1.1 --- a/src/aica/aica.c	Tue Jan 10 14:00:00 2006 +0000
     1.2 +++ b/src/aica/aica.c	Thu Jan 12 11:30:19 2006 +0000
     1.3 @@ -1,5 +1,5 @@
     1.4  /**
     1.5 - * $Id: aica.c,v 1.11 2006-01-10 13:56:54 nkeynes Exp $
     1.6 + * $Id: aica.c,v 1.12 2006-01-12 11:30:19 nkeynes Exp $
     1.7   * 
     1.8   * This is the core sound system (ie the bit which does the actual work)
     1.9   *
    1.10 @@ -39,8 +39,6 @@
    1.11  int aica_load_state( FILE *f );
    1.12  uint32_t aica_run_slice( uint32_t );
    1.13  
    1.14 -#define IS_TIMER_ENABLED() (MMIO_READ( AICA2, AICA_TCR ) & 0x40)
    1.15 -
    1.16  struct dreamcast_module aica_module = { "AICA", aica_init, aica_reset, 
    1.17  					aica_start, aica_run_slice, aica_stop,
    1.18  					aica_save_state, aica_load_state };
    1.19 @@ -82,23 +80,16 @@
    1.20      int reset = MMIO_READ( AICA2, AICA_RESET );
    1.21      if( (reset & 1) == 0 ) { /* Running */
    1.22  	int num_samples = (nanosecs_done + nanosecs) / AICA_SAMPLE_RATE - samples_done;
    1.23 -	int i;
    1.24 -	for( i=0; i<num_samples; i++ ) {
    1.25 -	    nanosecs = arm_run_slice( AICA_SAMPLE_PERIOD );
    1.26 -	    audio_mix_sample();
    1.27 -	    if( IS_TIMER_ENABLED() ) {
    1.28 -		uint8_t val = MMIO_READ( AICA2, AICA_TIMER );
    1.29 -		val++;
    1.30 -		if( val == 0 )
    1.31 -		    aica_event( AICA_EVENT_TIMER );
    1.32 -		MMIO_WRITE( AICA2, AICA_TIMER, val );
    1.33 -	    }
    1.34 -	    if( !dreamcast_is_running() )
    1.35 -		break;
    1.36 -	}
    1.37 +	num_samples = arm_run_slice( num_samples );
    1.38 +	audio_mix_samples( num_samples );
    1.39 +
    1.40  	samples_done += num_samples;
    1.41  	nanosecs_done += nanosecs;
    1.42      }
    1.43 +    if( nanosecs_done > 1000000000 ) {
    1.44 +	samples_done -= AICA_SAMPLE_RATE;
    1.45 +	nanosecs_done -= 1000000000;
    1.46 +    }
    1.47      return nanosecs;
    1.48  }
    1.49  
    1.50 @@ -256,9 +247,9 @@
    1.51      case 0x00: /* Config + high address bits*/
    1.52  	channel->start = (channel->start & 0xFFFF) | ((val&0x1F) << 16);
    1.53  	if( val & 0x200 ) 
    1.54 -	    channel->loop_count = -1;
    1.55 +	    channel->loop = TRUE;
    1.56  	else 
    1.57 -	    channel->loop_count = 0;
    1.58 +	    channel->loop = FALSE;
    1.59  	switch( (val >> 7) & 0x03 ) {
    1.60  	case 0:
    1.61  	    channel->sample_format = AUDIO_FMT_16BIT;
    1.62 @@ -289,8 +280,8 @@
    1.63      case 0x08: /* Loop start */
    1.64  	channel->loop_start = val;
    1.65  	break;
    1.66 -    case 0x0C: /* Loop end */
    1.67 -	channel->loop_end = channel->end = val;
    1.68 +    case 0x0C: /* End */
    1.69 +	channel->end = val;
    1.70  	break;
    1.71      case 0x10: /* Envelope register 1 */
    1.72  	break;

     2.1 --- a/src/aica/armcore.c	Tue Jan 10 14:00:00 2006 +0000
     2.2 +++ b/src/aica/armcore.c	Thu Jan 12 11:30:19 2006 +0000
     2.3 @@ -1,5 +1,5 @@
     2.4  /**
     2.5 - * $Id: armcore.c,v 1.16 2006-01-10 13:56:54 nkeynes Exp $
     2.6 + * $Id: armcore.c,v 1.17 2006-01-12 11:30:19 nkeynes Exp $
     2.7   * 
     2.8   * ARM7TDMI CPU emulation core.
