizeof(struct mad_stream));
  memset(&Frame, 0, sizeof(struct mad_frame));
  memset(&Synth, 0, sizeof(struct mad_synth));
  memset(&Timer, 0, sizeof(mad_timer_t));
  
  mad_stream_init(&Stream);
  mad_frame_init(&Frame);
  mad_synth_init(&Synth);
  mad_timer_reset(&Timer);

  /* We do this so libmad doesn't try to call codec_calloc() */
  memset(mad_frame_overlap, 0, sizeof(mad_frame_overlap));
  Frame.overlap = &mad_frame_overlap;
  Stream.main_data = &mad_main_data;
  /* This label might need to be moved above all the init code, but I don't
     think reiniting the codec is necessary for MPEG. It might even be unwanted
     for gapless playback */
  next_track:
  
#ifdef DEBUG_GAPLESS
  if (first)
    fd = rb->open("/first.pcm", O_WRONLY | O_CREAT);
  else
    fd = rb->open("/second.pcm", O_WRONLY | O_CREAT);
  first = false;
#endif
  
  info = ci->mp3data;
  first_frame = false;
  file_end = 0;
  OutputPtr = OutputBuffer;
    
  while (!*ci->taginfo_ready)
    rb->yield();
  
  
  ci->request_buffer(&size, ci->id3->first_frame_offset);
  ci->advance_buffer(size);
 
  if (info->enc_delay >= 0 && info->enc_padding >= 0) {
      stop_skip = info->enc_padding - mpeg_latency[info->layer];
      if (stop_skip < 0) stop_skip = 0;
      start_skip = info->enc_delay + mpeg_latency[info->layer];
  } else {
      stop_skip = 0;
      /* We want to skip this amount anyway */
      start_skip = mpeg_latency[info->layer];
  }
  
  /* NOTE: currently this doesn't work, the below calculated samples_count
     seems to be right, but sometimes libmad just can't supply us with
     all the data we need... */
  if (info->frame_count) {
      /* TODO: 1152 is the frame size in samples for MPEG1 layer 2 and layer 3,
         it's probably not correct at all for MPEG2 and layer 1 */
      samplecount = info->frame_count*1152 - (start_skip + stop_skip);
      samplesdone = ci->id3->elapsed * (ci->id3->frequency / 100) / 10;
  } else {
      samplecount = ci->id3->length * (ci->id3->frequency / 100) / 10;
      samplesdone = ci->id3->elapsed * (ci->id3->frequency / 100) / 10;
  }
  /* rb->snprintf(buf2, sizeof(buf2), "sc: %d", samplecount);
  rb->splash(0, true, buf2);
  rb->snprintf(buf2, sizeof(buf2), "length: %d", ci->id3->length);
  rb->splash(HZ*5, true, buf2);
  rb->snprintf(buf2, sizeof(buf2), "frequency: %d", ci->id3->frequency);
  rb->splash(HZ*5, true, buf2); */
  
  /* This is the decoding loop. */
  while (1) {
    rb->yield();
    if (ci->stop_codec || ci->reload_codec) { 
        break ;
    }
    
    if (ci->seek_time) {
        unsigned int sample_loc;
        int newpos;
        
        sample_loc = ci->seek_time/1000 * ci->id3->frequency;
        newpos = ci->mp3_get_filepos(ci->seek_time-1);
        if (ci->seek_buffer(newpos)) {
            if (sample_loc >= samplecount + samplesdone)
                break ;
            samplecount += samplesdone - sample_loc;
            samplesdone = sample_loc;
        }
        ci->seek_time = 0;
    }

    /* Lock buffers */
    if (Stream.error == 0) {
        InputBuffer = ci->request_buffer(&size, INPUT_CHUNK_SIZE);
        if (size == 0 || InputBuffer == NULL)
            break ;
        mad_stream_buffer(&Stream, InputBuffer, size);
    }
    
