1.1 --- a/src/aica/audio.c	Mon Jan 16 11:23:05 2006 +0000
     1.2 +++ b/src/aica/audio.c	Sun Jan 22 22:40:05 2006 +0000
     1.3 @@ -1,5 +1,5 @@
     1.4  /**
     1.5 - * $Id: audio.c,v 1.3 2006-01-16 11:23:05 nkeynes Exp $
     1.6 + * $Id: audio.c,v 1.4 2006-01-17 12:54:02 nkeynes Exp $
     1.7   * 
     1.8   * Audio mixer core. Combines all the active streams into a single sound
     1.9   * buffer for output. 
    1.10 @@ -25,7 +25,7 @@
    1.11  #include <string.h>
    1.12  
    1.13  #define NUM_BUFFERS 3
    1.14 -#define MS_PER_BUFFER 100
    1.15 +#define MS_PER_BUFFER 1000
    1.16  
    1.17  #define BUFFER_EMPTY   0
    1.18  #define BUFFER_WRITING 1
    1.19 @@ -186,12 +186,14 @@
    1.20  	audio_channel_t channel = &audio.channels[i];
    1.21  	if( channel->active ) {
    1.22  	    int32_t sample;
    1.23 +	    int vol_left = (channel->vol * (32 - channel->pan)) >> 5;
    1.24 +	    int vol_right = (channel->vol * (channel->pan + 1)) >> 5;
    1.25  	    switch( channel->sample_format ) {
    1.26  	    case AUDIO_FMT_16BIT:
    1.27  		for( j=0; j<num_samples; j++ ) {
    1.28  		    sample = *(int16_t *)(arm_mem + channel->posn + channel->start);
    1.29 -		    result_buf[j][0] += sample * channel->vol_left;
    1.30 -		    result_buf[j][1] += sample * channel->vol_right;
    1.31 +		    result_buf[j][0] += sample * vol_left;
    1.32 +		    result_buf[j][1] += sample * vol_right;
    1.33  		    
    1.34  		    channel->posn_left += channel->sample_rate;
    1.35  		    while( channel->posn_left > audio.output_rate ) {
    1.36 @@ -213,8 +215,8 @@
    1.37  	    case AUDIO_FMT_8BIT:
    1.38  		for( j=0; j<num_samples; j++ ) {
    1.39  		    sample = (*(int8_t *)(arm_mem + channel->posn + channel->start)) << 8;
    1.40 -		    result_buf[j][0] += sample * channel->vol_left;
    1.41 -		    result_buf[j][1] += sample * channel->vol_right;
    1.42 +		    result_buf[j][0] += sample * vol_left;
    1.43 +		    result_buf[j][1] += sample * vol_right;
    1.44  		    
    1.45  		    channel->posn_left += channel->sample_rate;
    1.46  		    while( channel->posn_left > audio.output_rate ) {
    1.47 @@ -236,8 +238,8 @@
    1.48  	    case AUDIO_FMT_ADPCM:
    1.49  		for( j=0; j<num_samples; j++ ) {
    1.50  		    sample = (int16_t)channel->adpcm_predict;
    1.51 -		    result_buf[j][0] += sample * channel->vol_left;
    1.52 -		    result_buf[j][1] += sample * channel->vol_right;
    1.53 +		    result_buf[j][0] += sample * vol_left;
    1.54 +		    result_buf[j][1] += sample * vol_right;
    1.55  		    channel->posn_left += channel->sample_rate;
    1.56  		    while( channel->posn_left > audio.output_rate ) {
    1.57  			channel->posn_left -= audio.output_rate;
    1.58 @@ -273,8 +275,8 @@
    1.59  	audio_buffer_t buf = audio.output_buffers[audio.write_buffer];
    1.60  	uint16_t *data = (uint16_t *)&buf->data[buf->posn];
    1.61  	for( j=0; j < num_samples; j++ ) {
    1.62 -	    *data++ = (int16_t)(result_buf[j][0] >> 8);
    1.63 -	    *data++ = (int16_t)(result_buf[j][1] >> 8);	
    1.64 +	    *data++ = (int16_t)(result_buf[j][0] >> 6);
    1.65 +	    *data++ = (int16_t)(result_buf[j][1] >> 6);	
    1.66  	    buf->posn += 4;
    1.67  	    if( buf->posn == buf->length ) {
    1.68  		audio_next_write_buffer();