     2.9   *
    2.10 @@ -18,8 +18,9 @@
    2.11  
    2.12  #define MODULE aica_module
    2.13  #include "dream.h"
    2.14 +#include "mem.h"
    2.15  #include "aica/armcore.h"
    2.16 -#include "mem.h"
    2.17 +#include "aica/aica.h"
    2.18  
    2.19  #define STM_R15_OFFSET 12
    2.20  
    2.21 @@ -46,6 +47,7 @@
    2.22  uint32_t arm_cpu_freq = ARM_BASE_RATE;
    2.23  uint32_t arm_cpu_period = 1000 / ARM_BASE_RATE;
    2.24  
    2.25 +#define CYCLES_PER_SAMPLE ((ARM_BASE_RATE * 1000000) / AICA_SAMPLE_RATE)
    2.26  
    2.27  static struct breakpoint_struct arm_breakpoints[MAX_BREAKPOINTS];
    2.28  static int arm_breakpoint_count = 0;
    2.29 @@ -85,35 +87,40 @@
    2.30      return 0;
    2.31  }
    2.32  
    2.33 -uint32_t arm_run_slice( uint32_t nanosecs )
    2.34 +#define IS_TIMER_ENABLED() (MMIO_READ( AICA2, AICA_TCR ) & 0x40)
    2.35 +
    2.36 +uint32_t arm_run_slice( uint32_t num_samples )
    2.37  {
    2.38 -    int i;
    2.39 -    uint32_t target = armr.icount + nanosecs / arm_cpu_period;
    2.40 -    uint32_t start = armr.icount;
    2.41 -    while( armr.icount < target ) {
    2.42 -	armr.icount++;
    2.43 -	if( !arm_execute_instruction() )
    2.44 +    int i,j,k;
    2.45 +    for( i=0; i<num_samples; i++ ) {
    2.46 +	for( j=0; j < CYCLES_PER_SAMPLE; j++ ) {
    2.47 +	    armr.icount++;
    2.48 +	    if( !arm_execute_instruction() )
    2.49 +		return i;
    2.50 +#ifdef ENABLE_DEBUG_MODE
    2.51 +	    for( k=0; k<arm_breakpoint_count; k++ ) {
    2.52 +		if( arm_breakpoints[k].address == armr.r[15] ) {
    2.53 +		    dreamcast_stop();
    2.54 +		    if( arm_breakpoints[k].type == BREAK_ONESHOT )
    2.55 +			arm_clear_breakpoint( armr.r[15], BREAK_ONESHOT );
    2.56 +		    return i;
    2.57 +		}
    2.58 +	    }
    2.59 +#endif	
    2.60 +	}
    2.61 +
    2.62 +	if( IS_TIMER_ENABLED() ) {
    2.63 +	    uint8_t val = MMIO_READ( AICA2, AICA_TIMER );
    2.64 +	    val++;
    2.65 +	    if( val == 0 )
    2.66 +		aica_event( AICA_EVENT_TIMER );
    2.67 +	    MMIO_WRITE( AICA2, AICA_TIMER, val );
    2.68 +	}
    2.69 +	if( !dreamcast_is_running() )
    2.70  	    break;
    2.71 -#ifdef ENABLE_DEBUG_MODE
    2.72 -	for( i=0; i<arm_breakpoint_count; i++ ) {
    2.73 -	    if( arm_breakpoints[i].address == armr.r[15] ) {
    2.74 -		break;
    2.75 -	    }
    2.76 -	}
    2.77 -	if( i != arm_breakpoint_count ) {
    2.78 -	    dreamcast_stop();
    2.79 -	    if( arm_breakpoints[i].type == BREAK_ONESHOT )
    2.80 -		arm_clear_breakpoint( armr.r[15], BREAK_ONESHOT );
    2.81 -	    break;
    2.82 -	}
    2.83 -#endif	
    2.84      }
    2.85  
    2.86 -    if( target != armr.icount ) {
    2.87 -	/* Halted - compute time actually executed */
    2.88 -	nanosecs = (armr.icount - start) * arm_cpu_period;
    2.89 -    }
    2.90 -    return nanosecs;
    2.91 +    return i;
    2.92  }
    2.93  
    2.94  void arm_save_state( FILE *f )

     3.1 --- a/src/aica/armcore.h	Tue Jan 10 14:00:00 2006 +0000
     3.2 +++ b/src/aica/armcore.h	Thu Jan 12 11:30:19 2006 +0000
     3.3 @@ -1,5 +1,5 @@
     3.4  /**
     3.5 - * $Id: armcore.h,v 1.12 2006-01-10 13:56:54 nkeynes Exp $
     3.6 + * $Id: armcore.h,v 1.13 2006-01-12 11:30:19 nkeynes Exp $
     3.7   * 
     3.8   * Interface definitions for the ARM CPU emulation core proper.
