/*****************************************************************
 * gmerlin-avdecoder - a general purpose multimedia decoding library
 *
 * Copyright (c) 2001 - 2012 Members of the Gmerlin project
 * gmerlin-general@lists.sourceforge.net
 * http://gmerlin.sourceforge.net
 *
 * This program is free software: you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation, either version 2 of the License, or
 * (at your option) any later version.
 *
 * This program 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 General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with this program.  If not, see <http://www.gnu.org/licenses/>.
 * *****************************************************************/

#include <avdec_private.h>
#include <stdlib.h>
#include <stdio.h>

#define BLOCKS_PER_PACKET 1024
#define LOG_DOMAIN "au"

/* AU demuxer */

typedef struct
  {
  uint32_t magic;       /* magic number */
  uint32_t hdr_size;    /* size of this header */ 
  uint32_t data_size;   /* length of data (optional) */ 
  uint32_t encoding;    /* data encoding format */
  uint32_t sample_rate; /* samples per second */
  uint32_t channels;    /* number of interleaved channels */
  } Audio_filehdr;

typedef struct
  {
  int samples_per_block;
  int bytes_per_second;
  } au_priv_t;

/* Define the magic number */ 
#define AUDIO_FILE_MAGIC ((uint32_t)0x2e736e64)

/* Define the encoding fields */ 
#define AUDIO_FILE_ENCODING_MULAW_8      (1)  /* 8-bit ISDN u-law */
#define AUDIO_FILE_ENCODING_LINEAR_8     (2)  /* 8-bit linear PCM */
#define AUDIO_FILE_ENCODING_LINEAR_16    (3)  /* 16-bit linear PCM */
#define AUDIO_FILE_ENCODING_LINEAR_24    (4)  /* 24-bit linear PCM */ 
#define AUDIO_FILE_ENCODING_LINEAR_32    (5)  /* 32-bit linear PCM */ 
#define AUDIO_FILE_ENCODING_FLOAT        (6)  /* 32-bit IEEE floating point */ 
#define AUDIO_FILE_ENCODING_DOUBLE       (7)  /* 64-bit IEEE floating point */ 
#define AUDIO_FILE_ENCODING_ADPCM_G721   (23) /* 4-bit CCITT g.721 ADPCM */ 
#define AUDIO_FILE_ENCODING_ADPCM_G722   (24) /* CCITT g.722 ADPCM */ 
#define AUDIO_FILE_ENCODING_ADPCM_G723_3 (25) /* CCITT g.723 3-bit ADPCM */ 
#define AUDIO_FILE_ENCODING_ADPCM_G723_5 (26) /* CCITT g.723 5-bit ADPCM */ 
#define AUDIO_FILE_ENCODING_ALAW_8       (27) /* 8-bit ISDN A-law */

static int probe_au(bgav_input_context_t * input)
  {
  uint8_t test_data[4];
  if(bgav_input_get_data(input, test_data, 4) < 4)
    return 0;
  if((test_data[0] == '.') &&
     (test_data[1] == 's') &&
     (test_data[2] == 'n') &&
     (test_data[3] == 'd'))
    return 1;
  return 0;
  }

static int64_t pos_2_time(bgav_demuxer_context_t * ctx, int64_t pos)
  {
  au_priv_t * priv;
  bgav_stream_t * s = bgav_track_get_audio_stream(ctx->tt->cur, 0);
  
  priv = ctx->priv;
  return ((pos - ctx->data_start) * priv->samples_per_block) / (s->data.audio.block_align);
  }

static int64_t time_2_pos(bgav_demuxer_context_t * ctx, int64_t time)
  {
  au_priv_t * priv;
  bgav_stream_t * s = bgav_track_get_audio_stream(ctx->tt->cur, 0);

  priv = ctx->priv;
  return ctx->data_start +
    (time *
     s->data.audio.block_align)/(priv->samples_per_block);
  
  }


static int open_au(bgav_demuxer_context_t * ctx)
  {
  Audio_filehdr hdr;
  au_priv_t * priv;
  bgav_stream_t * as;
  int samples_per_block = 0;

  /* Create track */
  ctx->tt = bgav_track_table_create(1);
  
  if(!bgav_input_read_fourcc(ctx->input, &hdr.magic) ||
     !bgav_input_read_32_be(ctx->input, &hdr.hdr_size) ||
     !bgav_input_read_32_be(ctx->input, &hdr.data_size) ||
     !bgav_input_read_32_be(ctx->input, &hdr.encoding) ||
     !bgav_input_read_32_be(ctx->input, &hdr.sample_rate) ||
     !bgav_input_read_32_be(ctx->input, &hdr.channels))
    return 0;