    //if ((int)ci->curpos >= ci->id3->first_frame_offset)
        //first_frame = true;
        
    if(mad_frame_decode(&Frame,&Stream))
    {
      if (Stream.error == MAD_FLAG_INCOMPLETE || Stream.error == MAD_ERROR_BUFLEN) {
          // rb->splash(HZ*1, true, "Incomplete");
        /* This makes the codec to support partially corrupted files too. */
        if (file_end == 30)
            break ;
        
        /* Fill the buffer */
        Stream.error = 0;
        file_end++;
        continue ;
      }
      else if(MAD_RECOVERABLE(Stream.error))
      {
        if(Stream.error!=MAD_ERROR_LOSTSYNC || Stream.this_frame!=GuardPtr)
        {
          // rb->splash(HZ*1, true, "Recoverable...!");
        }
        continue;
      }
      else if(Stream.error==MAD_ERROR_BUFLEN) {
          //rb->splash(HZ*1, true, "Buflen error");
          break ;
      } else {
          //rb->splash(HZ*1, true, "Unrecoverable error");
          Status=1;
          break;
      }
    }
    if (Stream.next_frame)
        ci->advance_buffer_loc((void *)Stream.next_frame);
    file_end = false;
    /* ?? Do we need the timer module? */
    // mad_timer_add(&Timer,Frame.header.duration);

/* DAVE: This can be used to attenuate the audio */
//    if(DoFilter)
//      ApplyFilter(&Frame);   

    mad_synth_frame(&Synth,&Frame);
    
    //if (!first_frame) {
        //samplecount -= Synth.pcm.length;
        //continue ;
    //}
    
    /* Convert MAD's numbers to an array of 16-bit LE signed integers */
    /* We skip start_skip number of samples here, this should only happen for
       very first frame in the stream. */
    /* TODO: possible for start_skip to exceed one frames worth of samples? */
    for (i = start_skip;i<Synth.pcm.length;i++)
    {
      start_skip = 0; /* not very elegant, and might want to keep this value */
      samplesdone++;
      //if (ci->mp3data->padding > 0) {
      //  ci->mp3data->padding--;
      //  continue ;
      //}
      /*if (!first_frame) {
        if (detect_silence(Synth.pcm.samples[0][i]))
            continue ;
        first_frame = true;
      }*/
      
      /* Left channel */
      Sample=scale(Synth.pcm.samples[0][i],&d0);
      *(OutputPtr++)=Sample>>8;
      *(OutputPtr++)=Sample&0xff;

      /* Right channel. If the decoded stream is monophonic then
       * the right output channel is the same as the left one.
       */
      if(MAD_NCHANNELS(&Frame.header)==2)
        Sample=scale(Synth.pcm.samples[1][i],&d1);
      *(OutputPtr++)=Sample>>8;
      *(OutputPtr++)=Sample&0xff;
      
      samplecount--;
      if (samplecount == 0) {
#ifdef DEBUG_GAPLESS
        rb->write(fd, OutputBuffer, (int)OutputPtr-(int)OutputBuffer);
#endif
        while (!ci->audiobuffer_insert(OutputBuffer, (int)OutputPtr-(int)OutputBuffer))
            rb->yield();
        goto song_end;
      }
  
      if (yieldcounter++ == 200) {
        rb->yield();
        yieldcounter = 0;
      }
      
      /* Flush the buffer if it is full. */
      if(OutputPtr==OutputBufferEnd)
      {
#ifdef DEBUG_GAPLESS
        rb->write(fd, OutputBuffer, OUTPUT_BUFFER_SIZE);
#endif
        while (!ci->audiobuffer_insert(OutputBuffer, OUTPUT_BUFFER_SIZE))
            rb->yield();
        OutputPtr=OutputBuffer;
      }
    }
    ci->set_elapsed(samplesdone / (ci->id3->frequency/1000));
  }
  
  song_end:
#ifdef DEBUG_GAPLESS
  rb->close(fd);
#endif
  Stream.error = 0;
  
  if (ci->request_next_track())
    goto next_track;
  return PLUGIN_OK;
}