decode_a52.c

基于linux的DVD播放器程序

/* Ogle - A video player
 * Copyright (C) 2001, 2002 Bj鰎n Englund, H錵an Hjort
 *
 * 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, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 */

#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <inttypes.h>

#include <a52dec/a52.h>
#include <a52dec/mm_accel.h>

#include <libogleao/ogle_ao.h> // Remove me

#include "parse_config.h"
#include "decode.h"
#include "decode_private.h"

#include "audio_config.h"
#include "conversion.h"
#include "audio_play.h"
#include "audio_types.h"
#include "debug_print.h"

typedef struct {
  adec_handle_t handle;
  uint64_t PTS;
  int pts_valid;
  int scr_nr;
  uint8_t *coded_buf;
  uint8_t *buf_ptr;
  int sample_rate;
  int bytes_needed;
  int availflags;
  int output_flags;
  int decoding_flags;
  /* A/52 */
  a52_state_t *state;
  sample_t *samples;
  int disable_dynrng;
  sample_t level;
  int adjust_level;
  SampleFormat_t output_format;
  int stereo_mode;
} adec_a52_handle_t;


static
int a52flags_to_channels(int flags)
{
  ChannelType_t chtypemask = 0;
  
  switch(flags & A52_CHANNEL_MASK) {
  case A52_CHANNEL:
    //TODO ???
    chtypemask = ChannelType_Left | ChannelType_Right;
    break;
  case A52_MONO:
    chtypemask = ChannelType_Mono;
    break;
  case A52_STEREO:
  case A52_DOLBY:
    chtypemask = ChannelType_Left | ChannelType_Right;
    break;
  case A52_3F:
    chtypemask = ChannelType_Left | ChannelType_Center | ChannelType_Right;
    break;
  case A52_2F1R:
    chtypemask = ChannelType_Left | ChannelType_Right | ChannelType_Surround;
    break;
  case A52_3F1R:
    chtypemask = ChannelType_Left | ChannelType_Center | ChannelType_Right | 
      ChannelType_Surround;
    break;
  case A52_2F2R:
    chtypemask = ChannelType_Left | ChannelType_Right | 
      ChannelType_LeftSurround | ChannelType_RightSurround;
    break;
  case A52_3F2R:
    chtypemask = ChannelType_Left | ChannelType_Center | ChannelType_Right |
      ChannelType_LeftSurround | ChannelType_RightSurround;
    break;
    
  }
  
  if(flags & A52_LFE) {
    chtypemask |= ChannelType_LFE;
  }
  
  return chtypemask;
}

static
int config_to_a52flags(audio_config_t *conf, int stereo_mode)
{
  int i;
  ChannelType_t chtypemask = 0;
  int hasLFE = 0;

  for(i = 0; i < conf->dst_format.nr_channels; i++) {
    if(conf->dst_format.ch_array[i] == ChannelType_LFE)
      hasLFE = A52_LFE;
    else
      chtypemask |= conf->dst_format.ch_array[i];
  }
  switch(chtypemask) {
  case ChannelType_Mono:
    return A52_MONO | hasLFE;
  case ChannelType_Left | ChannelType_Right:
    switch(stereo_mode) {
    case 0:                    // left right
      return A52_3F2R | hasLFE;    // we say 3ch so no mixing will be done
    case 1:                    // front channels mixed to stereo
      return A52_STEREO | hasLFE; 
    case 2:                    // all channels mixed to dolby stereo
      return A52_DOLBY | hasLFE; 
    }
  case ChannelType_Left | ChannelType_Center | ChannelType_Right:
    return A52_3F | hasLFE;
  case ChannelType_Left | ChannelType_Right | ChannelType_Surround:
    return A52_2F1R | hasLFE;
  case ChannelType_Left | ChannelType_Center | ChannelType_Right | ChannelType_Surround:
    return A52_3F1R | hasLFE;
  case ChannelType_Left | ChannelType_Right |  ChannelType_LeftSurround | ChannelType_RightSurround:
    return A52_2F2R | hasLFE;
  case ChannelType_Left | ChannelType_Center | ChannelType_Right | ChannelType_LeftSurround | ChannelType_RightSurround:
    return A52_3F2R | hasLFE;
  case ChannelType_AC3:
  default:
    break;
  }
  return A52_3F2R | hasLFE; // Some strange sound configuration...
}


static 
void a52flags_to_format(int flags, int *channels, ChannelType_t *channel[])
{
  int f = flags & A52_CHANNEL_MASK;
  int nr_channels = 0;
  ChannelType_t *ch_array;

  ch_array = malloc(6 * sizeof(ChannelType_t));