  /* Get codec */
  
  switch(hdr.encoding)
    {
    case AUDIO_FILE_ENCODING_MULAW_8:      /* 8-bit ISDN u-law */
      as = bgav_track_add_audio_stream(ctx->tt->cur, ctx->opt);
      samples_per_block = 1;
      as->fourcc = BGAV_MK_FOURCC('u', 'l', 'a', 'w');
      as->data.audio.block_align = hdr.channels;
      break;
    case AUDIO_FILE_ENCODING_ALAW_8:       /* 8-bit ISDN A-law */
      as = bgav_track_add_audio_stream(ctx->tt->cur, ctx->opt);
      samples_per_block = 1;
      as->fourcc = BGAV_MK_FOURCC('a', 'l', 'a', 'w');
      as->data.audio.block_align = hdr.channels;
      
      break;
    case AUDIO_FILE_ENCODING_LINEAR_8:     /* 8-bit linear PCM */
      as = bgav_track_add_audio_stream(ctx->tt->cur, ctx->opt);
      samples_per_block = 1;
      as->fourcc = BGAV_MK_FOURCC('t', 'w', 'o', 's');
      as->data.audio.block_align = hdr.channels;
      as->data.audio.bits_per_sample = 8;

      break;
    case AUDIO_FILE_ENCODING_LINEAR_16:    /* 16-bit linear PCM */
      as = bgav_track_add_audio_stream(ctx->tt->cur, ctx->opt);
      samples_per_block = 1;
      as->fourcc = BGAV_MK_FOURCC('t', 'w', 'o', 's');
      as->data.audio.block_align = hdr.channels * 2;
      as->data.audio.bits_per_sample = 16;
      
      break;
#if 0
    case AUDIO_FILE_ENCODING_LINEAR_24:    /* 24-bit linear PCM */ 
      break;
    case AUDIO_FILE_ENCODING_LINEAR_32:    /* 32-bit linear PCM */ 
      break;
    case AUDIO_FILE_ENCODING_FLOAT:        /* 32-bit IEEE floating point */ 
      break;
    case AUDIO_FILE_ENCODING_DOUBLE:       /* 64-bit IEEE floating point */ 
      break;
    case AUDIO_FILE_ENCODING_ADPCM_G721:   /* 4-bit CCITT g.721 ADPCM */ 
      break;
    case AUDIO_FILE_ENCODING_ADPCM_G722:   /* CCITT g.722 ADPCM */ 
      break;
    case AUDIO_FILE_ENCODING_ADPCM_G723_3: /* CCITT g.723 3-bit ADPCM */ 
      break;
    case AUDIO_FILE_ENCODING_ADPCM_G723_5: /* CCITT g.723 5-bit ADPCM */ 
      break;
#endif
    default:
      gavl_log(GAVL_LOG_ERROR, LOG_DOMAIN,
               "Unsupported encoding %d", hdr.encoding);
      return 0;
    }

  /* Get audio format */
  
  as->data.audio.format->samplerate = hdr.sample_rate;
  as->data.audio.format->num_channels = hdr.channels;
  
  priv = calloc(1, sizeof(*priv));
  ctx->priv = priv;
  
  if(ctx->input->flags & BGAV_INPUT_CAN_SEEK_BYTE)
    ctx->flags |= BGAV_DEMUXER_CAN_SEEK;
  
  /* Skip everything until data section */

  if(hdr.hdr_size > 24)
    bgav_input_skip(ctx->input, hdr.hdr_size - 24);

  /* Get data start and duration */

  ctx->data_start = ctx->input->position;

  if(ctx->input->total_bytes > 0)
    as->stats.total_bytes = ctx->input->total_bytes - ctx->data_start;
  
  priv->samples_per_block = samples_per_block;

  if(as->stats.total_bytes)
    {
    as->stats.pts_end = pos_2_time(ctx, ctx->data_start + as->stats.total_bytes);
    }
  
  ctx->flags |= BGAV_DEMUXER_HAS_DATA_START;
  
  bgav_track_set_format(ctx->tt->cur, "AU/SND", "audio/x-au");
  
  ctx->index_mode = INDEX_MODE_PCM;
  return 1;
  }

static int next_packet_au(bgav_demuxer_context_t * ctx)
  {
  bgav_packet_t * p;
  bgav_stream_t * s;
  int bytes_read;
  s = bgav_track_get_audio_stream(ctx->tt->cur, 0);
  p = bgav_stream_get_packet_write(s);
  
  bgav_packet_alloc(p, s->data.audio.block_align * BLOCKS_PER_PACKET);

  p->pts = pos_2_time(ctx, ctx->input->position);
  PACKET_SET_KEYFRAME(p);
  bytes_read = bgav_input_read_data(ctx->input, p->data,
                                    s->data.audio.block_align * BLOCKS_PER_PACKET);
  p->data_size = bytes_read;
  bgav_stream_done_packet_write(s, p);
  if(!bytes_read)
    return 0;
  return 1;
  }

static void seek_au(bgav_demuxer_context_t * ctx, gavl_time_t time, int scale)
  {
  bgav_stream_t * s;
  int64_t position;
  s = bgav_track_get_audio_stream(ctx->tt->cur, 0);
  
  position = time_2_pos(ctx, gavl_time_rescale(scale, s->timescale, time));
  position -= ctx->data_start;
  position /= s->data.audio.block_align;
  position *= s->data.audio.block_align;
  position += ctx->data_start;
  bgav_input_seek(ctx->input, position, SEEK_SET);
  STREAM_SET_SYNC(s, pos_2_time(ctx, position));
  }

static void close_au(bgav_demuxer_context_t * ctx)
  {
  au_priv_t * priv;
  priv = ctx->priv;
  free(priv);
  }

const bgav_demuxer_t bgav_demuxer_au =
  {
    .probe =       probe_au,
    .open =        open_au,
    .next_packet = next_packet_au,
    .seek =        seek_au,
    .close =       close_au
  };