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lxdream.org :: lxdream/src/aica/audio.c
lxdream 0.9.1
released Jun 29
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filename src/aica/audio.c
changeset 697:479b8c213f61
prev657:c4143facbfcb
next700:4650d0c7f6f9
author nkeynes
date Sun Jun 22 04:01:27 2008 +0000 (13 years ago)
permissions -rw-r--r--
last change Commit work-in-progress CoreAudio driver
(along with various changes to the audio subsystem)
file annotate diff log raw
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/**
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 * $Id$
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 * 
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 * Audio mixer core. Combines all the active streams into a single sound
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 * buffer for output. 
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 *
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 * Copyright (c) 2005 Nathan Keynes.
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 *
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 * This program is free software; you can redistribute it and/or modify
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 * it under the terms of the GNU General Public License as published by
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 * the Free Software Foundation; either version 2 of the License, or
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 * (at your option) any later version.
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 *
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 * This program is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
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 * GNU General Public License for more details.
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 */
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#include "aica/aica.h"
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#include "aica/audio.h"
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#include "glib/gmem.h"
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#include "dream.h"
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#include <assert.h>
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#include <string.h>
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extern struct audio_driver audio_null_driver;
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extern struct audio_driver audio_osx_driver;
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extern struct audio_driver audio_pulse_driver;
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extern struct audio_driver audio_esd_driver;
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extern struct audio_driver audio_alsa_driver;
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audio_driver_t audio_driver_list[] = {
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#ifdef HAVE_CORE_AUDIO
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      &audio_osx_driver,
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#endif
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#ifdef HAVE_PULSE
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      &audio_pulse_driver,
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#endif
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#ifdef HAVE_ESOUND
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      &audio_esd_driver,
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#endif
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#ifdef HAVE_ALSA
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      &audio_alsa_driver,
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#endif
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      &audio_null_driver,
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      NULL };
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#define NUM_BUFFERS 3
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#define MS_PER_BUFFER 1000
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#define BUFFER_EMPTY   0
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#define BUFFER_WRITING 1
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#define BUFFER_FULL    2
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struct audio_state {
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    audio_buffer_t output_buffers[NUM_BUFFERS];
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    int write_buffer;
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    int read_buffer;
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    uint32_t output_format;
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    uint32_t output_rate;
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    uint32_t output_sample_size;
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    struct audio_channel channels[AUDIO_CHANNEL_COUNT];
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} audio;
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audio_driver_t audio_driver = NULL;
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#define NEXT_BUFFER() ((audio.write_buffer == NUM_BUFFERS-1) ? 0 : audio.write_buffer+1)
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extern char *arm_mem;
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/**
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 * Preserve audio channel state only - don't bother saving the buffers
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 */
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void audio_save_state( FILE *f )
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{
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    fwrite( &audio.channels[0], sizeof(struct audio_channel), AUDIO_CHANNEL_COUNT, f );
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}
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int audio_load_state( FILE *f )
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{
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    int read = fread( &audio.channels[0], sizeof(struct audio_channel), AUDIO_CHANNEL_COUNT, f );
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    return (read == AUDIO_CHANNEL_COUNT ? 0 : -1 );
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}
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audio_driver_t get_audio_driver_by_name( const char *name )
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{
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    int i;
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    if( name == NULL ) {
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        return audio_driver_list[0];
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    }
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    for( i=0; audio_driver_list[i] != NULL; i++ ) {
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        if( strcasecmp( audio_driver_list[i]->name, name ) == 0 ) {
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            return audio_driver_list[i];
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        }
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    }
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    return NULL;
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}
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audio_driver_t audio_init_driver( const char *preferred_driver )
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{
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    audio_driver_t audio_driver = get_audio_driver_by_name(preferred_driver);
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    if( audio_driver == NULL ) {
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        ERROR( "Audio driver '%s' not found, aborting.", preferred_driver );
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        exit(2);
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    } else if( audio_set_driver( audio_driver ) == FALSE ) {
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        ERROR( "Failed to initialize audio driver '%s', using null driver", 
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                audio_driver->name );
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        audio_set_driver( &audio_null_driver );
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    }    
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}
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/**
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 * Set the output driver, sample rate and format. Also initializes the 
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 * output buffers, flushing any current data and reallocating as 
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 * necessary.
