4 * Audio mixer core. Combines all the active streams into a single sound
7 * Copyright (c) 2005 Nathan Keynes.
9 * This program is free software; you can redistribute it and/or modify
10 * it under the terms of the GNU General Public License as published by
11 * the Free Software Foundation; either version 2 of the License, or
12 * (at your option) any later version.
14 * This program is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 * GNU General Public License for more details.
20 #include "aica/aica.h"
21 #include "aica/audio.h"
22 #include <glib/gmem.h>
28 extern struct audio_driver audio_null_driver;
29 extern struct audio_driver audio_osx_driver;
30 extern struct audio_driver audio_pulse_driver;
31 extern struct audio_driver audio_esd_driver;
32 extern struct audio_driver audio_alsa_driver;
34 audio_driver_t audio_driver_list[] = {
35 #ifdef HAVE_CORE_AUDIO
51 #define MS_PER_BUFFER 100
53 #define BUFFER_EMPTY 0
54 #define BUFFER_WRITING 1
58 audio_buffer_t output_buffers[NUM_BUFFERS];
61 uint32_t output_format;
63 uint32_t output_sample_size;
64 struct audio_channel channels[AUDIO_CHANNEL_COUNT];
67 audio_driver_t audio_driver = NULL;
69 #define NEXT_BUFFER() ((audio.write_buffer == NUM_BUFFERS-1) ? 0 : audio.write_buffer+1)
74 * Preserve audio channel state only - don't bother saving the buffers
76 void audio_save_state( FILE *f )
78 fwrite( &audio.channels[0], sizeof(struct audio_channel), AUDIO_CHANNEL_COUNT, f );
81 int audio_load_state( FILE *f )
83 int read = fread( &audio.channels[0], sizeof(struct audio_channel), AUDIO_CHANNEL_COUNT, f );
84 return (read == AUDIO_CHANNEL_COUNT ? 0 : -1 );
87 audio_driver_t get_audio_driver_by_name( const char *name )
91 return audio_driver_list[0];
93 for( i=0; audio_driver_list[i] != NULL; i++ ) {
94 if( strcasecmp( audio_driver_list[i]->name, name ) == 0 ) {
95 return audio_driver_list[i];
102 void print_audio_drivers( FILE * out )
105 fprintf( out, "Available audio drivers:\n" );
106 for( i=0; audio_driver_list[i] != NULL; i++ ) {
107 fprintf( out, " %-8s %s\n", audio_driver_list[i]->name,
108 gettext(audio_driver_list[i]->description) );
112 audio_driver_t audio_init_driver( const char *preferred_driver )
114 audio_driver_t audio_driver = get_audio_driver_by_name(preferred_driver);
115 if( audio_driver == NULL ) {
116 ERROR( "Audio driver '%s' not found, aborting.", preferred_driver );
118 } else if( audio_set_driver( audio_driver ) == FALSE ) {
119 ERROR( "Failed to initialize audio driver '%s', using null driver",
120 audio_driver->name );
121 audio_set_driver( &audio_null_driver );
126 * Set the output driver, sample rate and format. Also initializes the
127 * output buffers, flushing any current data and reallocating as
130 gboolean audio_set_driver( audio_driver_t driver )
132 uint32_t bytes_per_sample = 1;
133 uint32_t samples_per_buffer;
136 if( audio_driver == NULL || driver != NULL ) {
138 driver = &audio_null_driver;
139 if( driver != audio_driver ) {
140 if( !driver->init() )
142 audio_driver = driver;
146 switch( driver->sample_format & AUDIO_FMT_SAMPLE_MASK ) {
148 bytes_per_sample = 1;
150 case AUDIO_FMT_16BIT:
151 bytes_per_sample = 2;
153 case AUDIO_FMT_FLOAT:
154 bytes_per_sample = 4;
158 if( driver->sample_format & AUDIO_FMT_STEREO )
159 bytes_per_sample <<= 1;
160 if( driver->sample_rate == audio.output_rate &&
161 bytes_per_sample == audio.output_sample_size )
163 samples_per_buffer = (driver->sample_rate * MS_PER_BUFFER / 1000);
164 for( i=0; i<NUM_BUFFERS; i++ ) {
165 if( audio.output_buffers[i] != NULL )
166 free(audio.output_buffers[i]);
167 audio.output_buffers[i] = g_malloc0( sizeof(struct audio_buffer) + samples_per_buffer * bytes_per_sample );
168 audio.output_buffers[i]->length = samples_per_buffer * bytes_per_sample;
169 audio.output_buffers[i]->posn = 0;
170 audio.output_buffers[i]->status = BUFFER_EMPTY;
172 audio.output_format = driver->sample_format;
173 audio.output_rate = driver->sample_rate;
174 audio.output_sample_size = bytes_per_sample;
175 audio.write_buffer = 0;
176 audio.read_buffer = 0;
182 * Mark the current write buffer as full and prepare the next buffer for
183 * writing. Returns the next buffer to write to.