     3.9   *
    3.10 @@ -23,7 +23,7 @@
    3.11  #include <stdint.h>
    3.12  #include <stdio.h>
    3.13  
    3.14 -#define ARM_BASE_RATE 25 /* MHZ */
    3.15 +#define ARM_BASE_RATE 2 /* MHZ */
    3.16  extern uint32_t arm_cpu_freq;
    3.17  extern uint32_t arm_cpu_period;
    3.18  

     4.1 --- a/src/aica/audio.c	Tue Jan 10 14:00:00 2006 +0000
     4.2 +++ b/src/aica/audio.c	Thu Jan 12 11:30:19 2006 +0000
     4.3 @@ -1,5 +1,5 @@
     4.4  /**
     4.5 - * $Id: audio.c,v 1.1 2006-01-10 13:56:54 nkeynes Exp $
     4.6 + * $Id: audio.c,v 1.2 2006-01-12 11:30:19 nkeynes Exp $
     4.7   * 
     4.8   * Audio mixer core. Combines all the active streams into a single sound
     4.9   * buffer for output. 
    4.10 @@ -25,7 +25,7 @@
    4.11  #include <string.h>
    4.12  
    4.13  #define NUM_BUFFERS 3
    4.14 -#define MS_PER_BUFFER 4000
    4.15 +#define MS_PER_BUFFER 500
    4.16  
    4.17  #define BUFFER_EMPTY   0
    4.18  #define BUFFER_WRITING 1
    4.19 @@ -71,7 +71,7 @@
    4.20  	if( audio.output_buffers[i] != NULL )
    4.21  	    free(audio.output_buffers[i]);
    4.22  	audio.output_buffers[i] = g_malloc0( sizeof(struct audio_buffer) + samples_per_buffer * bytes_per_sample );
    4.23 -	audio.output_buffers[i]->length = samples_per_buffer;
    4.24 +	audio.output_buffers[i]->length = samples_per_buffer * bytes_per_sample;
    4.25  	audio.output_buffers[i]->posn = 0;
    4.26  	audio.output_buffers[i]->status = BUFFER_EMPTY;
    4.27      }
    4.28 @@ -175,10 +175,12 @@
    4.29  /**
    4.30   * Mix a single output sample.