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 */
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gboolean audio_set_driver( audio_driver_t driver )
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{
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    uint32_t bytes_per_sample = 1;
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    uint32_t samples_per_buffer;
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    int i;
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    if( audio_driver == NULL || driver != NULL ) {
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        if( driver == NULL  )
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            driver = &audio_null_driver;
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        if( driver != audio_driver ) {	
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            if( !driver->init() )
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                return FALSE;
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            audio_driver = driver;
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        }
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    }
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    switch( driver->sample_format & AUDIO_FMT_SAMPLE_MASK ) {
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    case AUDIO_FMT_8BIT:
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        bytes_per_sample = 1;
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        break;
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    case AUDIO_FMT_16BIT:
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        bytes_per_sample = 2;
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        break;
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    case AUDIO_FMT_FLOAT:
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        bytes_per_sample = 4;
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        break;
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    }
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    if( driver->sample_format & AUDIO_FMT_STEREO )
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        bytes_per_sample <<= 1;
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    if( driver->sample_rate == audio.output_rate &&
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            bytes_per_sample == audio.output_sample_size )
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        return TRUE;
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    samples_per_buffer = (driver->sample_rate * MS_PER_BUFFER / 1000);
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    for( i=0; i<NUM_BUFFERS; i++ ) {
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        if( audio.output_buffers[i] != NULL )
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            free(audio.output_buffers[i]);
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        audio.output_buffers[i] = g_malloc0( sizeof(struct audio_buffer) + samples_per_buffer * bytes_per_sample );
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        audio.output_buffers[i]->length = samples_per_buffer * bytes_per_sample;
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        audio.output_buffers[i]->posn = 0;
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        audio.output_buffers[i]->status = BUFFER_EMPTY;
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    }
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    audio.output_format = driver->sample_format;
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    audio.output_rate = driver->sample_rate;
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    audio.output_sample_size = bytes_per_sample;
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    audio.write_buffer = 0;
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    audio.read_buffer = 0;
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    return TRUE;
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}
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/**
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 * Mark the current write buffer as full and prepare the next buffer for
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 * writing. Returns the next buffer to write to.
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 * If all buffers are full, returns NULL.
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 */
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audio_buffer_t audio_next_write_buffer( )
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{
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    audio_buffer_t result = NULL;
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    audio_buffer_t current = audio.output_buffers[audio.write_buffer];
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    current->status = BUFFER_FULL;
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    if( audio.read_buffer == audio.write_buffer &&
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            audio_driver->process_buffer( current ) ) {
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        audio_next_read_buffer();
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    }
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    int next_buffer = NEXT_BUFFER();
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    result = audio.output_buffers[next_buffer];
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    if( result->status == BUFFER_FULL )
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        return NULL;
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    else {
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        audio.write_buffer = next_buffer;
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        result->status = BUFFER_WRITING;
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        return result;
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    }
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}
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/**
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 * Mark the current read buffer as empty and return the next buffer for
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 * reading. If there is no next buffer yet, returns NULL.
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 */
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audio_buffer_t audio_next_read_buffer( )
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{
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    audio_buffer_t current = audio.output_buffers[audio.read_buffer];
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    if( current->status == BUFFER_FULL ) {
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        // Current read buffer has data, which we've just emptied
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        current->status = BUFFER_EMPTY;
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        current->posn = 0;
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        audio.read_buffer++;
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        if( audio.read_buffer == NUM_BUFFERS )
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            audio.read_buffer = 0;
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        current = audio.output_buffers[audio.read_buffer];
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        if( current->status == BUFFER_FULL ) {
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            current->posn = 0;
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            return current;
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        }
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        else return NULL;
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    } else {
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        return NULL;
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    }
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}
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/*************************** ADPCM ***********************************/
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/**
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 * The following section borrows heavily from ffmpeg, which is
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 * copyright (c) 2001-2003 by the fine folks at the ffmpeg project,
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 * distributed under the GPL version 2 or later.