  // the order of the channels is
  // lfe, left, center, right, leftsurround, rightsurround

  if(flags & A52_LFE) { // sub
    ch_array[nr_channels++] = ChannelType_LFE;
  }

  if(f == A52_CHANNEL) { 
    ch_array[nr_channels++] = ChannelType_Left; //??
    ch_array[nr_channels++] = ChannelType_Right; //??
  } if(f == A52_MONO || f == A52_CHANNEL1 || f == A52_CHANNEL2) {
    ch_array[nr_channels++] = ChannelType_Mono;
  } else {
    if(1) { // left
      ch_array[nr_channels++] = ChannelType_Left;
    }
    if(f == A52_3F || f == A52_3F1R || f == A52_3F2R) { // center
      ch_array[nr_channels++] = ChannelType_Center;
    }
    if(1) { // right
      ch_array[nr_channels++] = ChannelType_Right;
    }
    if(f == A52_2F1R || f == A52_3F1R) { // mono surround
      ch_array[nr_channels++] = ChannelType_Surround;
    }
    if(f == A52_2F2R || f == A52_3F2R) { // left and right surround
      ch_array[nr_channels++] = ChannelType_LeftSurround;
      ch_array[nr_channels++] = ChannelType_RightSurround;
    }
  }

  *channels = nr_channels;
  *channel = ch_array;
}



static 
int decode_a52(adec_a52_handle_t *handle, uint8_t *start, int len,
	       int pts_offset, uint64_t new_PTS, int scr_nr)
{
  static uint8_t *indata_ptr;
  int bytes_left;
  int frame_len;
  int new_sample_rate;
  int bit_rate;
  int n;
  int bytes_to_get;

  indata_ptr = start;
  bytes_left = len;
  

  while(bytes_left > 0) {
    if(handle->bytes_needed > bytes_left) {
      bytes_to_get = bytes_left;
      memcpy(handle->buf_ptr, indata_ptr, bytes_to_get);
      bytes_left -= bytes_to_get;
      handle->buf_ptr += bytes_to_get;
      indata_ptr += bytes_to_get;
      handle->bytes_needed -= bytes_to_get;
      
      //fprintf(stderr, "bytes_needed: %d > bytes_left: %d\n",
      //      handle->bytes_needed, bytes_left);
      return handle->bytes_needed;
    } else {
      bytes_to_get = handle->bytes_needed;
      memcpy(handle->buf_ptr, indata_ptr, bytes_to_get);
      bytes_left -= bytes_to_get;
      handle->buf_ptr += bytes_to_get;
      indata_ptr += bytes_to_get;
      handle->bytes_needed -= bytes_to_get;
    }
    
    // when we come here we have all data we want
    // either the 7 header bytes or the whole frame
    
    if(handle->buf_ptr - handle->coded_buf == 7) {
      // 7 bytes header data to test
      int new_availflags;
      static int has_sync = 0;
      frame_len = a52_syncinfo(handle->coded_buf,
			       &new_availflags, &new_sample_rate, &bit_rate);
      if(frame_len == 0) {
	//this is not the start of a frame
	if(has_sync) {
	  DNOTE("%s", "decode_a52: lost sync\n");
	  has_sync = 0;
	}
	for(n = 0; n < 6; n++) {
	  handle->coded_buf[n] = handle->coded_buf[n+1];
	}
	handle->buf_ptr--;
	handle->bytes_needed = 1;
      } else {
	if(!has_sync) {
	  DNOTE("%s", "decode_a52: found sync\n");
	  has_sync = 1;
	}