    4.31   */
    4.32 -void audio_mix_sample( )
    4.33 +void audio_mix_samples( int num_samples )
    4.34  {
    4.35      int i, j;
    4.36 -    int32_t result_left = 0, result_right = 0;
    4.37 +    int32_t result_buf[num_samples][2];
    4.38 +
    4.39 +    memset( &result_buf, 0, sizeof(result_buf) );
    4.40  
    4.41      for( i=0; i < 64; i++ ) {
    4.42  	audio_channel_t channel = &audio.channels[i];
    4.43 @@ -186,74 +188,113 @@
    4.44  	    int32_t sample;
    4.45  	    switch( channel->sample_format ) {
    4.46  	    case AUDIO_FMT_16BIT:
    4.47 -		sample = *(int16_t *)(arm_mem + channel->posn + channel->start);
    4.48 +		for( j=0; j<num_samples; j++ ) {
    4.49 +		    sample = *(int16_t *)(arm_mem + channel->posn + channel->start);
    4.50 +		    result_buf[j][0] += sample * channel->vol_left;
    4.51 +		    result_buf[j][1] += sample * channel->vol_right;
    4.52 +		    
    4.53 +		    channel->posn_left += channel->sample_rate;
    4.54 +		    while( channel->posn_left > audio.output_rate ) {
    4.55 +			channel->posn_left -= audio.output_rate;
    4.56 +			channel->posn++;
    4.57 +			
    4.58 +			if( channel->posn == channel->end ) {
    4.59 +			    if( channel->loop )
    4.60 +				channel->posn = channel->loop_start;
    4.61 +			    else {
    4.62 +				audio_stop_channel(i);
    4.63 +				j = num_samples;
    4.64 +				break;
    4.65 +			    }
    4.66 +			}
    4.67 +		    }
    4.68 +		}
    4.69  		break;
    4.70  	    case AUDIO_FMT_8BIT:
    4.71 -		sample = (*(int8_t *)(arm_mem + channel->posn + channel->start)) << 8;
    4.72 -		break;
    4.73 -	    case AUDIO_FMT_16BIT|AUDIO_FMT_UNSIGNED:
    4.74 -		sample = (int8_t)((*(uint16_t *)(arm_mem + channel->posn + channel->start)) - 0x8000);
    4.75 -		break;
    4.76 -	    case AUDIO_FMT_8BIT|AUDIO_FMT_UNSIGNED:
    4.77 -		sample = (int8_t)((*(uint8_t *)(arm_mem + channel->posn + channel->start)) - 0x80);
    4.78 +		for( j=0; j<num_samples; j++ ) {
    4.79 +		    sample = (*(int8_t *)(arm_mem + channel->posn + channel->start)) << 8;
    4.80 +		    result_buf[j][0] += sample * channel->vol_left;
    4.81 +		    result_buf[j][1] += sample * channel->vol_right;
    4.82 +		    
    4.83 +		    channel->posn_left += channel->sample_rate;
    4.84 +		    while( channel->posn_left > audio.output_rate ) {
    4.85 +			channel->posn_left -= audio.output_rate;
    4.86 +			channel->posn++;
    4.87 +			
    4.88 +			if( channel->posn == channel->end ) {
    4.89 +			    if( channel->loop )
    4.90 +				channel->posn = channel->loop_start;
    4.91 +			    else {
    4.92 +				audio_stop_channel(i);
    4.93 +				j = num_samples;
    4.94 +				break;
    4.95 +			    }
    4.96 +			}
    4.97 +		    }
    4.98 +		}
    4.99  		break;
   4.100  	    case AUDIO_FMT_ADPCM:
   4.101 -		sample = (int16_t)channel->adpcm_predict;
   4.102 +		for( j=0; j<num_samples; j++ ) {
   4.103 +		    sample = (int16_t)channel->adpcm_predict;
   4.104 +		    result_buf[j][0] += sample * channel->vol_left;
   4.105 +		    result_buf[j][1] += sample * channel->vol_right;
   4.106 +		    channel->posn_left += channel->sample_rate;
   4.107 +		    while( channel->posn_left > audio.output_rate ) {
   4.108 +			channel->posn_left -= audio.output_rate;
   4.109 +			if( channel->adpcm_nibble == 0 ) {
   4.110 +			    uint8_t data = *(uint8_t *)(arm_mem + channel->posn + channel->start);
   4.111 +			    adpcm_yamaha_decode_nibble( channel, (data >> 4) & 0x0F );
   4.112 +			    channel->adpcm_nibble = 1;
   4.113 +			} else {
   4.114 +			    channel->posn++;
   4.115 +			    if( channel->posn == channel->end ) {
   4.116 +				if( channel->loop )
   4.117 +				    channel->posn = channel->loop_start;
   4.118 +				else
   4.119 +				    audio_stop_channel(i);
   4.120 +				break;
   4.121 +			    }
   4.122 +			    uint8_t data = *(uint8_t *)(arm_mem + channel->posn + channel->start);
   4.123 +			    adpcm_yamaha_decode_nibble( channel, data & 0x0F );