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 */
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#define CLAMP_TO_SHORT(value) \
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if (value > 32767) \
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    value = 32767; \
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else if (value < -32768) \
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    value = -32768; \
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static const int yamaha_indexscale[] = {
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    230, 230, 230, 230, 307, 409, 512, 614,
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    230, 230, 230, 230, 307, 409, 512, 614
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};
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static const int yamaha_difflookup[] = {
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    1, 3, 5, 7, 9, 11, 13, 15,
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    -1, -3, -5, -7, -9, -11, -13, -15
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};
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static inline short adpcm_yamaha_decode_nibble( audio_channel_t c, 
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						unsigned char nibble )
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{
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    if( c->adpcm_step == 0 ) {
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        c->adpcm_predict = 0;
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        c->adpcm_step = 127;
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    }
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    c->adpcm_predict += (c->adpcm_step * yamaha_difflookup[nibble]) >> 3;
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    CLAMP_TO_SHORT(c->adpcm_predict);
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    c->adpcm_step = (c->adpcm_step * yamaha_indexscale[nibble]) >> 8;
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    c->adpcm_step = CLAMP(c->adpcm_step, 127, 24567);
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    return c->adpcm_predict;
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}
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/*************************** Sample mixer *****************************/
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/**
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 * Mix a single output sample.
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 */
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void audio_mix_samples( int num_samples )
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{
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    int i, j;
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    int32_t result_buf[num_samples][2];
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    memset( &result_buf, 0, sizeof(result_buf) );
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    for( i=0; i < AUDIO_CHANNEL_COUNT; i++ ) {
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        audio_channel_t channel = &audio.channels[i];
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        if( channel->active ) {
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            int32_t sample;
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            int vol_left = (channel->vol * (32 - channel->pan)) >> 5;
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            int vol_right = (channel->vol * (channel->pan + 1)) >> 5;
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            switch( channel->sample_format ) {
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            case AUDIO_FMT_16BIT:
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                for( j=0; j<num_samples; j++ ) {
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                    sample = ((int16_t *)(arm_mem + channel->start))[channel->posn];
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                    result_buf[j][0] += sample * vol_left;
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                    result_buf[j][1] += sample * vol_right;
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                    channel->posn_left += channel->sample_rate;
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                    while( channel->posn_left > audio.output_rate ) {
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                        channel->posn_left -= audio.output_rate;
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                        channel->posn++;
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   291
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   292
                        if( channel->posn == channel->end ) {
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                            if( channel->loop ) {
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                                channel->posn = channel->loop_start;
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   295
                                channel->loop = LOOP_LOOPED;
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   296
                            } else {
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                                audio_stop_channel(i);
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                                j = num_samples;
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                                break;
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   300
                            }
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                        }