184 * If all buffers are full, returns NULL.
186 audio_buffer_t audio_next_write_buffer( )
188 audio_buffer_t result = NULL;
189 audio_buffer_t current = audio.output_buffers[audio.write_buffer];
190 current->status = BUFFER_FULL;
191 if( audio.read_buffer == audio.write_buffer &&
192 audio_driver->process_buffer( current ) ) {
193 audio_next_read_buffer();
195 int next_buffer = NEXT_BUFFER();
196 result = audio.output_buffers[next_buffer];
197 if( result->status == BUFFER_FULL )
200 audio.write_buffer = next_buffer;
201 result->status = BUFFER_WRITING;
207 * Mark the current read buffer as empty and return the next buffer for
208 * reading. If there is no next buffer yet, returns NULL.
210 audio_buffer_t audio_next_read_buffer( )
212 audio_buffer_t current = audio.output_buffers[audio.read_buffer];
213 if( current->status == BUFFER_FULL ) {
214 // Current read buffer has data, which we've just emptied
215 current->status = BUFFER_EMPTY;
218 if( audio.read_buffer == NUM_BUFFERS )
219 audio.read_buffer = 0;
221 current = audio.output_buffers[audio.read_buffer];
222 if( current->status == BUFFER_FULL ) {
233 /*************************** ADPCM ***********************************/
236 * The following section borrows heavily from ffmpeg, which is
237 * copyright (c) 2001-2003 by the fine folks at the ffmpeg project,
238 * distributed under the GPL version 2 or later.
241 #define CLAMP_TO_SHORT(value) \
244 else if (value < -32768) \
247 static const int yamaha_indexscale[] = {
248 230, 230, 230, 230, 307, 409, 512, 614,
249 230, 230, 230, 230, 307, 409, 512, 614
252 static const int yamaha_difflookup[] = {
253 1, 3, 5, 7, 9, 11, 13, 15,
254 -1, -3, -5, -7, -9, -11, -13, -15
257 static inline short adpcm_yamaha_decode_nibble( audio_channel_t c,
258 unsigned char nibble )
260 if( c->adpcm_step == 0 ) {
261 c->adpcm_predict = 0;
265 c->adpcm_predict += (c->adpcm_step * yamaha_difflookup[nibble]) >> 3;
266 CLAMP_TO_SHORT(c->adpcm_predict);
267 c->adpcm_step = (c->adpcm_step * yamaha_indexscale[nibble]) >> 8;
268 c->adpcm_step = CLAMP(c->adpcm_step, 127, 24567);
269 return c->adpcm_predict;
272 /*************************** Sample mixer *****************************/
275 * Mix a single output sample.