   4.124 +			    channel->adpcm_nibble = 0;
   4.125 +			}
   4.126 +		    }
   4.127 +		}
   4.128 +		break;
   4.129  	    default:
   4.130 -		sample = 0; /* Unsupported */
   4.131 -	    }
   4.132 -	    result_left += sample * channel->vol_left;
   4.133 -	    result_right += sample * channel->vol_right;
   4.134 -	    
   4.135 -	    channel->posn_left += channel->sample_rate;
   4.136 -	    while( channel->posn_left > audio.output_rate ) {
   4.137 -		channel->posn_left -= audio.output_rate;
   4.138 -		if( channel->sample_format == AUDIO_FMT_ADPCM &&
   4.139 -		    channel->adpcm_nibble == 0 ) {
   4.140 -		    uint8_t data = *(uint8_t *)(arm_mem + channel->posn + channel->start);
   4.141 -		    adpcm_yamaha_decode_nibble( channel, (data >> 4) & 0x0F );
   4.142 -		    channel->adpcm_nibble = 1;
   4.143 -		    continue;
   4.144 -		}
   4.145 -
   4.146 -		channel->posn++;
   4.147 -
   4.148 -		if( channel->loop_count != 0 && 
   4.149 -		    channel->posn >= channel->loop_end ) {
   4.150 -		    channel->posn = channel->loop_start;
   4.151 -		    if( channel->loop_count != -1 )
   4.152 -			channel->loop_count --;
   4.153 -		} else if( channel->posn >= channel->end ) {
   4.154 -		    audio_stop_channel( i );
   4.155 -		    break;
   4.156 -		}
   4.157 -
   4.158 -		if( channel->sample_format == AUDIO_FMT_ADPCM ) {
   4.159 -		    uint8_t data = *(uint8_t *)(arm_mem + channel->posn + channel->start);
   4.160 -		    adpcm_yamaha_decode_nibble( channel, data & 0x0F );
   4.161 -		    channel->adpcm_nibble = 0;
   4.162 -		}
   4.163 +		break;
   4.164  	    }
   4.165  	}
   4.166      }
   4.167 -
   4.168 +	    
   4.169      /* Down-render to the final output format */
   4.170 -    audio_buffer_t buf = 
   4.171 -	audio.output_buffers[audio.write_buffer];
   4.172 +    
   4.173      if( audio.output_format & AUDIO_FMT_16BIT ) {
   4.174 -	uint16_t *data = (uint16_t *)&buf->data[buf->posn*audio.output_sample_size];
   4.175 -	*data++ = (int16_t)(result_left >> 8);
   4.176 -	*data++ = (int16_t)(result_right >> 8);
   4.177 +	audio_buffer_t buf = audio.output_buffers[audio.write_buffer];
   4.178 +	uint16_t *data = (uint16_t *)&buf->data[buf->posn];
   4.179 +	for( j=0; j < num_samples; j++ ) {
   4.180 +	    *data++ = (int16_t)(result_buf[j][0] >> 8);
   4.181 +	    *data++ = (int16_t)(result_buf[j][1] >> 8);	
   4.182 +	    buf->posn += 4;
   4.183 +	    if( buf->posn == buf->length ) {
   4.184 +		audio_next_write_buffer();
   4.185 +		buf = audio.output_buffers[audio.write_buffer];
   4.186 +		data = (uint16_t *)&buf->data[0];
   4.187 +	    }
   4.188 +	}
   4.189      } else {
   4.190 -	audio_buffer_t buf = 
   4.191 -	    audio.output_buffers[audio.write_buffer];
   4.192 -	uint8_t *data = (uint8_t *)&buf->data[buf->posn*audio.output_sample_size];
   4.193 -	*data++ = (int8_t)(result_left >> 22);
   4.194 -	*data++ = (int8_t)(result_right >>22);
   4.195 -    }
   4.196 -    buf->posn++;
   4.197 -    if( buf->posn == buf->length ) {
   4.198 -	audio_next_write_buffer();
   4.199 +	audio_buffer_t buf = audio.output_buffers[audio.write_buffer];
   4.200 +	uint8_t *data = (uint8_t *)&buf->data[buf->posn];
   4.201 +	for( j=0; j < num_samples; j++ ) {
   4.202 +	    *data++ = (uint8_t)(result_buf[j][0] >> 16);
   4.203 +	    *data++ = (uint8_t)(result_buf[j][1] >> 16);	
   4.204 +	    buf->posn += 2;
   4.205 +	    if( buf->posn == buf->length ) {
   4.206 +		audio_next_write_buffer();
   4.207 +		buf = audio.output_buffers[audio.write_buffer];
   4.208 +		data = (uint8_t *)&buf->data[0];
   4.209 +	    }
   4.210 +	}
   4.211      }
   4.212  }
   4.213  

     5.1 --- a/src/aica/audio.h	Tue Jan 10 14:00:00 2006 +0000
     5.2 +++ b/src/aica/audio.h	Thu Jan 12 11:30:19 2006 +0000
     5.3 @@ -1,5 +1,5 @@
     5.4  /**
     5.5 - * $Id: audio.h,v 1.1 2006-01-10 13:56:54 nkeynes Exp $
     5.6 + * $Id: audio.h,v 1.2 2006-01-12 11:30:19 nkeynes Exp $
     5.7   * 
     5.8   * Audio engine, ie the part that does the actual work.