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                    }
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                }
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                break;
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   305
            case AUDIO_FMT_8BIT:
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                for( j=0; j<num_samples; j++ ) {
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                    sample = ((int8_t *)(arm_mem + channel->start))[channel->posn] << 8;
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   308
                    result_buf[j][0] += sample * vol_left;
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   309
                    result_buf[j][1] += sample * vol_right;
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   310
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   311
                    channel->posn_left += channel->sample_rate;
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   312
                    while( channel->posn_left > audio.output_rate ) {
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   313
                        channel->posn_left -= audio.output_rate;
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   314
                        channel->posn++;
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   315
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   316
                        if( channel->posn == channel->end ) {
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   317
                            if( channel->loop ) {
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   318
                                channel->posn = channel->loop_start;
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   319
                                channel->loop = LOOP_LOOPED;
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   320
                            } else {
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   321
                                audio_stop_channel(i);
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   322
                                j = num_samples;
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   323
                                break;
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   324
                            }
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   325
                        }
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   326
                    }
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   327
                }
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   328
                break;
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   329
            case AUDIO_FMT_ADPCM:
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   330
                for( j=0; j<num_samples; j++ ) {
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   331
                    sample = (int16_t)channel->adpcm_predict;
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   332
                    result_buf[j][0] += sample * vol_left;
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   333
                    result_buf[j][1] += sample * vol_right;
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   334
                    channel->posn_left += channel->sample_rate;
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   335
                    while( channel->posn_left > audio.output_rate ) {
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   336
                        channel->posn_left -= audio.output_rate;
nkeynes@697
   337
                        channel->posn++;
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   338
                        if( channel->posn == channel->end ) {
nkeynes@697
   339
                            if( channel->loop ) {
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   340
                                channel->posn = channel->loop_start;
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   341
                                channel->loop = LOOP_LOOPED;
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   342
                                channel->adpcm_predict = 0;
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   343
                                channel->adpcm_step = 0;
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   344
                            } else {
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   345
                                audio_stop_channel(i);
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   346
                                j = num_samples;
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   347
                                break;
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   348
                            }
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   349
                        }
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   350
                        uint8_t data = ((uint8_t *)(arm_mem + channel->start))[channel->posn>>1];
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   351
                        if( channel->posn&1 ) {
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   352
                            adpcm_yamaha_decode_nibble( channel, (data >> 4) & 0x0F );
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   353
                        } else {
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   354
                            adpcm_yamaha_decode_nibble( channel, data & 0x0F );
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   355
                        }
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   356
                    }