277 void audio_mix_samples( int num_samples )
280 int32_t result_buf[num_samples][2];
282 memset( &result_buf, 0, sizeof(result_buf) );
284 for( i=0; i < AUDIO_CHANNEL_COUNT; i++ ) {
285 audio_channel_t channel = &audio.channels[i];
286 if( channel->active ) {
288 int vol_left = (channel->vol * (32 - channel->pan)) >> 5;
289 int vol_right = (channel->vol * (channel->pan + 1)) >> 5;
290 switch( channel->sample_format ) {
291 case AUDIO_FMT_16BIT:
292 for( j=0; j<num_samples; j++ ) {
293 sample = ((int16_t *)(arm_mem + channel->start))[channel->posn];
294 result_buf[j][0] += sample * vol_left;
295 result_buf[j][1] += sample * vol_right;
297 channel->posn_left += channel->sample_rate;
298 while( channel->posn_left > audio.output_rate ) {
299 channel->posn_left -= audio.output_rate;
302 if( channel->posn == channel->end ) {
303 if( channel->loop ) {
304 channel->posn = channel->loop_start;
305 channel->loop = LOOP_LOOPED;
307 audio_stop_channel(i);
316 for( j=0; j<num_samples; j++ ) {
317 sample = ((int8_t *)(arm_mem + channel->start))[channel->posn] << 8;
318 result_buf[j][0] += sample * vol_left;
319 result_buf[j][1] += sample * vol_right;
321 channel->posn_left += channel->sample_rate;
322 while( channel->posn_left > audio.output_rate ) {
323 channel->posn_left -= audio.output_rate;
326 if( channel->posn == channel->end ) {
327 if( channel->loop ) {
328 channel->posn = channel->loop_start;
329 channel->loop = LOOP_LOOPED;
331 audio_stop_channel(i);
339 case AUDIO_FMT_ADPCM:
340 for( j=0; j<num_samples; j++ ) {
341 sample = (int16_t)channel->adpcm_predict;
342 result_buf[j][0] += sample * vol_left;
343 result_buf[j][1] += sample * vol_right;
344 channel->posn_left += channel->sample_rate;
345 while( channel->posn_left > audio.output_rate ) {
346 channel->posn_left -= audio.output_rate;
348 if( channel->posn == channel->end ) {
349 if( channel->loop ) {
350 channel->posn = channel->loop_start;
351 channel->loop = LOOP_LOOPED;
352 channel->adpcm_predict = 0;
353 channel->adpcm_step = 0;
355 audio_stop_channel(i);
360 uint8_t data = ((uint8_t *)(arm_mem + channel->start))[channel->posn>>1];
361 if( channel->posn&1 ) {
362 adpcm_yamaha_decode_nibble( channel, (data >> 4) & 0x0F );
364 adpcm_yamaha_decode_nibble( channel, data & 0x0F );
375 /* Down-render to the final output format */
376 audio_buffer_t buf = audio.output_buffers[audio.write_buffer];
377 if( buf->status == BUFFER_FULL ) {
378 buf = audio_next_write_buffer();
379 if( buf == NULL ) { // no available space
384 switch( audio.output_format & AUDIO_FMT_SAMPLE_MASK ) {
385 case AUDIO_FMT_FLOAT: {
386 float scale = 1.0/SHRT_MAX;
387 float *data = (float *)&buf->data[buf->posn];
388 for( j=0; j<num_samples; j++ ) {
389 *data++ = scale * (result_buf[j][0] >> 6);
390 *data++ = scale * (result_buf[j][1] >> 6);
392 if( buf->posn == buf->length ) {
393 buf = audio_next_write_buffer();
397 data = (float *)&buf->data[0];
402 case AUDIO_FMT_16BIT: {
403 int16_t *data = (int16_t *)&buf->data[buf->posn];
404 for( j=0; j < num_samples; j++ ) {
405 *data++ = (int16_t)(result_buf[j][0] >> 6);
406 *data++ = (int16_t)(result_buf[j][1] >> 6);
408 if( buf->posn == buf->length ) {
409 buf = audio_next_write_buffer();
411 // All buffers are full
414 data = (int16_t *)&buf->data[0];
419 case AUDIO_FMT_8BIT: {
420 int8_t *data = (int8_t *)&buf->data[buf->posn];
421 for( j=0; j < num_samples; j++ ) {
422 *data++ = (int8_t)(result_buf[j][0] >> 16);
423 *data++ = (int8_t)(result_buf[j][1] >> 16);
425 if( buf->posn == buf->length ) {
426 buf = audio_next_write_buffer();
428 // All buffers are full
431 buf = audio.output_buffers[audio.write_buffer];
432 data = (int8_t *)&buf->data[0];
440 /********************** Internal AICA calls ***************************/
442 audio_channel_t audio_get_channel( int channel )
444 return &audio.channels[channel];
447 void audio_start_stop_channel( int channel, gboolean start )
449 if( audio.channels[channel].active ) {
451 audio_stop_channel(channel);
454 audio_start_channel(channel);
458 void audio_stop_channel( int channel )
460 audio.channels[channel].active = FALSE;
464 void audio_start_channel( int channel )
466 audio.channels[channel].posn = 0;
467 audio.channels[channel].posn_left = 0;
468 audio.channels[channel].active = TRUE;
469 if( audio.channels[channel].sample_format == AUDIO_FMT_ADPCM ) {
470 audio.channels[channel].adpcm_step = 0;
471 audio.channels[channel].adpcm_predict = 0;
472 uint8_t data = ((uint8_t *)(arm_mem + audio.channels[channel].start))[0];
473 adpcm_yamaha_decode_nibble( &audio.channels[channel], data & 0x0F );
.