     5.9   *
    5.10 @@ -40,9 +40,8 @@
    5.11      uint32_t posn_left;
    5.12      uint32_t start;
    5.13      uint32_t end;
    5.14 +    gboolean loop;
    5.15      uint32_t loop_start;
    5.16 -    uint32_t loop_end;
    5.17 -    int loop_count; /* 0 = no loop, -1 = loop forever */
    5.18      int vol_left; /* 0..255 */
    5.19      int vol_right; /* 0..255 */
    5.20      uint32_t sample_rate;
    5.21 @@ -56,8 +55,8 @@
    5.22  
    5.23  
    5.24  typedef struct audio_buffer {
    5.25 -    uint32_t length; /* Samples */
    5.26 -    uint32_t posn; /* Samples */
    5.27 +    uint32_t length; /* Bytes */
    5.28 +    uint32_t posn; /* Bytes */
    5.29      int status;
    5.30      char data[0];
    5.31  } *audio_buffer_t;
    5.32 @@ -96,7 +95,7 @@
    5.33  /**
    5.34   * Mix a single output sample and append it to the output buffers
    5.35   */
    5.36 -void audio_mix_sample( void );
    5.37 +void audio_mix_samples( int num_samples );
    5.38  
    5.39  /**
    5.40   * Retrieve the channel information for the channel, numbered 0..63. 

     6.1 --- a/src/drivers/audio_esd.c	Tue Jan 10 14:00:00 2006 +0000
     6.2 +++ b/src/drivers/audio_esd.c	Thu Jan 12 11:30:19 2006 +0000
     6.3 @@ -1,5 +1,5 @@
     6.4  /**
     6.5 - * $Id: audio_esd.c,v 1.1 2006-01-10 13:56:54 nkeynes Exp $
     6.6 + * $Id: audio_esd.c,v 1.2 2006-01-12 11:30:19 nkeynes Exp $
     6.7   * 
     6.8   * The esd (esound) audio driver
     6.9   *
    6.10 @@ -31,11 +31,9 @@
    6.11      esd_sample_size = 1;
    6.12      if( format & AUDIO_FMT_16BIT ) {
    6.13  	esd_format |= ESD_BITS16;
    6.14 -	esd_sample_size = 1;
    6.15      } else esd_format |= ESD_BITS8;
    6.16      if( format & AUDIO_FMT_STEREO ) {
    6.17  	esd_format |= ESD_STEREO;
    6.18 -	esd_sample_size = esd_sample_size << 1 ;
    6.19      }
    6.20      else esd_format |= ESD_MONO;
    6.21      
    6.22 @@ -46,7 +44,7 @@
    6.23  gboolean esd_audio_process_buffer( audio_buffer_t buffer )
    6.24  {
    6.25      if( esd_handle != -1 ) {
    6.26 -	write( esd_handle, buffer->data, buffer->length * esd_sample_size );
    6.27 +	write( esd_handle, buffer->data, buffer->length );
    6.28  	DEBUG("Wrote buffer" );
    6.29      } else {
    6.30  	ERROR( "ESD not initialized" );