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   357
                }
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   358
                break;
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   359
            default:
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   360
                break;
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   361
            }
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   362
        }
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   363
    }
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   364
	    
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   365
    /* Down-render to the final output format */
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   366
    audio_buffer_t buf = audio.output_buffers[audio.write_buffer];
nkeynes@697
   367
    if( buf->status == BUFFER_FULL ) {
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   368
        buf = audio_next_write_buffer();
nkeynes@697
   369
        if( buf == NULL ) { // no available space
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   370
            return;
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   371
        }
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   372
    }
nkeynes@73
   373
    
nkeynes@697
   374
    switch( audio.output_format & AUDIO_FMT_SAMPLE_MASK ) {
nkeynes@697
   375
    case AUDIO_FMT_FLOAT: {
nkeynes@697
   376
        float scale = 1.0/SHRT_MAX;
nkeynes@697
   377
        float *data = (float *)&buf->data[buf->posn];
nkeynes@697
   378
        for( j=0; j<num_samples; j++ ) {
nkeynes@697
   379
            *data++ = scale * (result_buf[j][0] >> 6);
nkeynes@697
   380
            *data++ = scale * (result_buf[j][1] >> 6);
nkeynes@697
   381
            buf->posn += 8;
nkeynes@697
   382
            if( buf->posn == buf->length ) {
nkeynes@697
   383
                buf = audio_next_write_buffer();
nkeynes@697
   384
                if( buf == NULL ) {
nkeynes@697
   385
                    break;
nkeynes@697
   386
                }
nkeynes@697
   387
                data = (float *)&buf->data[0];
nkeynes@697
   388
            }
nkeynes@697
   389
        }
nkeynes@697
   390
        break;
nkeynes@697
   391
    }
nkeynes@697
   392
    case AUDIO_FMT_16BIT: {
nkeynes@697
   393
        int16_t *data = (int16_t *)&buf->data[buf->posn];
nkeynes@697
   394
        for( j=0; j < num_samples; j++ ) {
nkeynes@697
   395
            *data++ = (int16_t)(result_buf[j][0] >> 6);
nkeynes@697
   396
            *data++ = (int16_t)(result_buf[j][1] >> 6);	
nkeynes@697
   397
            buf->posn += 4;
nkeynes@697
   398
            if( buf->posn == buf->length ) {
nkeynes@697
   399
                buf = audio_next_write_buffer();
nkeynes@697
   400
                if( buf == NULL ) {
nkeynes@697
   401
                    // All buffers are full
nkeynes@697
   402
                    break;
nkeynes@697
   403
                }
nkeynes@697
   404
                data = (int16_t *)&buf->data[0];
nkeynes@697
   405
            }
nkeynes@697
   406
        }
nkeynes@697
   407
        break;
nkeynes@697
   408
    }
nkeynes@697
   409
    case AUDIO_FMT_8BIT: {
nkeynes@697
   410
        int8_t *data = (uint8_t *)&buf->data[buf->posn];
nkeynes@697
   411
        for( j=0; j < num_samples; j++ ) {
nkeynes@697
   412
            *data++ = (int8_t)(result_buf[j][0] >> 16);
nkeynes@697
   413
            *data++ = (int8_t)(result_buf[j][1] >> 16);	
nkeynes@697
   414
            buf->posn += 2;
nkeynes@697
   415
            if( buf->posn == buf->length ) {
nkeynes@697
   416
                buf = audio_next_write_buffer();
nkeynes@697
   417
                if( buf == NULL ) {
nkeynes@697
   418
                    // All buffers are full
nkeynes@697
   419
                    break;
nkeynes@697
   420
                }
nkeynes@697
   421
                buf = audio.output_buffers[audio.write_buffer];
nkeynes@697
   422
                data = (uint8_t *)&buf->data[0];
nkeynes@697
   423
            }
nkeynes@697
   424
        }
nkeynes@697
   425
        break;
nkeynes@697
   426
    }
nkeynes@66
   427
    }
nkeynes@66
   428
}
nkeynes@66
   429
nkeynes@66
   430
/********************** Internal AICA calls ***************************/
nkeynes@66
   431
nkeynes@66
   432
audio_channel_t audio_get_channel( int channel ) 
nkeynes@66
   433
{
nkeynes@66
   434
    return &audio.channels[channel];
nkeynes@66
   435
}
nkeynes@66
   436
nkeynes@434
   437
void audio_start_stop_channel( int channel, gboolean start )
nkeynes@434
   438
{
nkeynes@434
   439
    if( audio.channels[channel].active ) {
nkeynes@434
   440
	if( !start ) {
nkeynes@434
   441
	    audio_stop_channel(channel);
nkeynes@434
   442
	}
nkeynes@434
   443
    } else if( start ) {
nkeynes@434
   444
	audio_start_channel(channel);
nkeynes@434
   445
    }
nkeynes@434
   446
}
nkeynes@434
   447
nkeynes@66
   448
void audio_stop_channel( int channel ) 
nkeynes@66
   449
{
nkeynes@66
   450
    audio.channels[channel].active = FALSE;
nkeynes@66
   451
}
nkeynes@66
   452
nkeynes@66
   453
nkeynes@66
   454
void audio_start_channel( int channel )
nkeynes@66
   455
{
nkeynes@66
   456
    audio.channels[channel].posn = 0;
nkeynes@66
   457
    audio.channels[channel].posn_left = 0;
nkeynes@66
   458
    audio.channels[channel].active = TRUE;
nkeynes@434
   459
    if( audio.channels[channel].sample_format == AUDIO_FMT_ADPCM ) {
nkeynes@434
   460
	audio.channels[channel].adpcm_step = 0;
nkeynes@434
   461
	audio.channels[channel].adpcm_predict = 0;
nkeynes@434
   462
	uint8_t data = ((uint8_t *)(arm_mem + audio.channels[channel].start))[0];
nkeynes@434
   463
	adpcm_yamaha_decode_nibble( &audio.channels[channel], data & 0x0F );
nkeynes@434
   464
    }
nkeynes@66
   465
}
.