	// we have the start of a frame
	if((int)(indata_ptr-7-start) == pts_offset) {
	  // this frame has a pts
	  //fprintf(stderr, "frame with pts\n");
	  handle->PTS = new_PTS;
	  handle->pts_valid = 1;
	  handle->scr_nr = scr_nr;
	} else {
	  //fprintf(stderr, "frame with no pts\n");
	}

	handle->bytes_needed = frame_len - 7;
	if((new_availflags != handle->availflags) || 
	   (new_sample_rate != handle->sample_rate)) {
	  ChannelType_t chtypemask;
	  SampleFormat_t format;
	  //fprintf(stderr, "new flags\n");

	  handle->availflags = new_availflags;
	  handle->sample_rate = new_sample_rate;
	  
	  //change a52 flags to generic channel flags
	  
	  chtypemask = a52flags_to_channels(handle->availflags);
	  chtypemask |= ChannelType_AC3;
	  
	  audio_config(handle->handle.config, chtypemask,
		       handle->sample_rate, 16);
	  format = handle->handle.config->dst_format.sample_format;
	  switch(format) {
	  case SampleFormat_IEC61937:
	    handle->output_format = SampleFormat_AC3Frame;
	    break;
	  case SampleFormat_Signed:
	    handle->output_format = SampleFormat_A52float;
	    break;
	  default:
	    WARNING("unknown sample format %d\n", format); 
	    handle->output_format = SampleFormat_A52float;
	    break;
	  }
	  //change config into a52dec flags
	  handle->output_flags = config_to_a52flags(handle->handle.config,
						    handle->stereo_mode);
	}
	
      }
    } else if(handle->output_format == SampleFormat_A52float) {
      int i;
      int flags;
      sample_t level; // Hack for the float_to_int function
      int bias = 384;
      //we have a whole frame to decode
      //fprintf(stderr, "decode frame\n");
      //decode
      
      switch(handle->availflags & A52_CHANNEL_MASK) {
      case A52_3F:
      case A52_2F1R:
      case A52_3F1R:
      case A52_2F2R:
      case A52_3F2R:
	level = handle->level;
	break;
      default:
	level = 1.0;
	break;
      }
      
      flags = handle->output_flags;

      if(handle->adjust_level) {
	flags |= A52_ADJUST_LEVEL;
      }

      if(a52_frame(handle->state, handle->coded_buf, &flags, &level, bias)) {
	DNOTE("%s", "a52_frame() error\n");
	goto error;
      }

      if(handle->decoding_flags != flags) {
	//new set of channels decoded
	//change into config format
	audio_format_t new_format;

	handle->decoding_flags = flags;
	
	a52flags_to_format(flags, &new_format.nr_channels,
			   &new_format.ch_array);
	DNOTE("%d channels decoded: ", new_format.nr_channels);
	new_format.sample_rate = handle->sample_rate;
	new_format.sample_resolution = 16;
	new_format.sample_format = SampleFormat_A52float;
	init_sample_conversion((adec_handle_t *)handle, &new_format, 256*6);
	{
	  int n;
	  for(n = 0; n < new_format.nr_channels; n++) {
	    DNOTEC(" %s", channeltype_str(new_format.ch_array[n]));
	  }
	  DNOTEC("%s", "\n");
	}
	free(new_format.ch_array);
      }

      convert_samples_start((adec_handle_t *)handle);

	
      if(handle->disable_dynrng) {
	a52_dynrng(handle->state, NULL, NULL);
      }
      for(i = 0; i < 6; i++) {

	if(a52_block(handle->state)) {
	  DNOTE("%s", "a52_block() error\n");
	  goto error;
	}
	convert_samples((adec_handle_t *)handle, handle->samples, 256);
      }

#if 0
      {
	int n;
	static short max_s = 0;
	static int t = 0;
	for(n = 0; n < 1536; n++) {
	  if(((short *)(handle->handle.output_buf))[n] > max_s) {
	    max_s = ((short *)(handle->handle.output_buf))[n];
	  }
	}
	t++;
	if(t == 10) {
	  t = 0;
	  fprintf(stderr, "max: %d\n", max_s);
	  max_s = 0;
	}
      }
#endif
      //output, look over how the pts/scr is handle for sync here 
      play_samples((adec_handle_t *)handle, handle->scr_nr, 
		   handle->PTS, handle->pts_valid);
      handle->pts_valid = 0;

    error:

      //make space for next frame
      handle->buf_ptr = handle->coded_buf;
      handle->bytes_needed = 7;

    } else if(handle->output_format == SampleFormat_AC3Frame) {
      int flags;
      sample_t level; // Hack for the float_to_int function
      int bias = 384;
      //we have a whole frame to decode
      //fprintf(stderr, "decode frame\n");
      //decode
      
      
      flags = handle->output_flags;


      if(a52_frame(handle->state, handle->coded_buf, &flags, &level, bias)) {
	DNOTE("%s", "a52_frame() error\n");
	goto error2;
      }

      if(handle->decoding_flags != flags) {
	//new set of channels decoded
	//change into config format
	audio_format_t new_format;

	handle->decoding_flags = flags;
	
	new_format.sample_rate = handle->sample_rate;
	new_format.sample_resolution = 16;
	new_format.sample_format = SampleFormat_AC3Frame;
	init_sample_conversion((adec_handle_t *)handle, &new_format, 256*6);
      }

      convert_samples_start((adec_handle_t *)handle);


      convert_samples((adec_handle_t *)handle, handle->coded_buf, 
		      0 /*_framesize*/);

  
      play_samples((adec_handle_t *)handle, handle->scr_nr, 
		   handle->PTS, handle->pts_valid);
      handle->pts_valid = 0;

    error2:

      //make space for next frame
      handle->buf_ptr = handle->coded_buf;
      handle->bytes_needed = 7;
    }
    
  } 
  //fprintf(stderr, "bytes_needed: %d\n", handle->bytes_needed);
  return handle->bytes_needed;
}


static
int flush_a52(adec_a52_handle_t *handle)
{
  handle->pts_valid = 0;
  handle->buf_ptr = handle->coded_buf;
  handle->bytes_needed = 7;
  
  // Fix this.. not the right way to do things I belive.
  if(handle->handle.config && handle->handle.config->adev_handle)
    ogle_ao_flush(handle->handle.config->adev_handle);
  return 0;
}


static
void free_a52(adec_a52_handle_t *handle)
{
  free(handle->coded_buf);
  a52_free(handle->state);
  audio_config_close(handle->handle.config);
  free(handle);
  return;
}


adec_handle_t *init_a52(void)
{
  adec_a52_handle_t *handle;
  
  if(!(handle = (adec_a52_handle_t *)malloc(sizeof(adec_a52_handle_t)))) {
    return NULL;
  }
  
  memset(&handle->handle, 0, sizeof(struct adec_handle_s));
  // not set: drain  
  handle->handle.decode = (audio_decode_t) decode_a52;  // function pointers
  handle->handle.flush  = (audio_flush_t)  flush_a52;
  handle->handle.free   = (audio_free_t)   free_a52;
  handle->handle.output_buf = NULL;
  handle->handle.output_buf_size = 0;
  handle->handle.output_buf_ptr = handle->handle.output_buf;

  handle->PTS = 0;
  handle->pts_valid = 0;
  handle->scr_nr = 0;
  handle->coded_buf = (uint8_t *)malloc(3840); // max size of a52 frame
  handle->buf_ptr = (uint8_t *)handle->coded_buf;
  handle->bytes_needed = 7;
  handle->sample_rate = 0;
  handle->decoding_flags = 0;
  //  handle->decoded_format = NULL;

  {
    uint32_t accel;
    accel = MM_ACCEL_DJBFFT;

    handle->state = a52_init(accel);
    if(handle->state == NULL) {
        FATAL("%s", "A/52 init failed\n");
        exit(1);
    }
    handle->samples = a52_samples(handle->state);
    if(handle->samples == NULL) {
        FATAL("%s", "A/52 samples failed\n");
        exit(1);
    }
  }
  handle->disable_dynrng = !get_a52_drc();
  handle->level = (sample_t)get_a52_level();
  handle->adjust_level = 1;
  handle->stereo_mode = get_a52_stereo_mode();
  return (adec_handle_t *)handle;
}